irclog2html for #asterisk on 20060217

00:00.00tbs_yesh
00:00.10[TK]D-FenderGuess you didn't have much spare time if it took you 4 weeks :)
00:00.20tbs_[TK]D-Fender: right :)
00:00.58[TK]D-FenderI do practically "drop-in" intalls most of the time.  The real effor tends to lie in the phone configs + custom menus
00:01.31tbs_well, yeah -- but we're fooling around with all kinds of features, trying to learn-by-doing
00:02.33Drew__sounds familiar ;)
00:03.16harryvvsixtel been down part of today.
00:03.25rayvdSome ham.
00:04.58_Sam--is there a way to hang up a call that is in a queue from cli?
00:05.52*** join/#asterisk Tamarisk (n=adrian@user-6887.lns5-c11.dsl.pol.co.uk)
00:07.51*** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net)
00:07.56_Sam--soft hangup ...i see
00:08.10_Sam--poor bastard.
00:08.37Juggietheres no queue command to boot someone from the queue?
00:08.38*** join/#asterisk austinnichols101 (n=austinni@dsl-10-169.cofs.net)
00:08.49_Sam--boot them to where?
00:09.17Juggieah, no theres not.
00:09.25Juggieonly show queue and show queues
00:09.27Juggieno control
00:11.35WillySillyhow do i install the webmin module?
00:12.05*** part/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net)
00:12.27*** part/#asterisk Tamarisk (n=adrian@user-6887.lns5-c11.dsl.pol.co.uk)
00:12.28[TK]D-Fender...what webmin module?
00:12.56*** part/#asterisk Utah_Dave (n=boucha@0-2pool130-215.nas28.salt-lake-city1.ut.us.da.qwest.net)
00:13.16WillySillyhttp://ftp.digium.com/pub/asterisk/webmin/
00:13.20*** join/#asterisk sprnova (n=oregonal@www.nemirovskyfamily.com)
00:14.29[TK]D-FenderHave you LOOKED at the file?
00:14.45Juggieheh
00:14.46WillySillyyeah
00:14.48Juggiei woudnt touch that thing
00:14.49Juggieever
00:14.50_Sam--wow people pay money for the thirdlane webmin thing?
00:14.58[av]baniyep
00:15.03_Sam--insane
00:15.09[av]banipeople even pay money for pre packaged amp
00:15.20[av]baniand you will pay me money for gxp autoprovisioner :)
00:15.21_Sam--how much for the thirdlane?
00:15.24_Sam--lol
00:15.30hertellis ser (SIP express router) something that is still usable?
00:15.57sprnovahello.. did there anything specially I have to do to enable the ztdummy driver in asterisk? Running 1.0.7 on Debian 31r2 and hearing "metalic" type audio when accessing voicemail.
00:15.58hertellfound over here: http://developer.berlios.de/projects/ser/
00:16.37_Sam--sprnova :  did you "modprobe ztdummy" before you started *
00:16.54_Sam--hertell :  i think alot of people use ser still, and openser?  im no expert there
00:17.25sprnova_Samm--.. no I didn't.. new to Linux.. was trying asterisk under BSD and was told to switch to Linux for this very problem. I will give that a try.
00:17.47_Sam--sprnova :  you also may not get much support until you try a newer version of *
00:17.59_Sam--there are newer debian packages
00:18.12sprnova_Samm--.. no problem.. I will try whatever version works.
00:18.22sprnovawhat's recommended?
00:18.28hertell_Sam: do you know if that is something that should be concidered, or can asterisk overcome potential NAT-problems?
00:18.35_Sam--i ran the debian 1.2.1 package fine
00:18.49_Sam--hertell:  there are alot of workarounds.
00:18.55_Sam--most people dont need ser
00:19.05hertell_Sam: ok
00:19.34[TK]D-FenderSER is good for large installations and those needing redundancy
00:19.35*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
00:19.39Ariel_hello everyone
00:19.44hertellare the workaround * specific, or does it require more tweaking of my firewall?
00:19.48[av]bani~ser
00:19.53jbotsomebody said ser was Sip Express Router - see http://www.iptel.org/ser/
00:19.59_Sam--the workarounds are alot of times phone specific.
00:20.32_Sam--i dont know your current firewall situation in relation to where your * is...but you can tell us
00:20.36_Sam--and maybe others will have more input
00:20.47hertellok. btw. is echo something that has to do with nat, or is there any other way in getting rid of it?
00:21.03_Sam--how do you connect to regular the regular phone network?
00:21.08hertelleg. skhype has minimal echo..
00:21.10_Sam--sorry cant ttalk.
00:21.14*** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net)
00:21.29_Sam--how do you get your calls out to the regular phone network?
00:21.33_Sam--pstn
00:21.54_Sam--analog pots lines?  pri?  remote provider?
00:21.55hertellyep. But I have not yet tested that route
00:22.03_Sam--so how do you hear the echo?
00:22.08hertelli got it working a few hours ago
00:22.18sprnova_Sam--: when I tell it to modprobe ztdummy it says not found.. I did a find and couldn't find it on my system.. I guess it is a seperate debian package?
00:22.33_Sam--yep.  its called zaptel
00:22.42sprnova_Sam--.. thanks
00:22.45_Sam--1) apt-get update   2) apt-cache search zatpel
00:22.53_Sam--3) apt-get install zaptel-something
00:23.01hertelli hear the echo when I was talking yesterday with my friend (like my voice as a light backgroundnoice behind my friend)
00:23.14_Sam--how did you call your friend
00:23.24hertellohh, over fwd
00:23.41_Sam--what type of device were you calling from
00:23.52hertellspa3k->asterisk->fwd->linphone
00:24.04_Sam--could have been anything
00:24.14hertell;-)
00:24.34_Sam--how far away (ms) are you from fwd?
00:24.41_Sam--<ive never used fwd>
00:24.51hertell130
00:24.58_Sam--could be part of it
00:25.01_Sam--who knows
00:25.25sprnova_Sam--.. should I get libzap1 too?
00:25.26_Sam--i dont have an spa3k..but i THINK they are supposed to be pretty good about echo
00:25.36hertellyeah, i will skip that route too now whey i figured out how to call directly to eg. my box.. :-)
00:25.49sprnovaI have a spa3k.. echo on the FXO is common
00:26.00sprnovano echo on FXS
00:26.35hertellok
00:26.36sprnovato remove echo I was able to adjust the SPA to PSTN gain
00:26.57hertellwhere do you do that?
00:27.04hertellin spa-admin or *
00:27.29[TK]D-FenderSPA
00:27.35_Sam--sprnova :  i dont think you need libzap1 to do ztdummy.
00:27.41_Sam--<famous last words>
00:27.49*** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net)
00:27.49_Sam--its easy enough to get if you do.
00:27.50sprnovain the PSTN tab.. towards the bottom.. might be under the FXS Internalization section
00:28.15hertell[TK]D-Fender: i btw got my setup working now :-) I finally figured out the logic behind iax.conf, sip.conf and extensions.conf :-)
00:30.19hertellsprnova: my valus are @ 0.. did you change do much changes to those values?
00:30.42hertelldid you do much changes to...
00:30.47sprnova_Sam--.. ok.. I installed zaptel and libzap1.. still not able to find a ztdummy module.. where should I find it.. I should know but I just switched to Linux from Bsd today. ;-)
00:31.21sprnovahertell.. try turning SPA to PSTN 4
00:31.33hertellsprnova: what's wrong with bsd?
00:31.35sprnovaalso make sure the Adaptive echo cancelation is turned on.
00:31.40hertellok. I'll try that
00:31.44_Sam--you could try 'updatedb' first, then 'locate ztdummy'
00:32.21sprnova_Sam--.. yup.. did a cd / ; find / -name ztdummy -print... will try the updatedb then locate
00:32.27_Sam--should be somewhere in /lib/modules/yourkernelhere
00:33.00hertellsprnova: do you have the kernelheaders installed?
00:33.15hertelluhh.. nevermind ;-)
00:33.34hertellthought you where looking in /usr/src...
00:33.40*** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net)
00:33.44sprnova_Sam--.. still don't have it..hmm.
00:33.50_Sam--wondering if the regular debian zaptel didnt come with ztdummy..hold.
00:34.14_Sam--check this page:
00:34.15_Sam--http://rapid.dotsrc.org/unstable/
00:34.26_Sam--what kernel are you running
00:34.35sprnova2.4.27-2
00:34.46_Sam--damn dude its the 21st century
00:35.02_Sam--you got a problem with 2.6? :)
00:35.05sprnovajust download Debian stable.. that is came with it. ;-)
00:35.13sprnova_Sam--.. whatever works.
00:35.21*** part/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net)
00:35.33_Sam--tzafrir_laptop :  you there?
00:35.47_Sam--sprnova :  tzafrir maintains that debian package
00:35.50_Sam--<zaptel>
00:35.54*** join/#asterisk MoutaPT (i=MoutaPT@85.139.183.206)
00:36.03sprnova_Sam.. so should I reinstall a different debian release?
00:36.18MoutaPTHello, I've been installing asterFax in my *@home 2.5
00:36.24_Sam--what type of machine is this for?
00:36.27MoutaPTand i get conflict with port25
00:36.31_Sam--i mean, for screwing around and learning?
00:36.34_Sam--or for setting up for something?
00:36.36sprnovaCeleron 800mhz
00:36.39MoutaPTseems to be with SendMAil
00:37.08_Sam--sprnova :  is it just a personal box, or?
00:37.11MoutaPTAny one knows if i should keep sendMail and AsterFAx in same machine?
00:37.19sprnova_Sam--.. setting up a reliable PBX for family.
00:37.41_Sam--i think it makes sense to install the 2.6 kernel
00:38.12_Sam--i dont know how you'd upgrade what you have to 2.6....ive never done it that way.
00:38.15sprnova_Sam--.. should I do that from my current kernel or do I need to reinstall via CDROM?
00:38.22sprnovaok
00:38.25_Sam--im sure there is a way
00:38.34_Sam--like 'apt-get upgrade"...but that may break more stuff than its worth
00:38.54hertellsprnova: you can upgrade the kernel by installing the 2.6 with apt..
00:38.55*** join/#asterisk fiftyCal (n=b@69-160-145-156.ontrca.adelphia.net)
00:39.13MoutaPTAny one here got AsterFax running with SendMail in asterisk@home ? ( igot conflict port:25)
00:39.14_Sam--that wont mess up other things that are compiled against/for the 2.4 kernel?
00:39.20hertellsprnova: what do you need the ztdummy module for?
00:39.51buZzcould someone tell me what they 'monitor' in the corner of their eye (maybe not literal) for their asterisk server?
00:39.53hertellMoutaPT: I have no experience about AsterFax..
00:40.07buZzi'm currently just doing online sip peers and active channels & calls
00:40.28sprnovahertell.. I guess I need it for timing.. I am getting metalic sounding audio and dropped packets.
00:40.35sprnovabetween SPA and asterisk
00:40.42MoutaPTthanks hertell
00:40.43hertellhmm..
00:40.48_Sam--the only thing ztdummy is help timing on MOH
00:40.49hertellis it over nat?
00:40.51_Sam--and conferences/meetme
00:40.54*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
00:40.56*** join/#asterisk WasPhantom (n=neil@203-86-192-98.tasman.net)
00:41.00_Sam--it wont help regular calls
00:41.07MoutaPTtalking with me hertell?
00:41.30sprnovahertell... voicemail  between asterisk and SPA is direct on same lan.
00:41.43sprnovahertell.. SPA has nat disabled.
00:41.57hertellMoutaPT: well, i just told you that i can't help you..
00:42.08MoutaPThertell: ok
00:42.22sprnovaso is the thought that ztdummy wont help me?
00:42.24MRH2anyone know what is up with polycom firmware looks like they released 1.6.4  recently and then just updates thebootrom and sip to 1.6.5 a few days after - i don;t see any release notes though
00:42.32hertellsprnova: strange.. I run also a spa on the same lan, but with no probs..
00:42.43hertelljust the female-voice suxx ,-)
00:42.48sprnovahertel.. what asterisk version and kernel?
00:43.05sprnovaI am using 1.0.7.
00:43.30hertelli run 1.0.9 (i removed debians stock-version and installed the 1.0.9 version so that I got the Wengo-patch working)
00:43.40hertellbut I had no problem with 1.0.7
00:44.24sprnovathe problem is even worse between SJPhone as Asterisk .. also on same lan
00:44.40hertelli had problems to get a parallel installation working on debian, so I removed the stock-version
00:44.45sprnovabetween SJPhone AND Asterisk
00:45.40Mavvieset verbose should be settable per console.
00:46.02MavvieI hate it when I can't read what I'm trying to debug....
00:46.10hertellon what platform are you running sjphone?
00:46.36hertellMavvie: eg: set verbose 10
00:46.54sprnovahertell.. did you mess with any of the RTP settings in the SPA?   running sjphone on XP. via 802.11b.. and no.. I don't have any 2.4Ghz cordless phones.
00:47.06Mavviehertell: normally I have it on 3, which gives me enough information to see what's happening.
00:47.22Mavviehertell: but when I try to see what the settings of a channel are.....
00:47.28Mavviescroll scroll scroll
00:47.39hertellsprnova: no, it's more or less default..
00:48.06hertellMavvie: yep. that suxx.. I would rather have it in a file and search for stuff directly
00:48.47sprnovahertell.. when you do a a "sip show peer" what kind of ping times do you show to your SPA? I am seeing about 10ms.
00:49.42hertellhow do you enable the ping-values?
00:50.05hertelli have the status-column"unmonitored"..
00:50.31austinnichols101qualify = yes
00:50.50Mavviehertell: qualify=yes
00:51.08hertellin sip.conf?
00:51.10Mavviein the sip-configuration
00:51.15Mavvie[foo]
00:51.17Mavviequalify=yes
00:51.18sprnovaok... maybe it one of my Ethernet switches along the way.. I am going through at least 4
00:51.27sprnovadarn things
00:51.41hertellok
00:52.01sprnovaI really should try a point to point connections to rule things out.
00:52.18austinnichols101sprnova: watch out for asymetric routing...
00:52.29fugitivoanyone using rxfax and txfax version 0.0.2pre25 with asterisk-1.2.4?
00:52.54sprnovaaustinnichols101.. not going throught NAT.. is there another way to be bitten by this?
00:53.50hertellsprnova: line1 has 24ms, and PSTN 36ms
00:53.59sprnovai am hoping it is a simple as a broken ethernet switch someplace.. the noice comes and goes almost periodically.
00:55.15*** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca)
00:55.32austinnichols101sprnova: not talking about nat.  You just need to make sure that the packets take the same route to and from the endpoints.
00:55.45hertellsprnova: now when i dropped all but the spa qualify = yes, they ping dropped to 7 and 14 ms
00:55.51austinnichols101if you take one path in one direction and one path in the return direction then you can have a big issue
00:56.01*** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl)
00:56.17austinnichols101typically you'll run spanning tree on the switches in a case where you have multiple links
00:56.26sprnovawhen I watch the ethernet link light it goes on solid when the voice is good.. just before the stutter/metalic noice the light gots off for a few sec then back on
00:56.27hertellsprnova: dont know if thiw would help: localnet = 192.168.1.0/255.255.255.0
00:56.34TelamonWhen a page says a codec uses X kbps, ie, GSM uses 13 kbps, is that bits or bytes?  And can we go back in time and kick the idiot who decided to pick two B words for data sizing in the nuts?
00:57.02hertellbtw. what codec-order do you have in your sip.conf?
00:57.12*** part/#asterisk WillySilly (n=WillySil@c-24-23-145-194.hsd1.ca.comcast.net)
00:57.22sprnovaastinnichols101.. I just have cheap ethernet switches.. not intelligent ones.. can this still happen?
00:58.05Telamonsprnova: It sound like a bad network cable, if the link light is not stable.
00:58.10austinnichols101sprnova: sure.  look at how they're cabled together.  Do you have multiple links between any two switches that form a loop?
00:58.26sprnovahertel.. using ulaw and alaw codec.. disallow everything else.
00:58.37*** join/#asterisk iq (n=iq@71-214-6-43.omah.qwest.net)
00:58.50sprnovaaustinnichols101.. I do have a loop
00:59.07austinnichols101sprnova: why?
00:59.22hertellsprnova: ok.. you might concider speex..
00:59.41hertell(not sure if it comes with stock-debian *..
00:59.50*** part/#asterisk ChaotY2k (n=juergen@port-195-158-180-70.dynamic.qsc.de)
00:59.56sprnovaaustinnichols101... trying to be a little to fancy in routing my company HW vpn onto my local lan.
01:00.07sprnovastupid stuff.. I guess I should know better.
01:00.35hertell(check with find /usr/libls -la /usr/lib/asterisk/modules/ |grep cod
01:00.48hertellls -la /usr/lib/asterisk/modules/ |grep cod
01:01.08sprnovawhat does speex do?
01:01.28hertellhttp://www.speex.org/comparison.html
01:01.36hertellopensource speech-codec..
01:01.48sprnovahertel.. ok.. thanks
01:01.51hertellchanges bitrate dynamically..
01:02.26hertellnot sure how good it is (not tested yet)
01:02.35hertellsounds anyway as a good option..
01:02.57hertellsprnova: did you have that codec installed?
01:03.06sprnovaok... so I am thinking I have some ethernet switches to checkout. thanks austinnichols101 for the pointer.
01:03.49sprnovahertel.. yes its here.
01:03.57hertellok..
01:04.11*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
01:04.19sprnovahertell.. did you check your echo cancelling switches on the SPA3k?
01:04.20hertellcause when i compiled * by hand, it was missing (had to install it separately for debian..)
01:05.25hertelland recompile
01:05.41hertellsprnova: no, i have not changed those settings
01:06.12sprnovaI have my SPA to PSTN gain cranked up.. to 7.. PSTN to SPA Gain at 0.
01:06.34hertellbtw. on my line1-tab all are set to yes (Echo Canc Enable: etc)
01:06.58sprnovathere are another set in the PSTN tab
01:07.07sprnovaunder Audio Configuration section
01:07.18hertelli will test those during daylight (it's 03.07 in the middle of the night over here ;-)
01:07.47sprnovaI have then all one.. Echo Cac Enable, Echo Canc Adapt Enable, Echo Supp Enable
01:07.54sprnovaall those are ON
01:08.04hertellyep. mine too.
01:08.29sprnovathe echo seems to be worse when both the gains to and from SPA are the same.
01:08.53hertelli've have to test that next time i'm in my office..
01:08.56sprnovahaving them at different levels helped
01:09.30hertellok. now it's time to get some sleep..
01:09.53sprnovagood night hertell.
01:09.56hertell(6h til i have to wake up.. ;-)
01:10.15hertellsprnova: good luck with your *..! :-)
01:10.21sprnovathanks
01:10.41hertelli've got finally myne working thatnks to [TK]D-Fender .-)
01:11.03hertellnext project is bakground music.. etc :-)
01:11.12sprnovafun stuff
01:11.27[TK]D-FenderI'm surprised your up this late.
01:11.38[TK]D-Fenderits like 2am for you isn't it?
01:11.46hertell[TK]D-Fender: 3..
01:11.48hertell;-)
01:11.52[TK]D-Fenderoops
01:11.57[TK]D-FenderSLEEP DAMMIT!
01:12.05hertellLOL!
01:12.29hertellI got so exited when all this wonderstuff is working :-)
01:12.44hertell[TK]D-Fender: Thanks again for your help!
01:17.15sprnovaasternnichols101... ethernet switch it is.. damn thing.. loop loop loop
01:17.26sprnovathat was the problem.
01:17.40sprnovaslap my hand.
01:17.50sprnovabad sprnova bad sprnova
01:18.05*** join/#asterisk SwK (n=Silik0nJ@12-219-147-107.client.mchsi.com)
01:24.05[av]banislap mah fro!
01:25.39*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
01:29.26*** join/#asterisk iq (n=iq@71-214-6-43.omah.qwest.net)
01:29.39*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
01:30.08ManxPowerWOW!  fxotune REALLY works!
01:30.25[av]banio.o
01:31.00ManxPowerUntil now I've always been able to remove echo on analog lines without needing FXO tune.
01:31.22ManxPowerMaybe having an 11 mile loop might have something to do about it.
01:31.25*** join/#asterisk bkw__ (n=bkw_@ppp-70-128-113-60.dsl.tulsok.swbell.net)
01:31.56ManxPowerI suspect the loop itelf is not actually 11 miles long, but the CO my line is connected to is 11 miles away.
01:34.12[av]baniits probably longer
01:34.26*** join/#asterisk anonymouz666 (n=lynx@allende.redetaho.com.br)
01:34.37[av]baniloop length only seems to affect impedance though, the real delay comes if you get switched between LECs
01:34.42[av]banior go out to cellphones
01:34.51SibRphrekthis is weird
01:34.57SibRphrekmy internal extensions work
01:35.01ManxPower[av]bani, normal echo can be removed with the Asterisk EC, this echo was not.
01:35.05SibRphreki can call every extension
01:35.10ManxPowerIn fact I got echo after the dial, but before the call connected.
01:35.12SibRphreki can call out from _my_ extension
01:35.26SibRphrekbut if you dial my outside number which points to my extension the caller get's a 404
01:35.45[av]baniManxPower: thats not suprising, since the path characteristics often change radically once the other end gets connected
01:36.14[av]baniManxPower: after dial but before connect, you're just talking to the CO switch. when connected, then you get the full path
01:36.32ManxPower*nod*
01:36.40ManxPoweri.e. the only analog is my loop at that point
01:36.44[av]banii get same thing here really
01:36.47Abydos313afternoon everyone
01:36.58[av]banii have an spa3k and i can hear it racheting the EC while the far end rings
01:37.00ManxPowerI'm just happy that fxotune works.
01:37.11[av]banii guess the new EC is far better than the old ones
01:37.18[av]banibut HW EC is still the way to go
01:38.15[av]banistill seems hard to beat throwing a tellabs EC in
01:38.29[av]banibut then, tellabs has been making EC hardware for ILECs for ~20 years
01:42.41ManxPowerI've never had significant issues with echo on analog loops before.
01:43.06ManxPowerI've had no luck with Tellabs on our PRIs.  Can't figure out why.
01:44.11*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
01:44.12SibRphrekhttp://pastebin.com/558834
01:44.16SibRphrekwhy would all of a sudden this happen
01:44.20SibRphrekhttp://pastebin.com/558834
01:47.14Ariel_means it did not find the server or device
01:47.24SibRphrekbut why all of a sudden would it start
01:47.31SibRphrekit was working earlier today
01:47.32Ariel_network issues
01:47.35SibRphrekand then just poof stopped working
01:47.54SibRphrekAriel_: i would agree, but i can make calls out
01:48.15SibRphrekthe setup is werid
01:48.18*** join/#asterisk santiago (n=afsdf@63.245.86.219)
01:48.20*** join/#asterisk PaulHuynh (n=paulhuyn@c-68-44-237-105.hsd1.de.comcast.net)
01:48.21SibRphrekwe have 1 asterisk server talking to antoerh
01:48.33SibRphrekso the number is being passed from one to the other
01:48.37SibRphrekand i can't figure out where it's dying
01:48.44PaulHuynhSam are u there?
01:48.57PaulHuynhi need another SC number
01:49.37PaulHuynhdo anyone have any recommend for DID
01:49.45PaulHuynhfor south carolina
01:51.14PaulHuynhAriel_
01:51.18PaulHuynhare u there?
01:51.31Ariel_yes are you?
01:52.32Ariel_I only have an did from connection at voicepulse. They have been good for me for the pass 2 years. but it's 11 dollars now.
01:52.46PaulHuynhdamn
01:52.52SibRphrekargh!
01:52.54SibRphrekwtf!!!!
01:53.20Ariel_it's unlimited inbound and I am able to get 4 calls in at one time via iax2
01:53.31PaulHuynhoh ok
01:53.37PaulHuynhSo that not bad
01:53.47PaulHuynhbut i want to use SIP
01:54.07PaulHuynhwell that i'm must familar with
01:54.14*** join/#asterisk bjohnson (n=bjohnson@i216-58-67-128.cybersurf.com)
01:55.12austinnichols101PaulHuynh: telasip
01:55.42Ariel_PaulHuynh, they have sip as well. But I needed the to only open inbound on one port.
01:56.05austinnichols101PaulHuynh: or Didx.org / virtualphoneline.com
01:56.11PaulHuynhI need more than one 2-4 would be idead
01:56.47Ariel_didx has some good numbers but it's on a sip unsecure ip to ip address setup which I don't like
01:57.04Abydos313how much do they charge you for 4 inbound?
01:57.08*** part/#asterisk santiago (n=afsdf@63.245.86.219)
01:57.44austinnichols101I've seen them for around 5 bucks a month with a bunch of minutes attached to them (didx.org).  virtualphoneline.com has the same numbers but is around 7 bucks.
01:58.12austinnichols101telasip is 14.95/month with unlimited us outdialing
01:58.15Abydos313so that would mean you could handle 4 outgoing at the same time too?
01:58.35Abydos313austinnichols101 what about incomming
01:58.39Abydos313and a number
01:58.50austinnichols101I believe incoming is 1:1
01:58.59austinnichols101not 100% sure on that one
01:59.02Ariel_voicepulse outbound charges by the minute. not by the channel. inbound via connections is 4 channels unlimite minutes
01:59.14Abydos313ok
01:59.31Ariel_so 11 dollars for 4 channels unlimite is not bad in this case.
01:59.34Abydos313i'd personally like to setup 2 in/out on a dsl line
01:59.45Abydos313that price sounds good
02:00.15*** part/#asterisk Drew__ (n=foo@zux221-186-224.adsl.green.ch)
02:00.25linlinok, so ive got some friends connecting to my asterisk machine through iax2, using the idefisk softphone
02:00.30PaulHuynhAriel
02:00.34linlinthey cal make calls to eachothers extentions
02:00.36*** join/#asterisk XnoN (n=xnon@200.8.30.23)
02:00.42XnoNhello
02:00.52XnoNanybody speak spanish here?
02:00.54PaulHuynhdo you have to buy a basic plan to get did from voicepulse
02:00.56linlinbut when they goto make an outgoing call, outside of the machine, it will ring and connect the other phone, but neither end can hear anything
02:01.16XnoNalguien que hable español en este channel?¿
02:01.31*** join/#asterisk welles (n=welles@61.150.43.113)
02:01.36*** join/#asterisk wellng (n=welles@61.150.43.113)
02:01.38Abydos313linlin your running sip and your asterisk server is behind nat?
02:01.42wellnghi all
02:01.42Abydos313i have the same issue
02:01.50*** part/#asterisk welles (n=welles@61.150.43.113)
02:02.40linlinno, they are connecting using iax2
02:02.48linlinbut it is behind a nat, i have forwarded 1 port though
02:02.56Abydos313nevermind, iax2 doesn't have that issue
02:03.04Abydos313so i read
02:03.13linlinyeah iax2 is supposed to work well through nat
02:03.17wellngwhy meetme list cmd can not work? when i run meetme list it show me the usage. but before it show me all the active conference
02:03.28Ariel_PaulHuynh,  no just from this site: http://connect.voicepulse.com/
02:03.30linlinbut its wierd, internal asterisk calls work, but calls to the pstn dont
02:04.06*** join/#asterisk krischnoff (n=chatzill@209.77.205.12)
02:04.40wellngtzafrir, can u help me? why meetme list cmd can not work? when i run meetme list it show me the usage. but before it show me all the active conference.
02:04.48Abydos313maybe something is wrong with your dailplan
02:04.51linlinany ideas anyone?
02:05.30Ariel_linlin, what is the error your getting when you try a pstn call
02:07.13*** join/#asterisk wellng (n=welles@61.150.43.113)
02:08.16wellngwhat's wrong?
02:09.57*** join/#asterisk voip470 (n=A_mail@pool-71-246-11-20.phlapa.fios.verizon.net)
02:09.59*** join/#asterisk wellng (n=welles@61.150.43.113)
02:11.50*** join/#asterisk wellng (n=welles@61.150.43.113)
02:12.23*** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-123.indy.res.rr.com)
02:12.46wellngzoa, r u there?
02:14.41*** join/#asterisk wellng (n=welles@61.150.43.113)
02:19.56*** join/#asterisk ReD-MaN (i=redman@207.210.19.76)
02:20.20PaulHuynhand i do this with asterisk
02:20.51PaulHuynhfax-linksyspap2-asterisk-sipprovider(sixtel)
02:21.04PaulHuynhwhat codec would i have to use
02:22.53Ariel_ulaw
02:23.43PaulHuynhOK anything else i should do for the pap2 setting
02:23.58PaulHuynhor just like i would do for a sip phone
02:24.53PaulHuynhi also need to do this (please let me know if it possible)
02:27.21PaulHuynhtwo pap2=4 different ext-nat-asterisk
02:27.48PaulHuynhdo i need to set anything special or just nat=yes in the extension conf?
02:28.02*** join/#asterisk johnsu01 (n=user@fsf/staff/johnsu01)
02:28.25*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
02:31.52*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
02:37.32XnoNalguien que hable español
02:37.33XnoN?
02:37.47De_Mon!spanish
02:37.57De_Monnope
02:38.19*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
02:38.30De_Mon!es
02:38.42PaulHuynhariel
02:38.48PaulHuynhare u there help
02:38.59PaulHuynhplease
02:39.25XnoNok
02:39.26XnoNsorry
02:39.37XnoNi need so help with asterisk
02:39.51XnoNi would like to take full information about this
02:40.04XnoNmy from is venezuela
02:40.22XnoNi need to make a asterisk server un my local area network
02:40.41QwellYour from is Venezuela?
02:40.46XnoNbut i dont know much about asterisk
02:40.51XnoNyes it is
02:41.01_Sam--babelfish.altavista.com
02:41.02XnoNIm a venezuelan boy friend!
02:41.08XnoNjejejeje ok!
02:41.36De_Monhttp://www.asterisk-es.org/ ?
02:41.42De_Monvoip-info.org
02:42.04_Sam--Qwell:  after seeing a few more GUIs i know why vi is your best friend
02:42.11_Sam--its crazy that people pay for some of that crap
02:42.13XnoNwoao jejeje
02:42.20De_Monvi? vim is better
02:42.21XnoNthanx so much De_Mon
02:42.30De_MonXnoN :)
02:43.00justinujejejeje
02:43.06PaulHuynhcan i do this
02:43.09justinureminds me of that robot on battlestar galatica
02:43.12PaulHuynhtwo pap2=4 different ext-nat-asterisk
02:43.15_Sam--lol
02:43.18PaulHuynhdo i need to set anything special or just nat=yes in the extension conf?
02:43.55XnoNtake a nice night
02:44.05[TK]D-FenderPaulHuynh : You need to set one of EXTERNIP or EXTERNHOST, and LOCALNET in [general] in sip.conf
02:44.28PaulHuynhwhat do you mean?
02:44.41[TK]D-FenderYour * is behind NAT, correct?
02:44.53PaulHuynhNO
02:45.02PaulHuynhasterisk is in public
02:45.16PaulHuynhtwo pap2 will be on a same NAT
02:45.25PaulHuynhconnect to outside asterisk
02:45.32[TK]D-FenderOk, then you should only need to use "qualify=yes" and "nat=yes" for those phones and you should be OK.  Though I would suggest you set a different UDP port for each of them.
02:45.53PaulHuynhoh ok
02:46.20[av]bani\o/
02:46.32_Sam--<PROTECTED>
02:46.39[TK]D-Fender69?
02:46.43_Sam--haha
02:47.44_Sam--[av]bani:  all of the phones for your ISP run over remote gateway?
02:48.22_Sam--or rather, all of your phone service to the outside world is implemented over a remote gw?
02:48.25*** join/#asterisk welles (n=welles@61.150.43.113)
02:48.39sprnova_Sam--... I figured out where ztdummy was hidding... it is not built by default.. so I built it from source..  any idea what is wrong here when I run modprobe.
02:48.55_Sam--what is the error
02:48.58sprnovaasterisk:/usr/src/modules/zaptel# !modpr
02:48.58sprnovamodprobe ztdummy
02:48.58sprnova/lib/modules/2.4.27-2-386/zaptel/zaptel.o: /lib/modules/2.4.27-2-386/zaptel/zaptel.o: unresolved symbol devfs_unregister_R65aaa37a
02:49.26sprnovait goes on like this
02:49.38_Sam--i cant tell ya...but i can tell ya if you get the .deb package from the site i showed you, it will work
02:49.48*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
02:50.03sprnovaI got the .deb source package and rebuilt
02:50.04welleshi _Sam--
02:50.11_Sam--http://rapid.dotsrc.org/unstable/
02:50.17sprnovaoh
02:50.20_Sam--there is a package just for your kernel already
02:50.23sprnovasorry about that.
02:50.26_Sam--there is nothing wrong with wanting to compile it
02:50.27sprnovacool
02:50.28_Sam--it SHOULD work
02:50.34_Sam--but that WILL work :)
02:50.50_Sam--also i still feel you should consider upgrading your asterisk version.
02:51.12PaulHuynhwhere can i get china did?
02:51.26PaulHuynhany help would be great
02:51.29_Sam--i was just reading on the asterisk-biz list its illegal to resell china DID
02:51.46PaulHuynhwhat?
02:51.59PaulHuynhthere go that deal out of the windows
02:52.17PaulHuynhcan we bring US did to china via internet in that case?
02:52.30_Sam--paste coming <sorry pastebin advocaters>
02:52.31_Sam-->> Has anyone ever gotten good quality DID's from China?
02:52.31_Sam-->> Is it illegal for SIP accounts to be sold outside of China?
02:52.31_Sam-->
02:52.31_Sam--> I read on this list a long time ago someone saying that it is illegal
02:52.31_Sam--> to
02:52.33_Sam--> offer DID's from China. From time to time someone sells DIDs, which are
02:52.35_Sam-->
02:52.37_Sam--> vanished after a while - some kind of disposable accounts.
02:52.45PaulHuynhalso can we somehow use ssl with asterisk to secure the call
02:53.07welles_Sam--, meetme list cmd on asterisk can not work .
02:54.00_Sam--welles:  im sorry i cant be much help there, dont use it
02:54.18puppet_sam--: why didnt u use pastebin ><
02:54.30_Sam--not enough action here to worry about it
02:54.35_Sam--it didnt disrupt any conversations
02:54.48puppetwell true but its still annoying when people do get back ;P
02:55.19_Sam--if that is the most annoying thing you deal with all day, i think you've had a good day, hopefully :)
02:57.19*** join/#asterisk freat (n=freat@h-72-244-84-43.chcgilgm.covad.net)
02:58.09*** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca)
02:58.28TelamonWhat does the format_g726 module do?
02:59.12*** join/#asterisk [hC] (n=hardcore@209.153.195.139)
02:59.22sprnova_Samm--... I figured it out.. I am such a Linux newbie..  I compiled the source linked to the 686 kernel source instead of 386 (the one I am running)
03:00.06_Sam--always helps to symlink the right kernel!  at least you are figuring it out
03:00.11*** join/#asterisk justinu (n=Justin@cpe-72-129-86-208.socal.res.rr.com)
03:00.12sprnova_Sam--... I went to that site you told me.. lots of cool stuff up there.. but didn't find the right module version.. so I went back to my source code again.
03:00.30_Sam--i saw the modules for your kernel there eaerlier but whatever works is fine by me
03:01.00sprnovait had the kernel version but I couldn't match the asterisk version (1.0.7)
03:01.12sprnovaearliest was 1.0.9
03:01.32_Sam--what is uname -sr say?
03:01.38_Sam--s/is/does/
03:01.51_Sam--i guess you dont need the s :)
03:01.58sprnovaLinux 2.4.27-2-386
03:02.48_Sam--if you cant get the compiled one to work, i think the other zaptel modules will work fine.
03:02.56sprnovais there a startup file to insure ztdummy is run at next reboot? sorry.
03:02.59_Sam--but in either event, i dont think it is going to solve your complaint
03:03.06sprnovait compiled and loaded ok
03:03.49X-RobOh man 'The IT Crowd' is fucking hilarious.
03:04.13_Sam--i usually make a 'local' file in /etc/init.d that has some modprobes in there for booting
03:04.18_Sam--i dont know how other people do it
03:04.20X-Robsprnova, there's a zaptel.init file you can copy to /etc/init.d
03:04.39sprnovathanks gentlemen
03:04.49X-Robas it's written, create an /etc/sysconfig/zaptel script and put in it 'TELEPHONY=yes' and 'MODULES=ztdummy'
03:05.00X-Rob(or, whatever other modules you're using)
03:05.17Telamonsprnova: What Linux distro are you using?
03:05.18_Sam--then you still have to run update-rc.d
03:06.01_Sam--rob how about we fight about phones instead? :)
03:06.04sprnovaTelamon.. Debian 31r2
03:06.04X-RobHowever, being that he's uding 2.4.27, his timing's going to suck. Someone convince him to upgrade to a kernel released this century.
03:06.17_Sam--i tried to convince him, rob
03:06.35sprnovaX-Rob... ok.. ok...
03:06.37TelamonX-Rob: Yeah, but different distros do things in different ways.  For instance, Gentoo puts all modules in /etc/modules.autoload.d/kernet-2.<whatever>
03:06.57X-RobTelamon, really. Why do I care?
03:06.58sprnovajust got Linux installed today.
03:07.12X-Robsprnova, seriously. Download Asterisk@Home.
03:07.12_Sam--robin_z:  you almost echoed my identical words..."2.4?  how about running something from the 21st century"
03:07.19_Sam--damn nick completion
03:07.19sprnovaits like pulling hair when you are used to *BSD
03:07.30TelamonX-Rob: *You* don't, but sprnova might.  If he's not using a RedHat distro, it might not use the sysconfig dir.
03:07.32*** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net)
03:07.47_Sam--<i odnt have problems with 2.4 kernels...but for a new install...>
03:07.50X-RobTelamon, that zaptel.init looks in /etc/sysconfig/zaptel.
03:07.57X-RobThat's why I said put it there.
03:07.57_Sam--i still have 2 out 5 boxes on 2.4
03:08.49_Sam--Telamon: XratedRob knows all.
03:09.42X-RobIt's actually X for X-Chat. I used to have 'Rob', then I used to fire up X-Rob on linux. Then I never used 'Rob' for ages, and it was stolen.
03:09.48TelamonHeh, my apologies for doubting you, X-Rob. :)
03:09.51_Sam--he has built in polygraphic abilities...he knew when i was lying, and he is 8000 miles away :)
03:10.36X-Rob_Sam--, nfi what you're talking about, but I'm guessing you said something blatantly wrong, and I was in a bad mood.
03:10.46_Sam--nah, you and i went at it over the GXPs :)
03:10.50X-RobAaaah yeah
03:11.04X-RobThey're not bad now at all.
03:11.10X-Rob.2.9 is going _very_ nicely.
03:11.15TelamonOkay, here's a test for you: Why doesn't g726 work with my GXP-2000? 1.0.1.13 firmware.
03:11.17X-Robstill has spastic screen on the original batch
03:11.18_Sam--they're not good, but they're just not as bad
03:11.38_Sam--robin_z:  how about you let me try the 2.9 and let you know how it works here? :)
03:11.43_Sam--dammit, nick completion
03:12.04TelamonThey are very good for the price.  Of course they cost half of what any other VOIP business phone costs, so that might not be saying much... :)
03:12.40X-RobTelamon, Is that like a riddle? I don't know, why DOESN'T g729 work with your gxp?
03:12.43X-RobI say I say I say
03:12.46_Sam--hahah
03:12.58_Sam--Telamon:  why dont you try the latest firmware
03:13.01_Sam--see if it fixes your prob
03:13.21_Sam--in my opinion, its a lot better than the .13 you're on
03:13.24X-RobOoh, he said 726. Ikky.
03:13.57_Sam--X-Rob:  where'd you get 29
03:13.58_Sam--?
03:14.02*** join/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com)
03:14.20_Sam--1.0.2.9 that is
03:14.30TelamonX-Rob: Heh, no, legitimate question.  uLAW and GSM work fine, but with g726 I get a weird dialtone (staticky) and no audio when calling the PBX.
03:14.38_Sam--i thought maybe the german place had it online
03:14.38X-Rob_Sam--, from Richard Huang@grandstream
03:14.59X-RobTelamon, it's probably broken. 726 is useless. Here's a hint:
03:15.04X-Robhttp://kvin.lv/pub/Linux/Asterisk/
03:15.06TelamonI'm a little leary of going to a 1.0.2 since you can't go back to 1.0.1 after the upgrade.
03:15.19_Sam--there is nothing in the newer firmware that would make you want to go back.
03:15.24X-RobYou want the 'icc' version, as they're faster.
03:15.26_Sam--as long as you have a decent mac address
03:15.44PaulHuynhso which codec do you have to paid digium licensed?
03:15.53X-RobOf course, that's only legal to use if you live in a country without software patents.
03:15.54_Sam--i run the new firmware in 2 production environments...total of about 40 gxps
03:16.04Telamon"Decent mac address"?  That referring to phone revision or something?
03:16.12_Sam--that is referring to the mac address
03:16.19_Sam--you know, macintosh
03:16.21X-RobTelamon, the first 5000 gxp's used slightly different hardware.
03:16.33X-RobThis causes the screen to go spastic with the 1.2 firmware
03:16.36_Sam--if you go to the gxp web status page....
03:16.39_Sam--and click on the phone status page
03:16.41*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
03:16.42_Sam--you see the mac address
03:17.04_Sam--that wasnt well worded
03:17.12_Sam--if you log into your gxp 2000 via the web interface...
03:17.14X-Rob000b8203a0e1
03:17.15_Sam--and click on the phone status page.
03:17.18TelamonX-Rob: Yeah, that's the problem.  I live in Canada, I don't think we can use 729 without paying license fees.  I'm looking for something better than GSM voice quality, but with some compression.  64kB/sec is a little high for phone if you aren't on a LAN.
03:17.19X-Rob^^^ bad mac
03:17.20_Sam--you can see your mac address.
03:17.34X-RobTelamon, well pay the $10 then. It's worth it
03:17.35_Sam--ulaw is 83kbps
03:17.40_Sam--according to qwell.
03:17.44_Sam--64 + overhead
03:17.51X-RobIt's actually best to pay the $10, don't bother with the digium codec, and use the icc one on that page. It's better8)
03:18.02*** join/#asterisk fdask (i=fdask@CPE0013d479c929-CM0011e6edd218.cpe.net.cable.rogers.com)
03:18.14X-Robor speak to trixter on here, he's got a new licencing scheme happening
03:18.36_Sam--rob does 2.9 fix anything that was b0rked in 8?
03:18.42_Sam--or?  give me some clues
03:18.49X-RobMy screen stays less broken for longer
03:18.54TelamonX-Rob: The licensing fee is only $10 for the whole server?  Hell, might as well...
03:18.54_Sam--lol
03:18.59X-RobTelamon, $10 per channel
03:19.12welles_Sam--, i want  two asterisk work together. they can now .when a call to asterisk A then asterisk A call Asterisk B. the call will continute 40 seconds and then hangup. why?
03:19.16X-Rob_Sam--, 2.8 is fine for most people, I've just been bitching about the screens 8)
03:19.26TelamonAh, that's a bit more expensive...  Still not bad though.
03:20.27_Sam--welles:  i have no idea.....is it exactly 40 seconds every single time?
03:20.39wellesyes
03:20.45_Sam--is it a timeout rule somewhere?
03:20.56wellesi use mp3player app
03:21.39_Sam--X-Rob:  you use hints/blf on the 7 buttons on the right side of the GXP?
03:21.47X-Rob_Sam--, no, not all of 'em
03:21.51X-RobI use snoms for hints
03:22.02_Sam--i had problems with the gxp using hints on *
03:22.11_Sam--* segfaulted
03:22.11X-Robin fact, I haven't got any BLF's configured on the GXP's at the moment
03:22.17X-RobOoh really. That's not good.
03:22.17_Sam--like to see if you could duplicate sometime
03:22.23X-RobWill try now
03:22.26_Sam--what * you have?
03:22.31X-RobI'm using 1.2-svn
03:22.37X-Robzaptel-trunk, libpri-trunk
03:22.38_Sam--it only happens when there are many gxps doing stuuff, and having the gxps show the lights
03:22.44_Sam--you cant breakt it with one single gxp
03:23.05[av]banisam breaks gxps
03:23.08_Sam--i had 10 extensions setup in asterisk to hint
03:23.14X-Robah. Well I've got a machine with 600 subscriptions with the snom-pickup patch, and that only segfaults due to bugs in the snom code
03:23.15[av]banifor fun and profit
03:23.21_Sam--and the gxps (10) were setup to use BLF
03:23.44_Sam--[av]bani :  rob has 1.0.2.9!
03:23.51X-RobWhen it crashes next, give me a backtrace
03:23.53_Sam--but apparently it doesnt fix much
03:24.01_Sam--i have the cores
03:24.08_Sam--it crashed twice in one day, and i havent tried it since
03:24.12[av]bani_Sam--: gimme
03:24.16_Sam--after i took the hints out / turned blf off...
03:24.21_Sam--havent crashed since
03:24.24X-RobSo you haven't recompiled asterisk yet?
03:24.35_Sam--i am running 1.2.4 release version
03:24.45[av]baniactually, i'd like to see BLA support for gxp/*
03:24.56[av]baniso our gxp's would work more like traditional pbx
03:25.00X-Robgdb /tmp/core.asd /usr/sbin/asterisk
03:25.09X-Rob[av]bani, you mean like a key system
03:25.15X-Robit won't happen for a while.
03:25.23_Sam--hold on, i have some pastebins of the output from before from gdb
03:25.34[av]banino, it's apparently on the list for next * release
03:25.50[av]banisince polycom suffers from issues related to buddy lists
03:25.52X-RobBLA? Busy LINE Activity?
03:25.59[av]baniand BLA is the only way to work around it
03:26.04_Sam--here's some of the gdb output:
03:26.06_Sam--http://sam.pastebin.com/545854
03:26.55[av]banii'm sure one might be able to fake it using Pickup() and some scripting
03:27.03TelamonIt would be nice if BLF didn't trigger when a phone went offline (ie, unplugged) but only when in use.
03:27.06X-RobWhat do you mean by BLA?
03:27.20[av]baniX-Rob: http://lists.digium.com/pipermail/asterisk-users/2006-February/146983.html
03:27.51*** join/#asterisk Cresl1n (n=matt@146.229.191.68)
03:27.52X-RobAh, fair enough
03:27.56X-RobCrappy polycom problem 8)
03:28.03[av]baniwell, its a generic issue
03:28.11X-RobNot really, BLF works fine for everyone else
03:28.20X-Robcall pickup, ringing/inuse, works fine.
03:28.30[av]banisorta, it doesnt let you do shared lines which is what one really wants
03:28.53[av]baniline status great, but i want to punch the line button and pick up that extension
03:29.01X-RobWhich, when you look at how there's no conecpt of a line _anywhere_ in asterisk, is gunna be damn hard to do.
03:29.06X-RobHang on
03:29.13X-Robpunch the _line_ button and pickup the _extension_?
03:29.18X-Robhow the hell are you planning on doing that?
03:29.24[av]baniquit being pedantic, you know what i mean
03:29.33_Sam--pedantic = y?
03:29.36X-Robpush the extension button on the GXP and it dials **xtn, use Pickup for that
03:29.50X-Robfor snoms, they just do a reinvite and it just works
03:30.07Telamonpednaic = overly literal.
03:30.27X-RobNo, it's different. You want shared lines (eg, 'Joe, call on line 3') or extension status?
03:30.39X-Roband extension pickup
03:30.39[av]banireinvite fine, what about joining an active line
03:30.53X-Rob[av]bani, barge-in?
03:30.59[av]baniyes
03:31.09X-RobThat's harder than you think
03:31.12[av]banithe current 'solution' would be to throw everything into meetme rooms
03:31.22TelamonX-Rob: I think he wants the incoming call to go to a queue, the line lights monitor that queue and allow you to take the call.
03:31.26*** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
03:31.27X-RobThere's 14000 lines of code to do conferencing.
03:31.37X-Robbut 'a lot'
03:31.57websaewhat do i add in my dial plan in order to be able to dial another person extension for example, i am 100, and want to dial 102...anyone :)
03:31.58[av]baniX-Rob: shared lines, as in extension 6800 exists on 12 phones
03:32.00X-Robbarge-in is conferencing, it's not the same as just splicing two wires in to the conversaion as you would with an analog pbx
03:32.19[av]banithe polycoms do it i guess
03:32.32[av]bani* does not yet
03:32.39X-Rob[av]bani, 6800,1,Dial(SIP/6801&SIP/6802&SIP6803)... but I see what you mean
03:32.51Darwin35poloy want a hand job swqak
03:33.03Telamonwebsae: <exten>,1,Dial(SIP/<username>)
03:33.03[av]banii think snom supports it also
03:33.06X-Robwebsae, 'exten => 102,1,Dial(SIP/102)'
03:33.12[av]banigxp "plans to"
03:33.14websaethanks
03:33.26EgonisI just re-compiled my kernel w/ CRC_blah = y, which has resolved some symbols.. however modprobing zaptel results in: Unknown symbol class_simple_device_add class_simple_destroy class_simple_device_remove class_simple_create -- I am using udev, coldplug, hotplug, and kernel-2.6.15-r1
03:33.48[av]baniwonder what the 'ext' is for on the back of the gxp2k... sidecar?
03:33.55X-Rob[av]bani, yup
03:34.02X-Robit's a 9600 serial link
03:34.19[av]banisame as the snom then
03:34.22X-Robyup
03:34.28_Sam--thats an svideo cable
03:34.38_Sam--in case you want to watch xml on your monitor
03:34.42X-Robalthough I think the snoms use a slower speed.
03:34.42[av]banisam yea! HBO !
03:34.46_Sam--hahah
03:35.00[av]banicombined with xml support = megapr0n ph0ne
03:35.42_Sam--[av]bani:  the only way to show queue status on the gxp would be with xml minibrow?
03:35.58johnsu01I'm just starting to get asterisk set up, trying to use kphone. When I try to make the test call my dialing 500, I lose audio after dialing 5. But from the console, it looks like the call is still going through.
03:36.08johnsu01s/my/by/
03:36.20johnsu01oh. what a bot.
03:36.53[av]baniyah, dunno what we'd do without it
03:37.18justinuspend time with the wife?
03:37.19justinulol
03:37.24*** join/#asterisk Rhizome (n=rhizome@tor/session/x-cf1330eddd772505)
03:37.26_Sam--hmmm that reminds me
03:37.30johnsu01I don't need the NAT settings on when both kphone and the asterisk server are on the same computer, do I?
03:37.52gaupenope
03:38.19gaupeanybody know how many characters the Snom 360 can show as callerid?
03:38.29X-Robnot enough 8)
03:38.37gaupehehe
03:39.17X-RobI set it to Number Only, otherwise it tends to go 004[02077155 0402077]155
03:39.20_Sam--X-Rob:  can you tell your grandstream pals to check out our wiki page and all the bugs:  http://www.voip-info.org/tiki-index.php?page=GXP-2000
03:39.25_Sam--bani found each and every bug himself
03:39.27X-Rob_Sam--, they have been
03:39.50[av]banihopefully they fixed the fecking speakerphone/handset volume
03:39.54X-RobThey're not interested in open source tho
03:40.03X-RobI've been asking them that since they _released_ the gxp
03:40.20[av]banithey also seem to have dropped ilbc
03:40.23X-Robbut, being that I was the one who promoted the pa1688 project, they prolly think of me as a bit biased.
03:40.38X-RobAnyway
03:40.42_Sam--how did the linksys wrt54 become open source?
03:40.49[av]bani_Sam--: the law required it
03:40.52X-Rob_Sam--, it always was linux based.
03:40.59X-RobAnyway. I'm doing some AMP stuff.
03:41.57[av]baniwell, grandstream purchased an echo can so that might make it hard to opensores it
03:42.26[av]banithough they could just hardcode it into the flash and then the opensores would just link when it loads
03:42.26_Sam--there would be no financial benefit to them to do so.
03:42.39[av]baniwell, it would mean us end users could fix the fecking bugs
03:42.46[av]banisince grandstream cant :/
03:42.47_Sam--that doesnt make them more money
03:43.13_Sam--and if it were open sourced, then in theory, people could take similar code and try to make similar phones?
03:43.13[av]banigrandstream doesnt sell software
03:43.25[av]banithey already can, source or not
03:43.34[av]banigrandstream uses a reference design
03:43.39gaupethey have probably signed NDAs for the DSPs and such
03:43.47[av]baninope
03:43.53[av]baniit all off the shelf, anyone can buy it
03:44.04gaupeok
03:44.15_Sam--i dont think there is any benefit to them open sourcing it...but maybe if it were, more people would buy the phones.
03:44.18gaupewell, buy it and build it :)
03:44.23_Sam--the benefit is to the users
03:44.28_Sam--which obviously come second
03:45.05_Sam--as a company, in my own opinion, they dont gain anything if they were to release it.
03:45.12[av]baniwell, i guess you would have to ask aredfox why they open sourced pa168, since it was obviously a dumb decision
03:45.18_Sam--they would ulimately gain bttter firmware
03:45.25*** join/#asterisk apardo (n=apardo@87.218.45.191)
03:45.48[av]baniwell, end users contributed stuff to aredfox for pa1688
03:46.04[av]banilocalized IVR recordings, ilbc, bugfixes, etc
03:46.13_Sam--the pa1688 is a dsp or what is it?
03:46.40[av]baniits an integrated hardware platform for phones built around an 8051 (8bit microcontroller) design
03:47.12_Sam--how come it looks like no decent phones use that platform?
03:47.32SkramXin a dial command, i want to go to extension 999.. then have it wait, and insert the correct password (dtmf tones) for me,... can I do this?
03:48.03X-RobSkramX, 'show application dial' on the asterisk console
03:48.06[av]banihttp://www.soyogroup.com/products/proddesc.php?id=307
03:48.13[av]banishrug.. doesnt look any shitier than gxp :)
03:48.21SkramXX-Rob: right.. any particular commands or whay
03:48.21TelamonSkramX: http://www.asteriskguru.com/tutorials/authenticate.html
03:48.26SkramXTelamon: okay..
03:48.47SkramXbtu see, i want the dial command or i want asterisk to insert dtmf while a call is in progress
03:48.53_Sam--that thing doesnt look near as fucntional as gxp
03:48.57_Sam--not as many buttons
03:49.22_Sam--in fact, i crap on that soyo phone!
03:49.32[av]bania phone.. for me to poop on?
03:49.35_Sam--lol
03:49.47_Sam--it looks like something that belongs in my moms house in her kitchen or something
03:50.22websaewhere's a good place to buy DIDs?
03:50.26websaeanyone have any suggestions?
03:50.35_Sam--dude you just figured out how to dial an extension
03:50.38websaeI tried didx, but never got an email to confirm my account
03:50.56_Sam--the chances of you successfully setting up a did are not great.
03:50.57TelamonHmm, not sure I know what you mean.  You want when someone dials extention 999 it dials out and enters a password?
03:51.06_Sam--if you could just now setup a way to dial extension 102
03:51.11[av]bani_Sam--: http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a0080159de2.html
03:51.30websaelol
03:51.44[av]bani_Sam--: isnt that just the bestest phone evar?
03:51.44_Sam--websae :  i think you can do it, just busting your balls.
03:52.06[av]banihttp://www.reseaudirect.com/catalog/images/Cisco%20IpPhone%207902G%20G.jpg
03:52.12websaewhere is a good place to buy DIDs?
03:52.16[av]baniand its CISCO!!1!
03:52.22websaethat's all i am asking
03:53.02_Sam--i need to get a high end phone for my own desk
03:53.11Telamonwebsae: Checkout http://voip-info.org/wiki/view/DID it lists a bunch of places.  I get mine from our phone company, so I don't have any opionions on the matter.
03:53.13[av]baniwhats high end mean
03:53.17[av]banistilts?
03:53.38_Sam--i HAVE to have an xml minibrowser :)
03:53.47[av]baniwhats your budget?
03:53.53_Sam--unlim.
03:53.58_Sam--its for my own desk
03:54.01[av]bani$385?
03:54.05_Sam--sure
03:54.06websaedo you have a PRI telamon?
03:54.10[av]banicisco 7970g it is
03:54.14Telamonwebsae: Yes.
03:54.22websaewhere are you located?
03:54.38Telamonwebsae: Prince Edward Island.  Eastern end of Canada.
03:54.40_Sam--what do you do with the pretty color screen?
03:54.44_Sam--display a jpg of your wife?
03:54.45tronix_Sam--: if it's unlimited, go for the 7985. ;)
03:55.15[av]bani_Sam--: i thought you would know by now
03:55.17tronixthough more seriously, 7970G sounds pretty nice. I like my 7960G but sure would be nice to have nicer contrast with color
03:55.22[av]bani_Sam--: comeon, you can figure it out
03:55.33[av]banistarts with p, ends with n and has a 0 in it
03:55.37_Sam--my light bulb is not buring at this hour
03:55.55[av]banitronix: backlight
03:56.06[av]banii would get a 7960g if it had a backlight
03:56.20*** join/#asterisk wellng (n=welles@61.150.43.113)
03:56.22_Sam--that screen on the 7970 is a touch screen?
03:56.26[av]baniyes
03:56.33_Sam--how do you program the on screen items?
03:56.41*** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net)
03:56.45[av]banihttp://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a00801c9638.html
03:56.48[av]banixml
03:57.00[av]banitouch screen pr0n!
03:57.21[TK]D-FenderPlug & play!
03:58.36johnsu01It seems like it's probably some kind of dtmf setting, but I've followed the dtmf settings on the wiki for kphone...
03:59.05*** join/#asterisk wellng (n=welles@61.150.43.113)
03:59.40_Sam--[av]bani :  that phones does SIP off the shelf?
04:01.23*** join/#asterisk wellng (n=welles@61.150.43.113)
04:02.08orlockDoes anybody know what the grandstream's ext port is for?
04:02.12orlockis it serial?
04:02.25_Sam--it hooks up to a dvd player to record calls
04:02.33_Sam--svideo
04:02.40_Sam--<just kidding>
04:03.33tuxinator_linuxDoes _Sam-- stand for Smart Ass Man?
04:03.45tuxinator_linux;-)
04:03.46[av]bani_Sam--: no, 7970g is sccp only
04:03.55[av]bani_Sam--: but Qwell swears by it
04:04.06tuxinator_linuxYes he does
04:04.18[av]baniso if you get a 7970g then you get 24/7 support from Qwell
04:04.23_Sam--lol
04:04.31tuxinator_linuxAsterisk is suppose to play pretty well with sccp now
04:04.36_Sam--sorry to be a dumbass, but how do yo do get SIP , or how do you make linux speak sccp?
04:04.59tuxinator_linuxyou need a smartnet agreement with Cisco either way
04:05.12tuxinator_linuxchan_sccp for asterisk
04:05.18_Sam--i see it now
04:05.23tuxinator_linuxSIP image from cisco for phone
04:05.38_Sam--if you used chan_sccp, you would have sccp.conf?
04:05.45tuxinator_linuxI ordered my smartnet a while ago, still waiting for access
04:06.12tuxinator_linuxI think I will be trying both, SIP and SCCP
04:06.14[av]banithere is no sip for 7970
04:06.20tuxinator_linuxnope
04:06.32[av]baniyou dont need smartnet
04:06.46tuxinator_linux[av]bani: Do tell
04:07.20tuxinator_linuxwell, I suppose you don't need it if you're doing chan_sccp
04:07.28[av]banituxinator_linux: and since there is no sip for 7970...
04:07.50tuxinator_linuxbut receiving updats is kinda nice, not required though
04:08.07[av]baniyeah, maybe a couple years ago it might ahve been an issue
04:08.14_Sam--thanks for all the info...it put me right to sleep....night.
04:08.23tuxinator_linuxnight
04:10.08*** join/#asterisk wellng (n=welles@61.150.43.113)
04:13.43*** join/#asterisk wellng (n=welles@61.150.43.113)
04:13.46Nuggetwhat's the state of chan_sccp these days?  is it viable?
04:13.56*** join/#asterisk wellng (n=welles@61.150.43.113)
04:13.56[av]banii guess its usable
04:15.18orlockHmmm
04:15.18*** join/#asterisk blackgecko (n=blackgec@201.144.217.221)
04:15.19*** join/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)
04:15.38orlocki'm trying to connect to upstream sipserver with a grandstream, and its not registereing.. any suggestions?
04:16.27*** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com)
04:16.47EgonisWhen loading zaptel, I get the message 'unable to open master device '/dev/zap/ctl' although the module is loading
04:17.20mogormanyou probably dont have udev rules correct
04:17.55Egonismogorman: How do I check that? :)
04:18.37X-Robor /etc/zaptel.conf isn't set up right.
04:18.39mogormanwell is there /dev/zapctl or /dev/zap/ctl
04:18.46Egonismogorman: nope
04:19.01QwellNugget: it works really well
04:19.02EgonisX-Rob, mogorman: /dev/udev/rules.d/10-zaptel.rules exists
04:19.12blackgeckoanyone has experience using a2billing ???
04:19.32mogormanwhat card do you have?
04:19.59*** join/#asterisk emergion (n=pauly@84.133.233.220.exetel.com.au)
04:20.17Egonismogorman: Sangoma A200, haven't started w/ that just yet tho
04:20.52mogormanoohhh....
04:20.57emergionHello all, Just wondering how many calls can a DID Simultaneously handle a VoipSP said pretty much unlimited but wouldnt it give an engaged signale when a call is in process
04:20.59mogormanwell i imagine you might have some wanpipe issue
04:21.10Egonismogorman: I haven't loaded that yet... so probably
04:21.14mogormanbut /me only really knows digium stuff
04:21.19Egonismogorman: just trying to get zaptel loaded properly first
04:21.31mogormanwell you can modprobe zaptel
04:21.56*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
04:22.01Egonismogorman: did that, it's the init.d/zaptel which complains about no /dev/zap/blah stuff, but the module loads fine, just doesn't create any devices, at all
04:22.34mogormanwell its probably not loading properly
04:22.38mogormandmesg will tell you why
04:22.45Egonismogorman: no complaints in dmesg at all about it
04:22.55MikeJ[Laptop]hello Mr. O...
04:23.00mogormanwell id be happy to take a look for ya
04:23.05mogormanhello Mr. J
04:23.20JunK-Yhey mogorman !
04:23.31mogormanand Mr. Y
04:23.43[av]bania200, oooo
04:23.46[av]banishiny
04:23.47JunK-YDrunK-Y!
04:24.00MikeJ[Laptop]heh
04:24.02Egonismogorman: it simply says -- Zapata Telephony Interface Registered on Major 196, nothing more
04:24.55mogormanhmm well it should be there then
04:25.03Egonismy point exactly.. :)
04:25.19mogormanwell i can look at it for you
04:25.21Egonismaybe I'll try re-emerging udev, just for kicks
04:26.31MikeJ[Laptop]Egonis, string and cans?
04:26.36mogormanreverse ssh?
04:26.56EgonisI suppose
04:27.00EgonisMikeJ[Laptop]: Not a bad idea
04:27.02mogormanlol
04:27.20mogormanyou are silly, sorry i cant help you
04:27.27Egonismogorman: I just upgraded udev, do I need to tell udev.conf that I have a 10-zaptel.rules file?
04:27.38mogormanyou just need the rules
04:27.39Egonisyes, yes I am
04:27.42mogormanand it should make it
04:27.52mogormanand it shouldnt matter
04:27.52Egonismogorman: So having the file makes it aware of the rules, right?
04:28.00mogormanas gentoo has had it in their rules for a few months now
04:28.07mogormanat least the digium cards
04:28.12mogormanso zaptel should be there too
04:28.36Egonisokay, time for a reboot to refresh udev, and other broken toys
04:28.38Egonisbrb
04:28.59*** join/#asterisk L|NUX (n=linux@202.5.145.57)
04:32.04X-Robhe could have just restarted udevd
04:32.38mogormani know that
04:32.39*** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com)
04:32.42mogormanand you know that
04:32.45Egonisno dice.. :(
04:32.46mogormanbut he did not know that
04:33.27Egoniswell, the zaptel module is loaded! :) but no devices in /dev.... should I disable hotplug?
04:33.38mogormanno it should make em
04:33.44mogormanyou could manually make dev nodes yourself
04:33.47mogormanbut that is lame
04:33.50X-Robnot with udev
04:33.53X-Robudev does it all
04:33.56mogormanyes you can
04:34.03mogormanyou can make dev nodes amongst the udev tree
04:34.05mogormanits not correct
04:34.09mogormanbut you can do it
04:34.19[av]banianyone used the zoom 5801 yet?
04:34.41EgonisX-Rob: I'm using udev
04:34.58EgonisX-Rob: but it's being gay
04:35.32mogormanhow can udev be gay????
04:35.55EgonisI would love to make the dev nodes just to shut it up, but I also have my sangoma card to get working (wanrouter) -- udev is being gay because it's wearing a scarf
04:36.22mogorman????
04:36.25Egonislol
04:36.36EgonisI have no idea.. :)
04:38.26Egonisso what do you suggest? manually create dev nodes? how do I do so/
04:38.36mogormani would not suggest that
04:38.56Egonisso where should I start with troubleshooting udev/coldplug/hotplug
04:38.57Egonislol
04:39.00Egonisyeah, that too
04:39.13Egonisplease excuse my nub-ness
04:39.23Egonisbut learnin this will help me show others
04:39.30Egonislearnin(g)
04:40.10mogormanwell the thing is
04:40.15mogormanthere is not much to screw up
04:40.24mogormanit should just work TM
04:40.43Egoniswhich is what really gets me
04:42.02EgonisI'm running a fresh as pie gentoo install, I've installed gentoo like 50 times on 50 different servers... this is my first time setting up a box w/ a sangoma card... but I haven't even started with that part yet
04:43.18mogormanahh canadia
04:46.05mogormanyou dont refer to your home land as canadia?
04:46.06[TK]D-FenderEgonis : Did you follow the zaptel compile reqs for it?
04:48.14*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
04:49.08mogormanwell he can load it [TK]D-Fender
04:49.11mogormanso its built
04:50.24[TK]D-FenderNot sure if it implies he did the POST WanPipe recompile....
04:50.37mogormanno he hasnt
04:50.43mogormanbut he is just trying to load zaptel
04:50.44*** join/#asterisk sonicGB- (n=Miranda@138.25.71.101)
04:51.29[TK]D-FenderI'm wondering if its being "incomplete" from the process might interfere with it loading...
04:51.32[TK]D-Fenderjust a though
04:52.08*** join/#asterisk BingoPajama (n=zcw100@68-233-16-90.stcgpa.adelphia.net)
04:52.30mogormanshouldnt
04:53.26sonicGB-I have hw_ip_phone <-----SIP----> asterisk(1) <-----IAX2-----> asterisk(2) working fine. I want to dial a number on hw_ip_phone and have asterisk(1) pass the call to asterisk(2) (that much works) and have asterisk(2) Dial(SIP/username@host,,r).  asterisk(2) says "Unable to create channel of type 'SIP' (cause 3 - No route to destination)". I've googled/faq'ed, but no clear answer as to what I'm doi
04:53.26sonicGB-ng wrong. Any suggestions please?
04:53.37BingoPajamaAnyone know were I can get a copy of the cisco 7912 firmware?
04:54.29mogormandont you buy it from cisco?
04:55.34BingoPajamaYou'd think that cisco would make it easy to take your money but they don't.
04:55.56sonicGB-It's been a long time since I played with anything Cisco, but IIRC, for the routers/switches at least, there's a 'tac' web site that has firmware for everything for download. Catch is that you need to authenticate to get on to that site, and that you have to get from Cisco (which I always did via  a third party reseller)
04:55.57Egonis[TK]D-Fender: yes
04:56.18Egonis[TK]D-Fender: nope, didn't touch wanpipe yet... I want zaptel working 100% first
04:56.52EgonisI figure that if the devnodes aren't created w/ zaptel, why would they be w/ wanpipe?
04:57.51justinuum, zaptel won't work without wanpipe
04:57.58mogorman?????
04:58.07mogormanjustinu that is the dumbest thing i have heard
04:58.09mogormantonight
04:58.09[TK]D-FenderEgonis : You need to follow the procedure for it.  Zaptel first, then Wanpipe, THEN ZAptel AGAIN
04:58.10justinuwanpipe is the foundation layer that creates the zaptel look alike
04:58.17justinuit's an abstraction layer ;)
04:58.26mogormanit doesnt matter, he is just trying to load zaptel
04:58.30mogormannot anything else
04:58.36Egonis[TK]D-Fender: emerge zaptel a second time, you mean?
04:59.10Egonis[TK]D-Fender: I haven't even touched wanpipe yet.. like, I didn't untar the wanpipe .tgz
04:59.39*** join/#asterisk masked (n=masked@static-203-87-16-192.vic.chariot.net.au)
04:59.41maskedhi
04:59.46[TK]D-Fendernot emerge... recompile.  You should do it from source.  Wanpipe patches Zaptel in order to function.
05:00.04maskedhas anyone here used telstra call waiting via a fxo?
05:00.23BingoPajamaYa, I asked my university departments admin. Maybe he'll get it to me maybe he won't.
05:00.30maskedX-Rob maybe?
05:00.43justinudude, you need to get the winpipe stuff installed
05:00.54justinuit needs the zaptel source, then it patches it
05:00.57justinuthen it installs it
05:01.04justinuit's like automagic
05:01.10BingoPajamaAnyone have any luck building chan_bluetooth?
05:01.11[TK]D-Fenderyup
05:01.30mogormanmust resist urge to troll
05:01.53justinuit's actually a very slick setup
05:02.03justinuconsidering their products work with other open source pbx softwares ;)
05:02.09Egonisjust a thought, I will try -devfs26 in package.use, just for kicks
05:02.31justinuegonis, you can continue down that line as long as you want... without wanpipe, it won't help you.
05:02.33Egonisdo I need bri?
05:03.02Egonisjustinu: Yes, however... if it doesn't create device nodes for zaptel... like, it doesn't have the ability -- where will I be when I try to do wanpipe?
05:03.17justinuwanpipe is a layer under zaptel
05:03.19*** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net)
05:03.58X-Robmasked, it's a pain in the arse, sending a flash.. I've never done it, but it is possible.
05:04.10EgonisFYI: Setting -devfs26 in package.use breaks zaptel... it now says: Unknown symbol -- class_simple_device_add, etc
05:04.34mogormanmost people Egonis build everything asterisk related straight from source
05:04.43mogormandont use gentoo debian redhat packages
05:04.50*** join/#asterisk xyklopz (n=xyklopx@216-91-89-21.biltmorecomm.com)
05:04.51Egonismogorman: maybe I should go that route
05:05.03X-Robkkkkkkk
05:05.09X-Robwups, vi'ing in the wrong window
05:05.31xyklopzcan someone clarify something for me ... if I have a call coming in via a SIP softphone, the SIP channel is used by I don't quite understand the relationship to the contexts ...
05:05.56*** part/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com)
05:06.06xyklopzwhere do I define which context is for say local extensions versus calls which must be proxied to a 3rd-party voip provider
05:06.23IronHelixyou make the call from the softfone
05:06.29IronHelixthe call goes in whatever context the phone is in
05:06.34IronHelixfrom there its matched by what you dial
05:07.08xyklopz"whatever context the phone is in" => meaning sip vs h.323 vs iax?
05:07.17IronHelixno
05:07.19IronHelixsip is a protocol
05:07.23IronHelixa signalling method
05:07.24xyklopzright
05:07.26xyklopzi know
05:07.28xyklopzclear text
05:07.29xyklopzINVITE
05:07.31IronHelixas is 323 and iax
05:07.31xyklopzREGISTER
05:07.32xyklopzetc
05:07.37IronHelixexactly, along with RTP for audio
05:07.50IronHelixin sip.conf, there is for each phone a context=
05:07.53xyklopzI'm actually researching a method of active RTP injections
05:08.05IronHelixthat tells you waht [context] in extensions.conf that phone is in
05:08.19IronHelixa context is a group of exten's (things you can dial) as well as any other included contexts
05:08.29IronHelixextens can include patterns and other such things
05:08.37sprnovahaving problems setting up Music on hold.. I get a bunch of weird sounds.. I think it is playing but it sounds all wierd.. how can I tell if the zfdummy driver is being used?? I loaded it via modprobe and restarted asterisk.
05:09.21IronHelixie exten => _1NXXNXXXXXX,1,Dial(SIP/myVOIPprovider/${EXTEN}) would match any 11 digit (american long distance) number and dial it out on your sip trunk
05:10.06xyklopzI don't quite understand the concept of a "trunk"
05:10.21IronHelixtrunk is a channel that can carry more than one call at once
05:10.31IronHelixin this case i mean it as what you connect to the pstn with
05:10.49IronHelixin the above example, your account with myVOIPprovider is you 'sip trunk'
05:10.56xyklopzi c
05:11.25xyklopzhow do you associate incoming trunk calls then?
05:11.35xyklopzdo they use the default context
05:11.45xyklopzor is that specified when you configure the trunk
05:12.08IronHelixyou can use a dial prefix to connect to another voip provider, for example:  exten => _91NXXNXXXXXX,1,Dial(SIP/myOTHERbetterVOIPprovider/${EXTEN:1})  exten:1 strips off the 0
05:12.25IronHelixasterisk does not see much of any difference between your softphone and your voip provider
05:12.35IronHelixthey are both sip channels to *
05:13.02IronHelixso the context thing still applies-  you just want the incoming calls in a different context
05:13.24IronHelixdoes that make sense?
05:13.43IronHelixyou can use the exten 's' for where no pattern can be matched
05:13.45IronHelixexample:
05:13.57IronHelixexten => s,1,Playback(welcome)
05:14.05IronHelixexten => s,2,Dial(SIP/1234)
05:14.12IronHelixexten => s,3,Voicemail(1234)
05:14.33IronHelixtoss that in a context, and associate a sip channel that might get an incoming call to that context
05:15.08IronHelixwhenever a call comes in from that account/provider/etc, it will start by playing a welcome message, then try to send the call to SIP/1234 (defined in sip.conf) then go to voicmeail
05:15.20IronHelixminor typo- it should actually be Dial(SIP/1234,20)
05:15.27IronHelixthe ,20 says to only ring 20sec
05:15.33IronHelixsorry if im overloading or not helping :\
05:17.07xyklopzbut okay, how do I say dial * from a softphone
05:17.22xyklopzw/o dialing an extension... or am I looking at it wrong?
05:17.27xyklopzI can't just type the IP ...
05:17.41xyklopzdo I setup an extension = my phone #?
05:17.54IronHelixyeah you are... assuming the softphone is registered to * and stuff, it has no need to dial * because whatever it dials is through *
05:17.54xyklopzNXXXXXXXXX
05:18.16IronHelixhow bout this
05:18.18xyklopzbut that's how you would get it to be able to dial * directly
05:18.32IronHelixbut what does * do when you dial it?
05:18.42IronHelixthats what the context/extension thing is
05:19.06IronHelixasterisk wont on its own answer your call and wait to be told what to do
05:19.16IronHelixthere has to already be a path for your call in place
05:19.18IronHelixhow bout this
05:19.23IronHelixtell me exactly what you want to setup
05:19.27IronHelixand i'll tell you what you need to do
05:20.42xyklopzI have a few softphones (future wifi phones) that I want to setup with individual extensions & voicemail (etc. etc.). after I get this configured and I understand it better, I'd like to move my exisitng # to vonage and then configure * as a client and do the turnk forwarding as we mentioned before
05:21.04IronHelixpretty common setup
05:21.07IronHelixone first gotcha
05:21.12xyklopzYeah ... nothing to complex
05:21.12IronHelixDONT USE VONAGE!
05:21.17IronHelix~vonage
05:21.18jbotrumour has it, vonage is a bunch of monkeys
05:21.22xyklopzthey only reason ... keep my #
05:21.29xyklopzi don't know about others that allow this
05:21.43IronHelixvonage will not work as a SIP trunk under any circumstances.
05:21.55IronHelixvonage will send you a device called an ATA
05:21.56IronHelix~ata
05:21.58jboti heard ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info
05:22.10IronHelixthe ATA is locked to their network and your account
05:22.17IronHelixand the only way to access your account si through the ata
05:22.32xyklopzokay ... I have a sipphone account
05:22.43xyklopzand they are only $35/yr for a #
05:22.50IronHelixits fine for ppl that dont use asterisk- they plug a phone into the box and they're done, but for asteirsk vonage is a bad way to go.
05:22.50xyklopzand 2c/min US
05:23.01IronHelixthats not a bad rate.  There are many such providers
05:23.12IronHelixalso companies like vonage but that allow byod (bring your own device)
05:23.16xyklopzi haven't seen anything comparable to the $35/year
05:23.18IronHelixasterisk counts as 'your own device'
05:23.27xyklopzright
05:23.48IronHelixkeep in mind tho the 35/year counts only for the number, you start paying when you make calls on it
05:23.51xyklopzso I'm staring my configs fron scratch
05:23.57xyklopzhuge configs scare me
05:23.57IronHelixit really depends on how many minutes you use
05:24.06xyklopznot incoming
05:24.17xyklopzonly outgoing @ 2c
05:24.19IronHelixyou can safely delete a good portion of the stuff in the default configs
05:24.22IronHelixyeah i mean outgoing
05:24.29xyklopzi barely use it
05:24.31IronHelixif you dont make a lot of calls then thats a sweet deal for you
05:24.35xyklopzto call out ... mostly in
05:24.37xyklopz:-)
05:24.38IronHelixhehe
05:24.43*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
05:24.43xyklopzso it's perfect
05:24.46IronHelixalso if you want a number to play around with
05:24.58IronHelixwww.ipkall.com will give you a free DID (phone number), only 360 area code tho
05:25.15IronHelixand many other providers can port your number, try teliax broadvoice quantumvoice
05:25.20IronHelixall support byod
05:25.22xyklopzso what is necessary in my sip.conf
05:25.26IronHelixi use QV, they have good service
05:25.27IronHelixanyway
05:25.29IronHelixsteps you need
05:25.38xyklopzI have ...
05:25.55sprnovaI want to use ztdummy.. but when I do a lsmod zaptel shows up as being used by ztdummy and ztdummy is (unused) is this normal?
05:26.06IronHelixthats fine
05:26.14sprnovathx
05:26.18IronHelix1. create each SIP provider a sip.conf entry, use sip show registry to make sure they register ok
05:26.34IronHelix2. create sip.conf entries for each of your softphones that you want to use
05:26.36IronHelix~softphone
05:26.38jbotsomething that should be drug out into the street and shot
05:26.57IronHelixso consider buying hardphones if you have ethernet- i can recommend some if you want.
05:27.12IronHelixonce you have the sip entries
05:27.19xyklopzwait wait
05:27.23xyklopz"SIP provider" ...?
05:27.30IronHelixie sipphone.com etc
05:27.32IronHelixif you have any
05:27.45IronHelixsipphone has a default config for sip.conf
05:27.59xyklopzin sip.conf?
05:28.03xyklopzthe sample
05:28.23IronHelixyou can dump all the default configs at the end of the thing
05:28.31IronHelixjust keep the top thing witht he variables and stuff
05:31.04IronHelixhttp://www.voip-info.org/wiki/view/SIPphone has a thing of the stuff to put
05:31.39xyklopzin the [authentication], do I have to add entiries for the REGISTER messages, I was getting errors before on that Username/auth name mismatch
05:31.55xyklopzaka <user>:<secret>@mydomain.com ??
05:32.49*** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com)
05:32.56IronHelixwell in the [general] section at the top you'll need to do something like register => yoursipphonenumber:yourpass@sipphone
05:33.02IronHelixthen put [sipphone]
05:33.06IronHelixand add the stuff like it has there
05:33.18IronHelixwhat should be [general] and [sipphone] are just underlined on that page
05:34.56*** join/#asterisk [hC] (n=hardcore@209.153.195.139)
05:37.11*** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
05:37.27*** join/#asterisk [hC] (n=hardcore@209.153.195.139)
05:37.49jeebusroxorshey there, can anyone recommend a web admin interface? Im pulling my hair out with voiceone...
05:38.18[av]baniahahaha. retarded workaround for grandstream limitation....
05:38.39IronHelix?
05:38.58[av]banimore autoprovisioning silliness from grandstream
05:39.07IronHelixhow so
05:39.19[av]banibut i have a workaround :)
05:39.25krischnoffjeebus: what is voiceone?
05:39.48IronHelixdo tell...
05:39.53jeebusroxorskrischnoff; its a web admin interface....
05:40.16krischnofffor a * GUI check out thirdlane.com
05:40.20IronHelix( i made the mistake of rolling 5 of them at a small site, after screwing with their cfg generator for a while i decided it would be faster to set them up manually )
05:40.22*** join/#asterisk YoMama (n=tchen@c-68-61-101-36.hsd1.mi.comcast.net)
05:40.40krischnoffit's also a company, that's why I was confused
05:40.53*** join/#asterisk xbmodder_lappy (i=nobody@unaffiliated/xbmodder)
05:40.59xbmodder_lappyhow can I see active channels
05:41.00xbmodder_lappy?
05:41.02jeebusroxorskrischnoff; thanks,
05:41.04IronHelixshow channels
05:41.04YoMamaanyone here have a GXP-2000?
05:41.06IronHelixyeah
05:41.12krischnoffPBX Manager is a webmin plugin
05:41.17YoMamahey IronHelix..ltns
05:41.19krischnofffrom thirdlane.com
05:41.23IronHelixim staring at the garbled upside down display right now :D
05:41.30IronHelixsup yomama
05:41.31IronHelixhwos life
05:41.44YoMamaIronHelix: eh...alright...haven't had time to monkey with * as much as I used to
05:41.51YoMamaIronHelix: were u saying u had a GXP-2000?
05:41.55IronHelixyeah i do
05:42.04YoMamaIronHelix: did u upgrade it to 1.0.2.8?
05:42.10IronHelixyuppers
05:42.26YoMamaIronHelix: k...why are the files named gxp2000a and boot55a?
05:42.34IronHelixnew bootloader
05:42.36YoMamainstead of gxp2000.bin and boot55.bin?
05:42.42IronHelixmaybe new firmware format or something
05:42.50YoMamawell, my 1.0.1.13 phone doesn't look for the a
05:42.56YoMamadid u haveta rename the files?
05:43.06jeebusroxorskrischnoff; got anything free? ;)
05:43.26*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
05:43.30[av]banithe beta firmware uses *a.bin suffix
05:43.36*** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk)
05:43.49IronHelixyomama- you gotta get 1.0.2.6 first
05:43.54[av]banibut you need the newer boot55.bin and gxp2000.bin also, so it knows to get a.bin
05:43.59IronHelixyeah
05:44.13IronHelixit has the last of the non-a.bin ones which tell it to look for the a.bin ones
05:44.13Dr-Linux:S
05:44.32IronHelixflashing 1.0.2.6 will make your phone reboot 2-3-infinity times
05:44.37krischnoffhey, you get what you pay for... ;)
05:44.52YoMamaIronHelix: so...i can't go straight from 1.0.1.13 to 1.0.2.8
05:45.03IronHelixyou can
05:45.10IronHelixjust get the non-a.bin's from the .6 release
05:45.13jeebusroxorskrischnoff; i know but im trying to put together something for a distro ;)
05:45.18IronHelixcombine them with the a.bin's from the .8 release
05:45.25IronHelixand put those 4 files on your tftp
05:45.27YoMamaIronHelix: gotcha
05:45.29YoMamathanks
05:45.29IronHelixthat said
05:45.33IronHelixif you have an older phone
05:45.40IronHelixuse the first one of the 1.0.2 series
05:45.55IronHelixthere the phone display only goes blank instead of garbling/wrapping/flipping/flippingout
05:46.05YoMamaIronHelix: what do u mean the first one
05:47.43IronHelixolder phone?
05:47.48IronHelixcertain MAC addresses
05:47.55krischnoffjeebus: there are the well known ones, AMP etc. Haven't used them myself though. Try to find the free O'Reilly Asterisk book, I recall that it reviews some GUI's
05:48.02YoMamaIronHelix: i'm pretty sure i have an older one..i ordered it soon after it came out
05:48.04IronHelix~book
05:48.06jbotbook is, like, a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
05:48.19IronHelixyeah
05:48.32IronHelixi find mine lasts for about two hours tops with .8 before the display starts to screw up
05:48.33IronHelixalso
05:48.45IronHelixbe very very careful as there is no way to downgrade to 1.0.1.13
05:48.50IronHelixor any 1.0.1 series
05:48.56YoMamaeh...now i'm wondering if i should even bother upgrading
05:48.58Mavviehmm... wonder if the congestion problem I experience now is related to the time-problem of asterisk 1.2.3 that friday afternoon.
05:49.02YoMamasounds like i'll mess it up more than fix anything
05:49.12Mavvieafter all, it always starts at 15:00 or 16:00.
05:49.43IronHelixi'd suggest unless you want to beta test it (its very beta) or you want one of the features in the changelog stick with .13
05:49.46IronHelixesp if you have older hw
05:50.17YoMamaIronHelix: good to know..k..screw it...it works just fine how it is
05:50.30IronHelixmy MAC is 00.0b.82.etc
05:50.59IronHelixhttp://www.voip-info.org/users/610/2610/images/420/medium.jpg   <- this could happen to you! :D
05:51.37IronHelixor perhaps this http://www.voip-info.org/users/557/15557/images/435/upsidedown-wraped.jpg
05:51.50justinuheh
05:51.52IronHelix(they still have a few bugs in the 1.0.2 series :)
05:51.53YoMamamine is 00.0b.82....
05:52.37YoMamaIronHelix: that's not so nice
05:52.42YoMamascrew it...it ain't worth it
05:52.46YoMamathanks for the warning
05:52.52IronHelixno problem :D
05:52.59IronHelixalso check out the wiki gxp2000 page
05:53.13YoMamayeah..that's what i've been reading
05:53.14*** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
05:53.27IronHelixlot of updates last few weeks as gs has been in active development
05:53.28YoMamai don't see any fabulous features in the 1.0.2.x series that make me want to jump up and down
05:53.54X-RobWell, except for sidetone, agc tha tworks, echo can on speaker phone, blf, paging,
05:54.07MikeJ[Laptop]1.0.2.x ????
05:54.07X-Robjust 'little' things.
05:54.09IronHelixthose were around in 1.0.1.13
05:54.28litagewould someone mind giving me an example FWD phone number? i just want to know its format
05:54.29IronHelixdunno about sidetone or agc
05:54.31MikeJ[Laptop]oh... nm
05:54.34X-Robsidetone and agc are _significantly_ broken in .1.13, resulting in feedback in a noisy environment.
05:54.37IronHelixlitage- its 5-7 digits
05:54.51X-Roblitage, 47876 is mine
05:54.54YoMamawhat's sidetown?
05:54.54litageIronHelix: does it have any other particular pattern?
05:54.57YoMamasidetone
05:54.57litagethanks X-Rob
05:54.58IronHelixnon-useless speakerphone, paging and blf were around in .13
05:55.08X-RobYoMama, when you speak in the mouthpiece, youhear your own voice in your ear.
05:55.17YoMamaX-Rob: oh yeah...
05:55.18IronHelixits a comfort thing
05:55.27IronHelixphone with no sidetone feels 'dead'
05:55.34X-Roband users always shout
05:55.44YoMamathe speakerphone isn't loud enough
05:55.47X-Robdeafening the person on the other end.
05:55.50YoMamapeople can rarely hear me well when i'm on it
05:55.57YoMamathey say i sound like i'm on the other side of the room
05:56.01IronHelixare you?
05:56.05YoMamahahaha..no
05:56.12[av]banigxp has no sidetone...
05:56.13YoMamaabout 2 feet away
05:56.18X-Rob[av]bani, yes it does.
05:56.20IronHelixim about 3' away from mine and its worked pretty well for a while
05:56.24[av]baniX-Rob: where?
05:56.30X-Robin the handset?
05:56.34X-Robwhere else would you put it?
05:56.34[av]banimine has zilch
05:56.35[av]banizero
05:56.37[av]baninada
05:56.46X-Robuse .2 then
05:56.47[av]banidead as a doornail
05:56.57[av]baniits never had any ...
05:57.01X-Rob.1 is the suxx0rz.
05:57.02YoMamaIronHelix: it's a cheap phone...too bad there's no way i'd sell it to a customer
05:57.05[av]bani1.0.1.9 through 1.0.2.8
05:57.10YoMamait's a bit too chintzy
05:57.14*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:57.15[av]baniive been waiting for them ot add it
05:57.17X-Rob[av]bani, sidetone is _definately_ there.
05:57.22X-Roblike, I just tested it then.
05:57.23X-Robit's there.
05:57.27YoMamasidetone works on mine
05:57.28[av]baniX-Rob: well, i dont get any sidetone on any of them.
05:57.29YoMamajust tested it
05:57.43IronHelixagreed- i sold them to one dude who must have said 'cheap, i want it cheap' 20 times while i was talking to him
05:57.44X-Rob[av]bani, you're on drugs.
05:57.44IronHelixPITA
05:57.51IronHelixi couldnt get the autoprovisioner to work
05:57.59YoMamai just wish i could get rid of my analog line echo...i even tried the echo preload and it didn't work
05:58.04IronHelixtheir config file generator utility kept asking for a license key
05:58.12[av]baniX-Rob: well, thats what grandstream told me when i discovered the http upgrade looping bug
05:58.35Mavviechan_sip.c: In function â:
05:58.36Mavviechan_sip.c:11066: warning: unused variable â
05:58.40Mavviewonder why FC4 does do that.
05:58.40YoMamaIronHelix: even the snoms are cheaper than most PBX handsets
05:58.45X-Rob[av]bani, I agree with them. The phones have sidetone.
05:59.13X-Robyou're on drugs
05:59.21IronHelixi can tell you as of 1.0.2.8 they have side tone (i just picked up my phone, hit 3 (kill dialtone) and blew into the mic
05:59.22YoMamammmm...drugs
05:59.48YoMamamine's at 1.0.1.13 and it's got sidetown
05:59.51YoMamasidetone
06:00.07X-RobYoMama, your sidetone can give you feedback
06:00.10X-Robyou'll enjoy that
06:00.17[av]banihmm, yes it has sidetone but its very low... its in a noisy environ
06:00.22YoMamaX-Rob: i've used the phone extensively...so far..it's been alright
06:00.24[av]banii can hear the sidetone in my snom 360 and polycom
06:00.32[av]banibut the grandstream i get very faint
06:00.41IronHelixyeah thats another thing
06:00.47IronHelixeither of you had the handset die on a gxp?
06:00.53X-RobYes, I agree, the grandstreams sidetone _is_ quieter than the snoms
06:00.59YoMamaIronHelix: not mine..still working
06:01.02X-Robbut the snoms have better acoustic baffling in the handset
06:01.03[av]baniits quieter than everything, i can barely hear it
06:01.10X-Robso they can crank the sidetone up more.
06:01.13YoMamaIronHelix: why?  do i have that to look forward to?
06:01.17IronHelixlol
06:01.18IronHelixdunno
06:01.25IronHelixone died the other day
06:01.30IronHelixgotta get them a new one
06:01.36YoMamaIronHelix: if this piece of shit breaks..i won't be buying another
06:01.44IronHelixbani- if i email GS asking for a replacement will i get a response?
06:01.54[av]baniIronHelix: ?
06:02.01IronHelixgxp handset died
06:02.13[av]baniyay?
06:02.14IronHelixyou've dealt directly with them right?
06:02.19[av]banino
06:02.20IronHelixoh
06:02.25IronHelixnm then
06:02.26IronHelixheh
06:02.35IronHelixi muust be thinking of thetatag
06:02.41[av]banii went round in circles with them over the http upgrade looping bug
06:02.43IronHelixthat did the alpha testing
06:02.56[av]baninever dealt with repairs
06:03.15YoMamahmm..atacomm has the polycom IP 301 for $115
06:03.28YoMamapolycom speakerphones rock
06:03.32IronHelixnot bad price...
06:03.33IronHelixhmmm
06:03.37[av]baniYoMama: 301 is monitor only
06:03.43IronHelixi just wish polycom would fix their firmware policy (its broken)
06:03.44YoMamaoh
06:04.01[av]baniIronHelix: just become a polycom reseller. problem solved
06:04.23*** part/#asterisk mogorman (n=mogorman@user-24-236-84-48.knology.net)
06:04.37IronHelixheh
06:04.40IronHelixthats one way to fix
06:04.43IronHelixwonder how hard that is...
06:04.46YoMamawhat's their firmware policy?
06:04.51[av]baniYoMama: "foad"
06:04.59YoMamafoad?
06:04.59[av]banipretty much sums it
06:05.06IronHelixF*** off and die
06:05.14xbmodder_lappyI keep getting this error:
06:05.14xbmodder_lappyexten => YOURNUMBER,1,Answer()
06:05.15xbmodder_lappyexten => YOURNUMBER,1,DIAL(SIP/user,20)
06:05.16xbmodder_lappyack
06:05.18xbmodder_lappywrong thing
06:05.22YoMamalovely...
06:05.23xbmodder_lappyFeb 16 22:04:49 NOTICE[7684]: chan_sip.c:10294 handle_request_invite: Failed to authenticate user "DHILLON AJAY" <sip:9252097318@207.174.111.12>;tag=as1cfbcb8d
06:05.25xbmodder_lappythat one
06:05.27IronHelixaka they wont give firmware to anybody except polycom registered partners and resellers, ever
06:05.28*** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com)
06:05.45YoMamaIronHelix: that's lame
06:05.56[av]baniYoMama: one of many complaints about polycom
06:06.03[av]baniprobably in the top 3
06:06.12YoMamaIronHelix: so u tell the people u bought it from that they have to make the firmware updates available to you
06:06.44YoMamaoh my god..this is scary..grandstream is making video phones now???
06:06.44[av]baniYoMama: good luck if you buy a used polycom from ebay.
06:07.06IronHelixyeah its lame...  i try to avoid them because of that (philosophical reasons)
06:07.12sprnovagot MOH working.. guess mpg123 did not like my complicated 192kb VBR files.
06:07.13xbmodder_lappyYoMama, yeah
06:07.16xbmodder_lappyand there good!
06:07.29IronHelixyeah whats with that- the gxv3000
06:07.38YoMamaxbmodder_lappy: nothing i've bought from GS would be defined as "good"...cheap maybe..but not good
06:07.39[av]banilooks like its a rebadged clone
06:07.51[av]banii found a phone exactly the same with slightly different styling
06:07.57[av]banilooks like grandstream just rebadged something
06:08.37YoMamaso the snom 320 is a nice phone?
06:08.45YoMamai've only heard good things..haven't used one in person
06:08.53[av]baniif you can stand the firmware, sure
06:09.00YoMamawhat's wrong with the firmware?
06:09.09[av]baniits buggy, and the ui is terrible
06:09.19[av]banisnom has no idea how to design a phone ui
06:09.19YoMamafor shit's sake..doesn't anyone have a stable, nice, inexpensive SIP phone?
06:09.31[av]baniinexpensive, stable, nice  choose 2
06:09.36YoMamaavban: neither does grandstream
06:09.43[av]baniYoMama: grandstream's ui is better than snom's
06:09.47YoMamathat's scary
06:09.50[av]baniyep
06:10.01YoMamai'm spoiled...before this..i was using a full blown cisco call manager solution
06:10.05[av]banigrandstream speaks better english than snom too
06:10.27YoMamafrightening
06:10.43[av]banievery phone has its warts, even cisco
06:11.17[av]baniat least with the gxp i dont feel like i got ripped off
06:11.48YoMamatrue...but what phones do u use in a business environment then?
06:11.59YoMamathey all seem...flaky...and crappy
06:12.22IronHelixim setting up a few aastra's later this week
06:12.23xbmodder_lappyI don't see how my family can talk on the phone for 60 frickin' hours!
06:12.23[av]banidepends on what warts you can tolerate
06:12.25IronHelixwe'll see how that goes
06:12.37[av]baniIronHelix: i hear aastra, at least the 9133i, has echo issues
06:12.39IronHelixnew firmware just released looks to clear up most of the warts
06:12.44IronHelixincluding that one
06:12.47[av]baniyay?
06:13.03[av]baniaastra seems the most progressive vendor, they test everything with *
06:13.04YoMamaaastra?
06:13.07kuku5YoMama: cisco call manager? so you used cisco 79xx phones?
06:13.14YoMamakuku5: yes
06:13.21[av]baniYoMama: www.aastra.com
06:13.27*** join/#asterisk ruza (n=ruza@holly.cervenytrpaslik.cz)
06:13.28YoMamakuku5: i know more about Cisco's CM than *..
06:13.28kuku5YoMama: how many do you have under you ?
06:13.31IronHelixyeah aastra telecom, used to be partnered wtih nortel or they did something with nortel as i recall
06:13.40YoMamakuku5: under me?
06:13.43[av]baniaastra used to supply phones for nortel i think
06:13.43kuku5that you mange
06:13.49[av]banitheyve been making analogue phones for decades
06:13.59YoMamakuku5: depends on how many projects i got going at once..i run a professional services division
06:14.03[av]baniyou can find jillions of aastra analogue phones on ebay
06:14.07YoMamakuku5: anywhere from 15-60
06:14.10IronHelixnow they partner with sayson, lots of adsi stuff
06:14.14IronHelixyeah
06:14.15[av]baniadsi :<
06:14.19[av]banieeeeeevil
06:14.30kuku5YoMama: I have 3 different clients using * and 7940/7960 's, each with like 20-30 phones
06:14.34YoMamaIronHelix: where do u get them?
06:14.40IronHelixi like voipsupply
06:14.46YoMamakuku5: using SCCP or SIP?
06:14.53IronHelixprices are decent and they have yet to royally screw anything up
06:14.57IronHelixvoipsupply.com
06:15.01kuku5YoMama: for some reason, within the last 4 weeks, at each client, the secretary'sphone (7960) just reboots, or hangs, or stops working, randomly
06:15.05kuku5sip
06:15.08[av]banivoipsupply are ok, they arent the cheapest
06:15.12[av]banitho their sales people are .. odd
06:15.19IronHelixhow so?  never talked to them
06:15.36kuku5i have a voipsupply guy on my aim - so its quick to get orders
06:15.56kuku5YoMama: did you see that happening with any of your phones?
06:16.01YoMamakuku5: so are u using them with SIP or SCCP?
06:16.03YoMamakuku5: never
06:16.06kuku5sip
06:16.09YoMamakuku5: they were as solid as a rock
06:16.17YoMamakuku5: try SCCP...i hear it's more stable
06:16.21kuku5its messed up -
06:16.22[av]banimaybe we just had bad luck, but they had trouble remembering that we declined their replacement and config contracts
06:16.29kuku5i tried 7.3, 7.4, 7.5
06:16.31QwellI don't know about more stable...
06:16.31YoMamakuku5: and i hear the sound quality is better
06:16.32kuku5but it wasnt like this
06:16.35Qwellsccp is fun though
06:16.53[av]banithe ciscos were designed for sccp. so of course they will work better with sccp
06:17.03YoMamais there a cheaper company than voipsupply..yeah..i agree..sounds like eveyrone's really happy with them, but their pricing isn't the best
06:17.19[av]baniwell, other pricing is usually only like $5 less
06:17.23YoMamakuku5: wait until your clients start asking u to write XML applications
06:17.34QwellXML apps are really fun
06:17.42[av]baniatacomm for example.. a little less, but their shipping prices are outrageous
06:17.54*** join/#asterisk _Thor (i=CS@dialup-4.250.138.230.Dial1.Weehawken1.Level3.net)
06:17.56[av]banilike $40 to ship a single spa3000
06:18.08Qwellatacomm is run by an assclown
06:18.16_Thorhello everyone
06:18.26YoMamaIronHelix: have u tried the speakerphone on the Aastra?
06:18.39IronHelixnot yet although i've heard its pretty good
06:19.07_ThorQwell: is it only one guy answering the phones at atacomm?
06:19.07[av]baniaastra also seem pretty responsive on firmware issues, making releases regularly
06:19.08YoMamahmm...$140 for the 9112i...too bad it's only one line
06:19.24IronHelixthey're all sitting in boxes excpet for the manager of the place who couldnt resist the temptation to tear his 480i-ct box apart and play with the little cordless thing
06:19.30[av]bani~phones
06:19.32jboti guess phones is at http://bani.anime.net/phones/
06:19.38Qwell_Thor: dunno
06:19.41IronHelix160 for the 9133...  got a bunch of those
06:19.46[av]banipolycom 501 is not bad, if you do not need xml
06:19.54_Thorit looks like it
06:20.02IronHelixholy crap bani that rocks
06:20.14[av]banishame polycom has yet to discover this mysterious device known as "backlight"
06:20.37[av]banimaybe theyre vampires or something, need to avoid exposure
06:21.05IronHelixhahahaha
06:21.36IronHelixsame with the linksys/sipura/cisco SPA-841/9xx...
06:21.40YoMamanice comparison chart
06:21.44justinudo you really talk on the phone in the dark ?
06:21.59YoMamajustinu: phone sex
06:22.00Qwell^ and if so, do you really need a light?
06:22.06YoMamahahahaha
06:22.16[av]baniIronHelix: sipura support has basically dried up since they were bought out.
06:22.21[av]baniIronHelix: and the 942... what a joke.
06:22.31*** join/#asterisk BugKham (n=lamer@202.8.86.170)
06:22.38jeebusroxorswhere can i get a cool linux shirt? haha
06:22.39IronHelixits too bad :(
06:22.46Qwelljeebusroxors: cafepress
06:22.47IronHelixsame thing with linksys and the wrt routers
06:22.50IronHelixall vxworks now
06:22.55justinui did like my old western electric bar phone with the backlit keypad tho
06:22.58justinuthat was a nice phone
06:23.12YoMamaIronHelix: so u think these Aastra phones are pretty rock solid?  what about the interface?
06:23.25[av]baniYoMama: http://www.o2m8.com/modules.php?name=News&file=article&sid=25
06:23.34IronHelixhavent even powered the thing yet...  they are mostly in boxes except one i had to bolt to the wall
06:23.37justinuaastra phone seems solid
06:23.46IronHelixthe one i did open (a 9133) felt pretty solid build-wise
06:23.56justinui have the 480i
06:24.01[av]banithe only complaint ive seen so far was echo... nobody complained about usability or construction
06:24.03IronHelixmore solid than the gs gxp2000
06:24.08IronHelixbrb
06:24.20[av]banicardboard is more solid than the gxp2000
06:25.04justinuit's not that bad
06:25.05YoMamaavban1: neat
06:25.06[av]baniwow justinu has a 480i?
06:25.10justinuyep
06:25.12[av]baniyou are full of suprises
06:25.16[av]banigxp, 480i, and snom
06:25.21justinuip601
06:25.26justinuip501
06:25.38Qwellwhat, no cisco?
06:25.40justinunope
06:25.44justinui need one for my zoo
06:25.50justinuthat 7970
06:25.51justinu:P
06:25.54Qwellzoo?
06:26.00justinuphone petting zoo
06:26.05QwellI see
06:26.08justinufor the clients
06:26.19[av]banijustinu: you agree, snom's ui is terrible?
06:26.26justinuyes
06:26.42justinumaybe slightly better than the gxp w/ 1.x firmware
06:26.58[av]banii wonder how much i can redesign it with xml
06:27.17YoMamathis phone's pricey though
06:27.28YoMamabut i guess u get what u pay for...
06:27.35YoMamais the 480i pretty rock solid?
06:27.40justinuyes
06:27.50[av]baniits been described as something you could club someone with
06:27.53justinucaveat emptor: it's PoE only
06:28.07[av]baniyea, thats a real suprise to people... no wallwart at all
06:28.12justinuyep
06:28.14YoMamawhich can be fixed with a $20 dongle
06:28.19justinujust a friendly tip
06:28.20[av]baniYoMama: its bizarre though
06:28.39YoMamaavban: how come?  if you're doing it in a business setting..not using PoE is lame
06:28.59[av]baniYoMama: lots of people arent setup for poe.. imagine them buying a pile of 480i's and going 'wtf'
06:29.10[av]baniafaik its the only voip phone which is _only_ poe
06:29.10YoMamaavban: oops..hehehe
06:29.30[av]banii was 'wtf' when i read the specs... i was looking for power supply for a while and didnt see one
06:29.37[av]banitook me a while to realize its poe only...
06:29.41YoMamaavban: Cisco is lame for PoE...i only recently found out that their polarity is opposite the standard
06:29.48IronHelixthe 480i ct comes with a wall wart...
06:29.53[av]baniYoMama: more recent ciscos are 802.3af
06:30.02YoMamaavban: yeah?  they fixed it?
06:30.06[av]baniYoMama: and many vendors support 802.3af _and_ cisco poe
06:30.12IronHelixhmmm
06:30.18IronHelixphone petting zoo
06:30.20[av]baniYoMama: ~phones
06:30.21IronHelixthats not a bad idea...
06:30.23IronHelix*schemes*
06:30.36YoMamaavban: i'm just saying that leave it up to Cisco to do something opposite everyone else
06:30.37[av]banithere is a poe column
06:30.40_Thorhelp, how do you place an unregister sip call with a prefix?
06:30.51[av]baniYoMama: to be fair, they implemented poe before the standard was ratified.
06:31.02IronHelixput the pattern exten in the default (guest) context
06:31.07YoMamaavban: hehe..ok ok..just ragging on them a little
06:31.07[av]baniYoMama: to their discredit, they kept using their mangled version after it was ratified
06:31.32YoMamaavban: it's hubris..they think they run the world
06:31.40[av]banithey do, mostly
06:31.41freata couple months ago I picked up a cisco catalyst switch... had to make this special cable to connect it to a serial port...
06:32.00YoMamafreat: dude..it's been like that for ages
06:32.01[av]banifreat: yes, thats cisco serial in general. welcome to 1994
06:32.07_ThorI am doing: dial(sip/xxxx${EXTEN}@domain.com)
06:32.18freatyeah I know... I'm just saying, it's nothing new for cisco to do that crap
06:32.23freateverything different
06:32.23_Thorwhereas xxxx is the prefix
06:32.24YoMamafreat: light blue cable...i got 400 of them laying around somewhere..gimme $15 and your address and i'll send you 10 of 'em
06:32.43freatthat was the point though... I wasn't about to spend more on the cable than on the switch hehe
06:32.43[av]banifreat: i dont mind that as much as i mind their list price for cisco db9 being $350
06:32.56kuku5anyone tried dell's poe switch ?
06:32.57[av]banifreat: or $700 for a 5 foot v35 cable
06:33.10YoMamahaha..yeah..the old cables were expensive as hell
06:33.28YoMamafreat: u know what i do?  so i don't haveta carry around a special cable?
06:33.36*** join/#asterisk |||sLaSh||| (i=th3_gam3@203.215.100.96)
06:33.45freatYoMama: no I don't know
06:33.48|||sLaSh|||hello where can i get cisco sip version 6 P0S3-06-0-00
06:33.51YoMamafreat: get one of those pin kits for a DB9..and pin your own thing so u can use a regular straight-through
06:33.55Qwell|||sLaSh|||: cisco
06:34.06YoMamafreat: that way you only carry around a little dongle rather than two cables..one ethernet and one cisco
06:34.16YoMamafreat: cost ya $3
06:34.20Himekoi just build my own cable
06:34.22YoMamafreat: and about 10 minutes of your time
06:34.30[av]baniYoMama: better yet, rj45 and then rj45->db9 connectors
06:34.39YoMamaavban: that's what i just said
06:34.50freatyeah I used a multimeter and some documentation online to make a cable
06:35.06YoMamaavban: but i didn't explain it so well
06:35.08freatripped apart a serial cable
06:35.12[av]bani:)
06:35.44[av]baniYoMama: kentrox has wacky serial too. so i have 3 sets of connectors around. rs232, cisco, and kentrox
06:35.45YoMamaavban: i used to carry around one for each of the companies..one for bay..one for cisco..one to make it null modem...etc etc etc
06:36.18[av]banioh yeah, APC has wacky serial too!
06:36.22YoMamaavban: and that one company that used to dominate the ISDN router space...what were they again?
06:36.30YoMamauhh..starts with a P
06:37.09YoMamaavban: yeah..but they're gracious enough to give u a serial cable for their equipment
06:37.29[av]banifortuantely they moved to USB, something they cant fuckup :)
06:38.44YoMamathese aastra 480i phoens are pretty badass
06:38.59YoMamaanyone have the CT with the cordless?  is the cordless any good?
06:39.05[av]banicept for the display being character based :/
06:39.07IronHelixim installing one this week
06:39.17linlinwhats an approiate dialing rule for a number in the UK that looks like this: +44 (0) 870 011 2988
06:39.18IronHelixi'll let you guys know how it goes
06:39.24YoMamaIronHelix: the CT?  with the cordless?
06:39.28IronHelixyeah
06:39.43YoMamalinlin: u don't need the 0 unless you're in the UK
06:39.50linlinok
06:40.00YoMamalinlin: it's just (your international dialing prefix) 44 870 011 2988
06:40.01linlinim in usa but im using a european sip provider
06:40.08*** join/#asterisk welles (n=welles@61.150.43.113)
06:40.16YoMamalinlin: doesn't matter unless you're in the UK
06:40.24YoMamawhich is why it's in paratheses
06:40.34linlinso... 1NXXXXXXXXXXXX
06:40.44YoMamauhh
06:40.48YoMama1?
06:40.53YoMamaif you're in teh us...don't you use 011?
06:40.56[av]baniIronHelix: 480i or 9133i?
06:41.11QwellIt's a UK provider...not an international call
06:41.18IronHelix9133's except the boss there who demanded a 480i-ct when i explained that you can check the weather on it
06:41.19linlinyou tel me, sorry really new at international dialing
06:41.27IronHelixaka he wanted the baddest most awesome phone in the office
06:41.28Qwelllinlin: Where is your provider located?
06:41.29*** join/#asterisk pengyong (n=lala@222.188.130.101)
06:41.33Qwelland how do they connect to the pstn?
06:41.39IronHelixthey dont yet
06:41.40[av]banihaha
06:41.41IronHelixnew business
06:41.42linlineurope somewhere, no idea
06:41.44IronHelixgonna be voip
06:41.46linlinits SIPDiscount
06:41.57Qwellsupport@sipdiscount.com
06:41.57*** join/#asterisk af_ (n=af@ip-165-17.sn2.eutelia.it)
06:42.11YoMamaIronHelix: great..now you're stuck writing an XML app :-P
06:42.15[av]bani480i ct is just 480i with support for wireless handset right?
06:42.24IronHelixyeah
06:42.33IronHelixyomama- they already have the weather one
06:42.33Qwell480i==aastra?
06:42.36IronHelixyea
06:42.38QwellWhy not get a 7970? :p
06:42.48IronHelixcuz aastra <3's asterisk
06:42.52YoMamaavban: the 480i supports a wireless headset?
06:42.57[av]baniheh yeah, cant say that about cisco
06:43.11IronHelix480i-ct comes with a cordless handset that looks like the linksys wip300
06:43.13IronHelixbrb
06:43.19Qwellcordless handset?
06:43.23Qwellor headset?
06:43.26IronHelixhand
06:43.30Qwellwhy?
06:43.41[av]banihttp://www.aastra.com/enterpriseip/pro_243.asp
06:43.46[av]banibecause
06:43.51YoMamaavban: how?  u can get a wireless headset with cisco's phone...
06:44.08[av]baniYoMama: handset not headset
06:44.16welleshi all. i need the confirm . in asterisk 1.2.4 the cmd 'meetme list 1000' only return the user numbers of conference room 1000?
06:44.30YoMamaavban: oh
06:44.41YoMamayou know what'd be sweet...if someone made these phones bluetooth compatible
06:44.43[av]baniQwell: i've just been informed chan_sccp and 7970 doesnt work with asterisk 1.2, only 1.0 ...
06:44.56YoMamathat way..you could use your existing headset with your business phone
06:44.56[av]baniQwell: does one need a different chan_sccp for 1.2 ?
06:45.16brookshireQwell: hi!
06:45.27Qwellbrookshire: omg hi!
06:45.33justinulike omg
06:45.35Qwell[av]bani: no, chan_sccp works on anything
06:45.36justinubarf out
06:45.42[av]baniYoMama: there's bluetooth support for *
06:45.43Qwell1.0, 1.2, svn, openpbx (allegedly)
06:46.10*** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net)
06:46.28[av]baniQwell: whats this chan_sccp2 business then?
06:46.35Qwellchan-sccp.berlios.de
06:46.37Qwellget that one
06:46.47Qwellthis one is actually maintained/used
06:46.48YoMamaavban: i meant the phones
06:47.11wellesany one will answer my question? i do need the info of meetme list <conferno>.
06:47.39YoMamawelles: huh?
06:48.32[av]baniah the berlios one is chan_sccp2
06:48.43[av]banior at least referred to as such
06:48.51wellesYoMama, cmd 'meetme list <conferno> ' in asterisk 1.2.4 only return the numbers of user in conference .i am right/
06:48.52QwellIt's THE chan_sccp, IMO
06:48.53welles?
06:49.00[av]banitheres three sccp drivers :/
06:49.01YoMamaIronHelix: so the 480i CT comes wiht a power brick?
06:49.05[av]baniwhat a mess
06:49.07Qwell[av]bani: the others suck
06:49.09IronHelixyeah
06:49.38YoMamaIronHelix: when i get my bonus..i'm gonna haveta buy one of these suckers :)
06:49.52YoMamaon a totally off-topic..anyone here got a smartphone?
06:49.59Qwellmy 7970 is smart
06:50.04YoMamawelles: lemme check
06:50.10[av]baniYoMama: if you want really top of the line phone, you can get a cisco 7970 for $385
06:50.17wellesYoMama, ok
06:50.35YoMamaavban: nah...i don't wanna spend over $200
06:50.40YoMamaavban: this is for home shit...
06:50.50YoMamaalthough the 480i looks like it'd make a good office phone
06:50.53*** join/#asterisk joaovianna (n=joaovian@ool-4351ce17.dyn.optonline.net)
06:51.31[av]banidepends on your requirements
06:51.53wellesYoMama, i remember this cmd will list the detail of users in the conference, at least in asterisk 1.0.9 it is true.
06:52.15YoMamawelles: mine's got the detail
06:52.30YoMamait shows extension, the user's name, the channel, etc.
06:52.33YoMamathen has a total count
06:52.42wellesYoMama, then what's wrong my asterisk?
06:53.03YoMamawelles: umm..i dunno...maybe your verbosity level is set too low?  that's a total guess
06:53.25sprnovaanyone know who MOH decides which song to play in the /usr/share/asterisk/mohmp3?? I have 3 songs.. it always plays the second one (alpabetically)
06:53.33wellesYoMama, my asterisk only return one line. i set verbose to 10 .no change
06:54.08YoMamasprnova: do a ps and look at the command line of mpg123
06:54.18YoMamasprnova: it might not be properly loading the first mp3
06:54.59[av]banihttp://movies.apple.com/movies/universal/miami_vice/miami_vice-tsr1_h640w.mov <- ONOES
06:55.10wellesYoMama, is there any other reason ?
06:55.48YoMamawelles: sorry..i'm not familiar enough with the code to tell u if there's a reason why there's no detail in your meetme list
06:56.24wellesYoMama, thanks ,any way. very strange
06:56.40sprnovaYoMama.. I have two mpg123 processes running wierd.
06:56.57YoMamasprnova: i've noticed that mpg123 doesn't die sometimes
06:57.04YoMamawhen u kill asterisk
06:57.38sprnovasounds like a bad first file.. I hear "thud" then it starts paying the second one.. LOL
06:57.54YoMamasprnova: there ya go
06:58.41sprnovaonly diff I can see is when I do a "file" command the one that does not play reports IDTag 2.3.0  the other two are 2.2.0
06:58.54YoMamaso who sells flash-based rackmountable PCs that u can run * on?
07:00.01ManxPowerYoMama, That would depend on your needs.
07:00.29YoMamaManxPower: put the OS and asterisk on flash...voicemail and everything bulky on a HD
07:00.59ManxPowerYoMama, interfaces to the PSTN and phones?
07:01.17YoMamaManxPower: it should have two PCI slots
07:01.37ManxPowerBest of luck.
07:01.43YoMamaone for some FXO/FXS ports..and one for dual-span T1 card
07:01.54YoMamawhat...are free PCI slots a problem?
07:02.02ManxPowerThings like Sokeris (sp!) tend to be pretty low power.
07:02.13ManxPowerbut ulaw only, no transcoding might work.
07:02.53maskedhas anyone here used telstra call waiting via a fxo?
07:03.11maskedu bout X-Rob ?
07:03.19YoMamaManxPower: 1U machines usually only have 1 PCI slot :(
07:03.56ManxPowerYoMama, You could prolly do a standard 1U if you can find one with 2 slots and use a CF IDE thingy
07:04.50YoMamaManxPower: yeah..just thinking about what it'd take to build a pretty good * box for business
07:05.11YoMamaPRI plus some analog for backup/emergency
07:05.24linlinhow can i manually force the hang up of a phone through asterisk -vvvr ?
07:05.31YoMamameans you haveta get a T1 card and a FXO card
07:05.49YoMamathat's two slots right there
07:07.34*** join/#asterisk welles (n=welles@61.150.43.113)
07:08.37*** join/#asterisk Fedoracore6 (n=dsd@219.95.15.117)
07:12.17Fedoracore6hello i try using ivr
07:12.27Fedoracore6but my amp didint work
07:14.08maskeddid you plug an instrument in?
07:15.17Fedoracore6what mean instrument
07:15.26Fedoracore6i press *77
07:15.32Fedoracore6to record my voice
07:15.36*** join/#asterisk welles (n=welles@61.150.43.113)
07:16.22*** join/#asterisk smurf (n=smurf@debian/developer/smurf)
07:16.24*** join/#asterisk smurfix (n=smurf@debian/developer/smurf)
07:16.24*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
07:17.15Fedoracore6masked its ... i must install to the addons version 1.2.1
07:17.31Fedoracore6and sound version 1.2.1
07:17.46Fedoracore6bacause i try to built touch tones registraion system
07:19.30*** join/#asterisk welles (n=welles@61.150.43.113)
07:21.17YoMamau still there Iron?
07:21.22IronHelixyeah just on phone
07:21.23IronHelixstand by
07:21.30*** join/#asterisk welles (n=welles@61.150.43.113)
07:22.38wellesmeetme list cmd can not work .anyone know what's wrong?
07:24.21*** join/#asterisk welles (n=welles@61.150.43.113)
07:25.00*** join/#asterisk trixter (n=trixter@65.172.209.246)
07:27.07*** join/#asterisk welles (n=welles@61.150.43.113)
07:27.46wellescmd 'meetme list <conferno>' can not return the detail of the conference'
07:29.10*** join/#asterisk L|NUX (n=linux@202.5.145.56)
07:29.30*** join/#asterisk welles (n=welles@61.150.43.113)
07:30.43IronHelixyomama- jabra makes a headset plug to bluetooth adapter, * also supoprts bluetooth.  welles- meetme list 1000 will list users in conf 1000 (i think).   yomama- yeah the 480i from what i've seen rocks, if this deployment goes well i'll probably pick one up, also FWIW the 9133 looks nice when wall mounted.
07:31.01IronHelixlinlin- do show channels, then soft hangup (channelname), ie soft hangup SIP/1234-abcd
07:31.17*** join/#asterisk welles (n=welles@61.150.43.113)
07:31.23*** join/#asterisk WasPhantom (n=neil@203-86-197-11-lightning.thepacific.net)
07:31.35YoMamaIronHelix: the 9133i apparently also has XML capabilities..that whole series uses the same firmware as the 480i
07:31.54IronHelixi dunno that it can do much with xml on a 2 line display
07:31.58IronHelixthe display is kinda small
07:32.06wellesanyone has some ideas?
07:32.14YoMamaIronHelix: yeah...but u can proabably do basic directory lookups and stuff
07:32.34YoMamaIronHelix: i wonder what u can access thru the XML interface
07:33.01IronHelixif it does have xml then theoretically at least the skys the limit
07:33.03IronHelixldap even maybe
07:33.24YoMamaIronHelix: as long as it supports push XML and query fields..then it'll do anything
07:33.33YoMamawell, and allow protected access to some of its features
07:35.05YoMamareading the API guide now
07:35.17*** join/#asterisk welles (n=welles@61.150.43.113)
07:35.35IronHelixi skimmed that doc once
07:36.42*** join/#asterisk MGSsancho (n=user@adsl-67-121-107-88.dsl.irvnca.pacbell.net)
07:37.06IronHelixyou can (as i recall) def. set password fields
07:37.23YoMamaIronHelix: seems pretty complete
07:37.56IronHelixyeah extremely helpful
07:38.06Qwellkernel-module-ntfs-2.6.15-1.1831_FC4.stk16-2.1.25-0.rr.10.4.i686.rpm
07:38.11QwellI freaking LOVE redhat
07:40.48YoMamaIronHelix: cool...u can push phone #'s for it to dial
07:40.55YoMamathe only probolem is...the only auth method is IP
07:41.05johnsu01ok, so I've got local extensions working, and I can make an iax test call, but I can't make local calls (using Junction)...
07:43.52IronHelixhey has polycom changed their firmware policy at all?  i just went to their site and i get a form that looks like its going to let me register to download FW...
07:44.14YoMamahehe
07:44.17YoMamathat'd be interesting
07:44.20MstlyHrmlsyou can get the older firmware now
07:44.27MstlyHrmlsbut not the up-to-date stuff
07:44.47IronHelixi still cannot understand what they hoep to gain with that
07:45.03*** join/#asterisk [Airwolf] (n=airwolf@attilla.nl)
07:45.10*** join/#asterisk jaike (n=a@203.131.137.76)
07:45.30YoMamaIronHelix: NOTE: At this time, end-user customers can only download previous software. Please work directly with the Polycom Certified VoIP Reseller you purchased the products from to obtain the most current and appropriate software.
07:45.33YoMamafrom their website
07:45.38YoMamaidiots
07:45.42IronHelixwtf
07:45.45IronHelixi want to smack them
07:45.46IronHelixso i do
07:45.56IronHelixby not buying their products *smirk*
07:46.31YoMamaIronHelix: they're trying to treat their resellers like value-added partners
07:46.45YoMamabut the problem is that if they're going to do that..then they shouldn't let companies like voipsupply sell them
07:46.54YoMamait should only be to integrators
07:46.56YoMamaso they're idiots
07:47.02IronHelixyeah
07:47.07IronHelixi find many companies have this problem
07:47.15YoMamayeah..stupid
07:47.23IronHelixthey do not know / understand (or at least act like they dont know/understand) who their customer actually is
07:47.23YoMamaCisco doesn't even have this policy
07:47.41YoMamau can get smartnet on their phones for $20/year or something like that
07:47.46IronHelixCisco charges for firmware- but they know you are their customer and they are happy to bill you like $8 to sell it to you
07:47.52YoMamayeah
07:47.57YoMamasmartnet is cheap for the phones
07:48.13IronHelixwhereas polycom needs to decide if their customers are the VARs or the customers
07:48.16YoMamathese aastra phones look pretty fancy schmancy
07:48.20IronHelixi say its the customers, but thats me cuz im a customer :)
07:48.21YoMamaIronHelix: exactly
07:48.35YoMamaIronHelix: what's a good phone for the receptionist?
07:48.42YoMamafor let's say..an office with 30 people
07:48.54YoMamaany phones with like 30 BLFs?
07:49.02IronHelixtry a snom 360, it supports expansion sidecars with i think 48blf's each
07:49.19IronHelixyou can link up to 2 sidecars onto a 360
07:49.29YoMamai wonder if the cisco 7960 with two sidecars works
07:49.34trixterMmmm..  ribs
07:49.37YoMamaa cisco 7960 will take two sidecars
07:49.41IronHelix<3 ribs
07:50.07YoMamaIronHelix: have u set up the 360 with a sidecar before?
07:50.18IronHelixno
07:50.20IronHelixbrb
07:50.39trixterthe sidecar must be nice, as you ride around town your dog can ride in the sidecar
07:50.45trixterget some wind in his face
07:52.12trixterits cold and stuff
07:52.20IronHelixlol
07:52.37YoMamaha
07:52.38jaikethere was a discussion yesterday about canreinvite able to lower system utilization. how?
07:52.55*** join/#asterisk Sajid_Khan (n=human@203.145.159.37)
07:53.19trixterif you directly connect the rtp endpoints asterisk doesnt have irqs from the network traffic, it doesnt have cpu overhead for processing rtp streams, it doesnt do codec translation, etc
07:53.41trixterthere are a lot of reasons why it lowers utilization on that specific server but there are some dangers depending on environment
07:54.38jaikedangers? like natted networks?
07:54.56trixterif you want to ensure that you get billing info (CDR) the media streams have to go through asterisk because if someone doesnt send end of call back to asterisk (say both endpoints just unplug their device instead of hanging up - especially becuase they know its a free call that way) or whatever
07:55.27trixterthat generally isnt as big of a problem because normally net->net is free and its harder for them to just unplug the pstn, but its something to consider
07:56.00jaikeoh ok..so it can mess up CDR
07:56.11trixteryeah but generally that isnt as easy in real world situations
07:56.15trixterits just 'possible'
07:56.51trixterif you have to do wiretapping you cant if you directly connect them to someone else..  so like CALEA requirements in america - if you dont have the RTP data you cant record it
07:57.23YoMamaok...bedtime
07:57.25YoMamagoodnight y'all
07:57.29YoMamathanks for the help
07:58.17trixterif you run a calling card system where someone can for example press *# to make another call without hanging up and calling back you must use sip-info for dtmf, which may not be supported on the remote end
07:59.52*** join/#asterisk |||sLaSh||| (i=th3_gam3@203.215.100.96)
08:01.49|||sLaSh|||hi im using 7960 and asterisk@home, i got the message tftp file not found, "system is unavailable"
08:03.08jaiketrixter: you lost me there :). googling canreinvite
08:03.35trixterum there was an american idol?  and I missed the chance to skew the voting results?
08:03.41trixterI have 120k votes to cast!
08:04.11trixterI gotta find someone to watch it for me so they can tell me who was worst
08:05.43IronHelixlol
08:06.46*** join/#asterisk oej (n=oej@62.97.243.70)
08:11.17zoahey ho olle
08:11.41zoaany news on the exams
08:16.09*** join/#asterisk Fedoracore6 (n=dsd@219.95.15.117)
08:20.00*** join/#asterisk Gunnar (n=gunnar@62.97.243.70)
08:21.56*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-244.claranet.co.uk)
08:22.20[av]banihmm anyone used avaya phones ?
08:24.48L|NUXhello every one
08:25.17L|NUXcan any one tell me is this possible to run asterisk on multiple ports ?
08:25.32trixterhow do you mean?
08:25.34trixterspecifically
08:25.35L|NUXlike i have provider for both 8891 and 5060 how can i bind them in my asterisk ?
08:25.45jaikeam able to get passthrough to work for g729 between my phone and our provider with asterisk in betwen, but if i use put Monitor in the dialplan, i see 2 decoder licenses used. Asterisk has to decode both in and out channels?
08:25.47*** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at)
08:26.04trixteryou need a license to encode or decode
08:26.07trixternot just decode
08:26.07*** join/#asterisk welles (n=welles@61.150.43.113)
08:26.24trixterthe way it works is that it will decode from you, then reencode to the remote site
08:26.28jaike0/2 encoders/decoders of 30 licensed channels are currently in use
08:26.37trixterit could be made such that it only needs 1 license but asterisk doesnt work that way
08:26.41jaikewithout monitor, its 0/0
08:26.58trixterok then its decoding both streams
08:27.05trixteras such it requires 2 licenses one for each endpoint
08:27.17trixtereither way, it still could be done to only use 1 license
08:27.20trixterbut it doesnt work that way
08:27.24*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
08:27.50wellestrixter, cmd 'meetme list <conferno>' can not return the detail info in my asterisk. can u help me?
08:28.13*** join/#asterisk apardo (n=apardo@62.97.121.93)
08:28.24wellestrixter, i really troubled by it.
08:28.36jaikewelles: are you using 1.2.4? it works fine with us
08:32.51*** join/#asterisk CaRb0n^ (i=Genocide@203.81.238.51)
08:34.34shido6anyone diabetic?
08:35.02jaikewww.voip-info.org currently down?
08:36.34[av]banii get to it fine
08:37.41*** join/#asterisk wellng (n=welles@222.90.175.97)
08:37.52jaikehmmm
08:38.11*** join/#asterisk psk (n=psk@golia.caltanet.it)
08:42.14johnsu01if I have this: "exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@jnctn)" in my extensions.conf, and an entry for [jnctn] in my sip.conf, why might asterisk still be treating a number like 15555551212 as a local extension?
08:42.51*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
08:43.00justinuipv6?
08:44.14johnsu01hm? don't think so.
08:44.55shido6errr
08:44.59shido6sip debug.
08:45.11shido6what context is it looking in
08:48.26wellnghi jaike  iam welles i come back
08:48.28johnsu01shido6: It's looking in default.
08:49.17wellngjaike, let 's continute to talk my issue .cmd meetme list <conferno> ok?
08:49.18*** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net)
08:49.38jaikewellng: meetme command works fine in 1.2.4, or even in 1.2.0
08:49.59*** join/#asterisk L|NUX (n=linux@202.5.145.56)
08:50.43wellngjaike, yes. i know. but now i don't know why my asterisk can return the detail but only one line .
08:51.07[av]banihttp://www.intertex.se/products/page.asp?iPageID=146
08:51.07[av]banio.o
08:51.30justinu(00:42:51) florz [n=florz@2001:1a50:503c:0:0:0:0:1] entered the room.
08:51.30justinu(00:43:01) justinu: ipv6?
08:51.37justinuthat's an ipv6 address
08:51.41johnsu01oh, right :)
08:51.59wellngjaike, yes. i know. but now i don't know why my asterisk can not reurn the detail but only one line .
08:53.20jaikethat i cant answer
08:53.36johnsu01ah, got it
08:54.39jaikethis is what i get when when i issue the meetme command
08:54.51jaikeUser #: 01          400 400                  Channel: SIP/400-f697    (unmonitored)
08:54.52jaike1 users in that conference.
08:55.25wellngjaiger, yes. this is also what i expect .but mine is only the last line
08:57.04wellngjaike, , yes. this is also what i expect .but mine is only the last line
08:57.11jaikerecompile? hehe
08:58.00wellngmaybe can not work.because my asterisk version was 1.2.1 and update to 1.2.4 .it still can not work
09:01.05*** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net)
09:02.30*** join/#asterisk walhala (n=walhala@stardust.noc.frontier.fr)
09:02.31walhaladoes anyone know how to setting up sms with * ? I use trunk version actually
09:06.02*** join/#asterisk sergeus|w (n=sergeus@ippe-245.ippe.ru)
09:10.01[av]banihttp://www.voip-info.org/wiki/view/Pingtel+Hardphone
09:10.08[av]banigood lord. ugliest phone ever?
09:10.17[av]banilooks like something out of the jetsons
09:12.49trixtermaybe it is
09:14.00trixterthe freeswitch hardphone doesnt sound too bad ...  hardphone that can act as either your phone, your pbx, both, color lcd touchscreen, runs linux, playing with the internals is encouraged, projected price $300
09:16.10[av]bani?
09:17.33liran_would anyone care to test my billing program? it's a config file, and cgi script. it reads everything from master.csv
09:17.51trixterhow does it do billing?
09:18.35*** join/#asterisk alexandrekeller (n=alexandr@mail.voffice.com.br)
09:19.29[av]banipaypal!
09:19.41trixtere164.org earned $1.50 today..  didnt know there were so many people using enum, its not a lot of cash, but its less than 1 day, it also suprised me cause I figured they would only do $5/month
09:20.00*** join/#asterisk jaike (n=a@203.131.137.76)
09:20.51trixter[av]bani: the freeswitch people are designing a hardphone, with the goal that its open enough that you can do whatever you want with it.  it will run freeswitch in linux, although you can swap that out if you want ...  it sounds really cool, and for some people that would be the only device they need..  for just email 640x480 lcd touchscreen is fine
09:21.09trixterCF slot too so that you can store your data and take it with you
09:21.19liran_trixter: with a configuration file. it looks up rates you define for the phone number field in master.csv
09:21.36trixterliran_: does it support custom rate tables per customer?
09:21.49liran_trixter, yeah, its very dynamic
09:22.28liran_trixter, i've got a sf project page up but i havent uploaded any files yet cause i want to test it somewhere first.
09:22.28alexandrekellerI think there's a bug on app_queue; can anybody help me ?!
09:24.22trixterI might look it at, where can I get it?  what language is it written in?
09:24.26zoatrixter: the phone is probably going to be very expensive then :)
09:24.30trixterwhat platforms does it work on?
09:24.43trixterzoa: projected price is $300
09:24.54trixterwhich isnt bad for what you get considering it can be your whole pbx if you wanted
09:25.17zoai dont think they will be able to make it for that price
09:25.23trixterwhy not?
09:25.41liran_trixter, ill email it to you. its in perl and requires ofcourse apache with cgi support and it needs a perl module.
09:25.41zoathey will need to have to pay for a big production run
09:26.17trixterthey seem to think that they can after talking to people who will be building it
09:26.31liran_trixter, if you see fit, i'd like to customize straight down to your needs and have some sort of restricted access just to edit the code and test it.
09:26.43trixteroh well if its just a cgi I am not that interested ...
09:26.51liran_how come?
09:27.17trixterbecause I am not that interested in a web based solution
09:27.19*** join/#asterisk cjk (n=cjk@80.92.64.103)
09:27.26liran_ahh ok
09:27.27cjkhi, any cdr master here?
09:27.31trixterI guyess I will look at it, but prolly wont use it for real..  trixter@0xdecafbad.com
09:27.47zoait would be cool though
09:27.54cjkif i do a forkcdr, any command following this modifiying cdr data should only be applied for the new cdr? correect?
09:27.56liran_trixter, ok. would it be possible for me to test it on your box?
09:28.10trixterit also probably looks at the src field, if so then it certainly wont work for what I am doing
09:28.19trixterI can test it for you on my box
09:30.35*** join/#asterisk apardo (n=apardo@62.97.121.93)
09:31.02wellnghi all, my asterisk ' cmd 'meetme list <conferno>' can not work.it can not return the detail of a conference room. anyone has ideas?
09:31.34trixterwhat details specifically are you looking for?
09:32.22wellngtrixter, i want control the conference using agi
09:32.48trixterthat doesnt answer the question
09:35.00wellngtrixter, for example .User #: 01  11234 welles  Channel: IAX2/11234    (unmonitored) what i want get is 11234
09:36.18trixterwhat is your verbosity set to?
09:36.29wellngtrixter, 10
09:36.46wellngtrixter, any value will not work
09:37.34alexandrekellerhi all
09:37.55[av]banitrixter: url for the freeswitch hardphone?
09:37.59alexandrekellerI'm looking for someone who can help me with app_queue ?!
09:38.08alexandrekelleranybody ?!
09:38.32wellngtrixter, more stange is that it will still occur i reboot the machine or recompile the asterisk
09:43.59*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
09:45.49[av]bani~phones
09:45.50jbotphones is probably at http://bani.anime.net/phones/
09:45.54[av]banilist is getting long :o
09:47.17*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
09:47.43fishboy1669hi
09:47.54fishboy1669is royk here today?
09:48.18zoahe just joined
09:48.27fishboy1669cooleo
09:49.53RoyKfishboy1669: tried the mrtg plugin as well?
09:49.54trixter[av]bani: afaik they dont have anything but conversations right now.  I just happen to have talked to them about it
09:50.05trixteryou are free to do the same ...
09:51.13trixterwellng: I am almost thinking that you have a mad module for meetme and that you may want to look at reinstalling that module and making sure that it gets overwritten with the current install.  that may not be it, but there is certainly something not right
09:51.13[av]baniheh. seems to me they could just take a reference design off the shelf (like snom and grandstream do) and build something around it
09:51.20trixterI dont use meetme enough to know what the problem is
09:52.55trixterfrom what they have said they are building from the ground up to get the features they want
09:53.01trixterthey also said the case alone is like $25 :(
09:53.19[av]baniunless you buy in lot 10,000, cases are usually expensive
09:53.20*** part/#asterisk Aragone (n=arathorn@puma.mxtelecom.com)
09:53.46trixterwell it was something about it being custom made rather than an off the shlef case they can shove something into
09:54.04[av]baniseems silly, unless they want to do something with the case nobody has ever done before
09:54.49trixterwell 1. it has to be a deskphone case, 2. it needs the lcd, I am unsure of the other properties that are required but at the very least those are design constraints
09:54.56trixtermost deskphones dont have a 640x480 lcd
09:55.01[av]banicustom made cases are $$$ just for the design, because they have to be made for the molding process
09:55.21[av]baniif you get an off the shelf case, thats already been taken care of
09:55.27trixterthat is aparently built into the cost from what I gathered
09:55.31trixterwhich results in $25/case
09:55.32[av]bani640x480 seems over the top to me
09:55.40[av]baniif youre doing a videophone, sure
09:55.42trixterwell its more than a simple phone
09:55.59trixterthe system runs linux, you can do email, sms, im, etc all off the phone
09:56.09trixterfor many people there wont be a great need for an actual computer
09:56.10[av]baninothing special there, i could do that with a zaurus
09:56.25trixteryou wont have the cpu power off the zaurus
09:56.32[av]banifor what?
09:56.37trixterscroll up
09:56.38*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
09:56.48[av]baniwhat, you need cpu for im? email?
09:57.03trixterI was refereing to the at least 2 times I siad that the phone could be your pbx
09:57.18trixteras in it runs freeswitch you can have *other* things connect to it, it can do voicemail for those *other* phones, etc
09:57.21trixtercodec translations
09:57.22[av]baninever tried, but i could probably put asterisk on zaurus
09:57.37trixterand how many channels can you get working?
09:57.39[av]baniits beefier than my wrt54gs, which runs zaurus
09:57.43[av]baniwhich runs *
09:57.55trixterhow about recording to voicemail at the same itme you are processing multiple other channels with codec translations?
09:57.58[av]banishrug, anything embedded is going to be about the same
09:58.07trixterif you say so
09:58.08[av]baniunless you go x86 variant
09:58.10[av]banior ppc
09:58.37[av]baniarm, mips, youre not going to be able to push them very far
09:58.43trixterregardless they are doing it and doing it differently than you said, such as lower price, etc..  they have talked to hardware designers/manufacturers about this specific phone as well
09:58.47trixterthey arent talking about arm
09:58.56trixterI never said they were building a zaurus
09:59.01trixterI said it had more cpu than a zaurus
09:59.13[av]baniwell, if you want anything approaching something for pbx, you want ppc then
09:59.23[av]banispendy...
09:59.32trixterI hope the release more info on it soon, especially pictures
09:59.43[av]banippc is not cheap to embed
09:59.46[av]banisadly
09:59.53trixterthe concept is nice, and the fact that they arent trying to make it all secret and closed is also nice
10:00.00trixterits not a ppc
10:00.04trixterbut anyway
10:00.28[av]baniwell, mips and arm dont have real processing power for much...
10:00.31trixterI would like to see some initial design pictures of the case
10:01.39[av]banii assume they rent going to sell it for any profit
10:01.41*** join/#asterisk cng^ (n=cg@217.23.169.4)
10:01.43wellngtrixter, yes. you are right. i rm a module . they work fine
10:02.10trixteras I understand it htey also want to put zeroconf in it so that when you plug it in to your network other zeroconf enabled devices will just work, granted you will have to enter a username and password, but you wont have to configure everything else
10:02.39trixter[av]bani: I dont know how much profit they are planning on making with it, I do know they arent profiting off the software side of things
10:02.43*** part/#asterisk cng^ (n=cg@217.23.169.4)
10:02.54[av]baniif they want to build something for themselves, great they can knock themselves out
10:03.14trixterwhy would they have a suggested sales price if that were the case?
10:03.17[av]banibut i have seen soooo many people make 'open hardware' which failed because they made some retarded design decision nobody wanted or which was just not practical
10:03.18wellngtrixter, thanks
10:03.40[av]bani300 sounds just about what it would cost for small production run of something
10:03.58trixterI fail to see why you want to argue with me when I have made it clear its someone else that is doing this, and the fact that they have presented, so I feel its time to say again, please let it go
10:04.04trixterwellng: np, glad it worked out for you..
10:07.01alexandrekelleris this a business channel ?!
10:09.58trixterthis is an asterisk channel, so basically anything that is compatible with asterisk is fair game
10:11.36trixterwow someone is selling for $89 a section of the CFR which you can get for free..  makes you wonder how some companies can even get customers
10:15.44*** join/#asterisk fulgas (n=fulgas@82.102.2.254)
10:18.22[av]banidoes editline still need to be exorcised from asterisk?
10:18.58pifhi, any user of the cisco 7920 (wi-phone)?
10:19.08[av]banithats an expensive phone.
10:19.21pifi know :)
10:19.57[av]baniyoure probably the only one here who has one. :/
10:20.17pifand it's crap to boot
10:20.31pifcisco is a bunch of theives
10:23.45*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
10:24.13[av]banino, theyre just often out of touch with their customer base
10:24.30[av]banisometimes it takes them a couple models before they get a technology right
10:24.49*** join/#asterisk fourcheeze (n=rich@82.153.215.21)
10:25.04[av]banithough lately they seem to be really stupid, making their support pages IE-only
10:25.08pifcisco is a profit-driven company, not a technology-driven company
10:25.48[av]banierr.. all companies are profit driven, or they wouldnt be companies :)
10:26.38[av]baniall the wifi phones on the market seem to suck, its suprising the expensive cisco one does too
10:27.08[av]banii might expect that of the zyxel
10:27.11pifwhat I mean is: they'll not stop at taking ugly technical shortcuts to stuff their customers (ans lose some in the way)
10:27.16[av]banibut cisco is suprising
10:27.48[av]banihmm.. i dont know if i ever saw cisco take a technical shortcut. they just like proprietary stuff
10:27.52pifwhereas a tech-driven co will not stoop to rebranding a chinese made piece of crap and mark it up 5X
10:28.06[av]banifortunately the market they are in is not favorable to proprietary
10:28.13[av]baniso they can only do so much damage :)
10:28.47pifthe suits like cisco now
10:28.52*** join/#asterisk Assid (n=assid@203.115.64.11)
10:28.53[av]banithey always did
10:29.00pif'cause it's 'safe'
10:29.11[av]baniit was always 'safe', and for a long time was the only game in town really
10:29.26johnsu01I can't get audio on my sip calls. I think I probably need some of the nat settings, since there is a firewall between my asterisk and Junction, but I can't seem to get them right.
10:29.46[av]banijohnsu01: nat=yes qualify=yes in sip.conf
10:30.40[av]banipif: what cisco is good at is obsoleting your investment very quickly, making you re-buy the same stuff over and over.
10:30.50[av]banipif: something microsoft is only just getting good at
10:31.09pifa pattern there
10:31.33[av]baniof course tech stuff becomes worthless insanely fast in general...
10:31.52piffree software is having remarkably little impact on these attitudes yey
10:31.54pifyet
10:31.55[av]bani$2000pc is worth $100 in 4 years :/
10:32.45[av]baniwhat? free software kicked sgi's and dec's ass
10:32.51[av]baniand it's pounding sun
10:33.13[av]banisun is having to do a major attitude readjustment thanks to free software
10:33.41[av]baniand.. juniper is competing with cisco, and they build their routers around bsd...
10:34.18johnsu01[av]bani: still no luck.
10:34.29[av]banigive asterisk a few more years and im sure it will start pressuring cisco's callmanager
10:34.37[av]banijohnsu01: :/
10:36.28johnsu01gah, now I've got some audio..
10:36.43*** join/#asterisk X-Gen (n=x-gen@dsl-146-124-49.telkomadsl.co.za)
10:36.54johnsu01I ring my own cell phone, and when I send the call to voicemail, the call drops.
10:37.01johnsu01But at least I hear the ringing :)
10:38.13*** join/#asterisk alexandrekeller (n=alexandr@mail.voffice.com.br)
10:38.42alexandrekellerI really need some help on app_queue....anybody please ?!
10:38.59johnsu01very strange, I hear ringing when I call some numbers, but not others.
10:39.53*** part/#asterisk X-Gen (n=x-gen@dsl-146-124-49.telkomadsl.co.za)
10:40.03jaikealexandre: just post your question
10:41.21alexandrekellerI have 5 agentes logged in. But only one call at a time rings on them, when a second call enters the queue it' kept on hold until the first call is answered
10:41.28alexandrekelleris it a bug or what ?!
10:43.56jaikeasterisk version?
10:44.25alexandrekeller1.2.4
10:44.36alexandrekellerbut it happens too on 1.0.10
10:45.11jaikeive never really noticed that
10:45.36alexandrekellerme neither until yesterday
10:45.37johnsu01[av]bani: So, I can hear ringing now, but as soon as the call is picked up by voicemail, I lose the audio. The call stays connected, though.
10:45.38*** join/#asterisk Falle (i=falstaf@213.141.80.88)
10:46.20jaikealexandre: lemme test that
10:46.27alexandrekellerok
10:47.03[av]banijohnsu01: ringing is via sip signaling, audio is via rtp. so your rtp is being blocked
10:47.30johnsu01hm. I have the rtp.conf ports set according to Junction's numbers, and I have my firewall forwarding that range of ports as well...
10:48.11[av]baniorigination from junction to your * ?
10:48.59*** join/#asterisk ramtha (n=ramtha@195.14.234.162)
10:49.03ramthahi
10:49.23ramtha2 tep wilcdards (quad span)
10:49.29ramthaone is working correct
10:49.36ramthathe second seems not so
10:49.41[av]baniif your * is behind a nat firewall, you'll probably need to set externip= in sip.conf
10:49.49ramthain /proc/zap/ i only have 5 spans
10:50.19ramthai think i must have 8
10:50.24jaikealexandre: same here..i guess its supposed to work that way
10:50.34jaikeFIFO
10:50.35ramthazttool did not display the second card
10:50.39ramthawhat is wrong here
10:50.39alexandrekellerreally ?!
10:50.45*** join/#asterisk L|NUX (n=linux@202.5.145.56)
10:50.54jaiketry strategy=ringall :)
10:51.40johnsu01[av]bani: I've added externip, but same result.
10:51.55alexandrekelleroh sh.....I don't believe.....
10:52.01alexandrekellerwell, thanks mate
10:52.08johnsu01Will that take effect on reload, or do I need to restart? I'll try restarting as well.
10:53.03[av]banijohnsu01: sip reload should be enough
10:53.05*** join/#asterisk SupZ (n=icechat5@200-158-166-207.dsl.telesp.net.br)
10:53.42[av]banijohnsu01: you should see if you can packet dump on your firewall, see if rtp packets are arriving and being blocked
10:56.25sternnls
10:57.10[av]banimv
10:57.20*** join/#asterisk darkskiez (n=mhb@bb-87-81-62-203.ukonline.co.uk)
10:58.10johnsu01[av]bani: they rtp packets are getting through. I can see them via ethereal on the machine behind the firewall.
10:59.33*** join/#asterisk __chris (n=chris@unaffiliated/redlined)
11:01.02*** join/#asterisk BhaalWTF (n=bhaal@CPE-141-168-108-119.qld.bigpond.net.au)
11:03.02trixterhttp://www.trxtel.com/index.php?page=Tollfree_Termination  make money sending voip traffic, what a bargain
11:03.37[av]banijohnsu01: so they are probably not being translated properly
11:08.37*** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk)
11:09.28*** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com)
11:09.41stoffellgoodday all
11:11.02johnsu01[av]bani: I'll have to work on it more later I guess. At least I've got the calls working with iax.
11:11.08johnsu01[av]bani: thanks for the tips.
11:13.15sternnftp ftp.digium.com
11:13.25sternndoh!  Wrong window again.
11:15.26*** join/#asterisk aymeric (n=ablazy@62.36.227.220)
11:16.15stoffelllol
11:19.51*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
11:21.57*** join/#asterisk DataCompBoy (n=datacomp@217.8.236.1)
11:22.06DataCompBoyHi all :)
11:22.27DataCompBoySorry, may be I have missed anything... But: is pre-compiled .deb's for ztdummy present?
11:22.45fourcheezeDataCompBoy: not last time I checked
11:22.59fourcheeze(about 2 days ago)
11:23.07DataCompBoywhat about asterisk 1.2.4 ?
11:23.16stoffellyou could make one yourself with checkinstall or something
11:23.16*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
11:23.19fourcheezeyeah, there's one of those I think
11:23.49fourcheezeDataCompBoy: it's easy to build the zaptel modules if you have the kernel headeres
11:24.29DataCompBoyI know... but I have trying to stay with "less hands" on that box :D
11:25.55DataCompBoyo, asterisk I see there: http://ftp-master.debian.org/new.html
11:28.37fourcheezeyou can easily make a kernel package though
11:28.46fourcheezeonly takes a couple of minutes
11:29.03aymerichi, anyone using the channel chan_h233 in production environment ?
11:29.17fourcheezeDataCompBoy: I do it on one box and then reuse the package on other boxen
11:29.49DataCompBoyfourcheeze: i'm about "easy upgrade" :)
11:36.44*** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
11:37.42fourcheezeDataCompBoy: how often do you upgrade kernels?
11:38.08trixterI do it daily
11:41.33*** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net)
11:41.59DataCompBoynot often....
11:42.17nextimether's no app_sql_postgres in latest svn trunk?
11:42.50fourcheezeDataCompBoy: so its a 2 minute job not very often
11:42.58DataCompBoy:DD
11:43.00*** join/#asterisk p0g0_ (n=pogo@madwifi/support/p0g0)
11:43.04DataCompBoyabout once a month
11:43.23DataCompBoybut in production -- doesn't touch anything
11:43.23DataCompBoy:)
11:43.34DataCompBoyfor now I have upgraded from 2.6.12 to 2.6.15 for ztdummy
11:43.38Mavvieis there any listener-thread reserved space to store thread specific data?
11:48.59DataCompBoyhmmm... where is packages, that listed at http://ftp-master.debian.org/new.html ??
11:53.05*** join/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com)
11:55.41hertell_sleepingIs there a way in how to create an extension that would return me the PSTN dialtone on a spa-3000?
11:56.22*** part/#asterisk DataCompBoy (n=datacomp@217.8.236.1)
11:57.50*** join/#asterisk Eitch (n=hugo@unaffiliated/eitch)
11:58.13Hertellwhat i mean that if i hit 0 (without the trailing #-key) asterisk would pick up the line to my PSTN
12:01.10*** join/#asterisk Agur (i=raha@h82-131-120-51.fiber.ee)
12:06.45*** part/#asterisk jaike (n=a@203.131.137.76)
12:10.01*** part/#asterisk alexandrekeller (n=alexandr@mail.voffice.com.br)
12:10.52*** join/#asterisk enemy^x (n=eqwrweqr@morpheus.dataguard.no)
12:11.24enemy^xwhy is the 0 removed when dialing out? Is that in zapata.conf?
12:12.20Hertellenemy^x: what do you mean?
12:13.38*** join/#asterisk pengyong (n=lala@218.93.152.51)
12:14.49*** join/#asterisk voip470 (n=A_mail@pool-71-246-11-20.phlapa.fios.verizon.net)
12:19.18nextimeis there something like app MySQL() but for postgres?
12:19.39nextime( maybe as an external addons? )
12:20.08*** join/#asterisk fugitivo (n=ajf@201.255.176.13)
12:27.36*** join/#asterisk linville (n=linville@azure.tuxdriver.com)
12:28.34*** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net)
12:30.45*** join/#asterisk areski (n=areski@polar.es6.egwn.net)
12:34.58*** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
12:35.20*** join/#asterisk cypromis (n=michael@asterisk.pl)
12:37.26*** join/#asterisk Inkubot (n=inkubot@adsl-200-119-231-61.manquehue.net)
12:37.30Inkubothi
12:37.39Inkuboti've got a simple questions about a linksys PAP2
12:37.49Inkubothow can i enable that both FXS act as one ?
12:38.17I-MODjust put a phone line splitter on one port and use that
12:38.38Inkuboti can make call with both but when i recieve just one ring
12:38.47Inkubotok.
12:40.31I-MODor do you want it to ring both phones whenever a call comes in, but still be able to make two separate calls at the same time?
12:40.58I-MODcause i dunno about that one
12:41.05*** join/#asterisk RoyK (n=roy@80.239.107.70)
12:43.56Inkubotnah.. both phones ring when a call comes in, and only one way call out.
12:44.09{zombie}Inkubot: configure asterisk to dial both - eg Dial(SIP/101&SIP/102)
12:44.12Inkubotthats what i wan't.. and its only to know how to do it..
12:44.18Inkubotjust that.
12:44.25trixterwouldnt each port have to have its own account?  and when a call comes in you dial both extensions?
12:44.36trixteryeah like zombie said
12:44.37iDunnoyup
12:44.48Inkubot{zombie} thanks.. but this sip server it is not an Asterisk :\
12:44.57{zombie}Inkubot: well this is #asterisk
12:45.03Inkuboti know..
12:45.06[swb]anyone had a problem where you are using n(label) and Goto(label) and it doesnt work?
12:45.14{zombie}so if you ask questions you get asterisk specific answers.. sorry :)
12:45.18[swb]Its just "exiting non zero" and hanging up
12:45.23Inkubotbut you always know how to do this things..
12:45.31{zombie}I guess the answer is to install an asterisk box and use that to talk to your sip provider
12:45.36Inkubot{zombie} it is not a problem that :D
12:45.39{zombie}and hang the PAP2 off the asterisk box
12:45.46Inkubotyeps
12:46.55Inkubotall i want to know is how to do this..
12:47.08Inkubot:)
12:47.51{zombie}start by reading the asterisk docs, then installing asterisk
12:47.59{zombie}~docs
12:48.01jbotextra, extra, read all about it, 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
12:48.24Inkuboti know how to do that :P
12:48.33{zombie}ask specific questions then
12:48.43Inkubotthat is why i came here to ask..
12:48.56Inkuboti know that here are answer for that kind of stuff
12:49.06[swb]Hello, anyone with a problem of using n(label) and then a Goto(label) and asterisk just hangsup without doing the goto?
12:49.09{zombie}I gave you the answer to that
12:49.15Mavvieinclude/asterisk/logger.h asterisk.c cli.c logger.c
12:49.17{zombie}you just need to log the PAP2 both ports into asterisk
12:49.19Inkubotyeps.. and i apreciate :D
12:49.22Mavviewoops. no enter.
12:49.24{zombie}and register asterisk with your sip provider
12:49.28[swb]this is the wierdest problem I have seen
12:49.45Inkubot{zombie} ok
12:57.30*** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net)
12:59.53*** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl)
13:00.38*** join/#asterisk alexandrekeller (n=alexandr@mail.voffice.com.br)
13:00.50*** part/#asterisk alexandrekeller (n=alexandr@mail.voffice.com.br)
13:01.58*** join/#asterisk stone (n=stone@debian/developer/stone)
13:02.29*** part/#asterisk Inkubot (n=inkubot@adsl-200-119-231-61.manquehue.net)
13:03.23*** join/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m)
13:03.41*** join/#asterisk tdonahue (n=tdonahue@208.51.101.201)
13:06.53*** part/#asterisk cypromis (n=michael@asterisk.pl)
13:07.00*** join/#asterisk cypromis (n=michael@asterisk.pl)
13:11.33*** join/#asterisk p0g0__ (n=pogo@madwifi/support/p0g0)
13:11.49Mavvienew toy! http://bugs.digium.com/view.php?id=6524
13:13.00trixterpersonally I think each console should be able to set its verbosity level, there is afaik no way, even with that patch to turn down verbosity levels
13:13.33crusherI agree
13:13.50Mavvietrixter: the verbose logging in asterisk is a very tricky bussiness.
13:13.53*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
13:14.01trixterbut it doesnt have to be
13:14.16trixterthe cli stuff is tricky far more than it has to be, have you looked at the code for the cli?
13:14.24Mavvieit's not until the lowest levels that it finds the remote consoles.
13:14.40trixterfurther why is there a client and daemon in the same program?  I dont want firefox embedded into apache
13:14.58mutilatorit's not a client
13:14.58trixterI failed to see why it was acceptable to have both into one program such as it is with asterisk
13:15.15trixterit is, when you run asterisk -r it makes a connection over a stream pipe as a client
13:15.24trixteryet that client has all the code for the daemon, which largely it doesnt need
13:16.00trixterit doesnt need to know how to load modules because when in -r mode it wont do that, it doesnt need to know how to load channel drivers because again, with -r it wont do that
13:16.29Mavviesome well placed #ifdefs could generate an "asterisk-client" program.
13:17.00trixterand technically anything that speaks the correct protocol can open that stream pipe and talk to asterisk, so someone could write a master program that talks to both the stream pipe and the manager interface allowing for more robust client programs
13:17.22trixterwell it would take more than a few #ifdefs and they would have to be VERY well placed, such that it eliminates much of the code that goes into the asterisk binary
13:17.30*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:19.11[swb]has anyone had a problem using Goto and priority labels, it works fine all over my dialplan but in one app it just hangs up and refuses to perform the goto, I cant figure out why as it looks the same as all the places it is working
13:19.16[swb]I get 2006-02-17 13:17:13 WARNING[12005] pbx.c: Priority 'test' must be a number > 0, or valid label in the logs
13:19.30BugKhamI have some questions about dialplan
13:23.02BugKhamwe always jump to priority + 101 in case of error, is that right?
13:23.22BugKham1 -> 102 -> 203 -> 304 ...
13:24.55*** join/#asterisk ChrisUK (n=chris@82.108.126.170)
13:26.00ChrisUKHello :)
13:26.45fugitivoanyone using rxfax spandsp-0.0.2 with asterisk 1.2.4?
13:27.30fugitivo(spandsp-0.0.2pre25)
13:27.53ChrisUKAnyone else got echo problems on SIP to SIP calls on a local LAN? :S
13:28.05ChrisUKwith the Grandstream GXP2000
13:28.30*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
13:28.54trixterno I do that all the time and have no problems
13:28.59trixterwith a gxp2000 specifically
13:29.09[swb]ChrisUK, yes before I upgraded the firmware
13:29.21ChrisUKwhat version of firmware you using now ?
13:29.35[swb]I got fed up with the phone and sent it back, using SNOM 320 instead
13:29.48trixterI was running 1.0.1.9 upgraded to 1.0.2.3 had problems with neither
13:29.58ChrisUKlol ok
13:30.09ChrisUKwell im sort of stuck ive got 50 of them
13:30.14[swb]hehe
13:30.22[swb]I hear they are alot better on the later version of the firmware
13:30.30[swb]and that the customer service from Grandstream is good
13:30.33trixter1.0.2.3 isnt bad and has soem features that I felt were missing
13:30.34[swb]have you tried contacting them?
13:30.39ChrisUKNo not yet
13:30.47trixterlike telling you about missed calls and quickly letting you view those calls and call those people back
13:31.16[swb]so anyone had problems Using Goto and Labeled priorities in the dialplan
13:31.19[swb]its driving me up the wall
13:31.24*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
13:31.32[swb]I have used them all over with no probs but suddenly I just cant get one example to work
13:31.39[swb]the pbx thinks its an invalid label
13:31.49*** join/#asterisk JohnJacob (n=m00p@pool-71-246-132-74.aubnin.fios.verizon.net)
13:32.16[swb]I have the most simple example possible now and it still wont work
13:32.22[swb]the goto just exits and hangsup
13:34.34[TK]D-Fender[swb] : Pastebin it....
13:34.41[TK]D-Fender~pb
13:34.47jbotrumour has it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
13:35.45[swb]how much should I paste bin? I cant see it being a problem with the two lines as they are THE most simple, must be something else upsetting it
13:35.47[swb]pasteing now
13:36.06fugitivowow, irc is great with a console with the size of the screen
13:37.23[swb][TK]D-Fender, http://pastebin.com/559466
13:39.08*** part/#asterisk beefsalad (n=crash3m@unaffiliated/crash3m)
13:41.06docelm0fugitivo, huh?
13:41.45fugitivodocelm0: i was using a console of 10 lines for irc, now i use the hole screen
13:42.09fugitivowhole
13:43.02trixterit amazes me how many people dont clean their caller id before placing a call to a random provider
13:43.24crusherwhy would you?
13:43.43crusherthe provider overrides it anyhow
13:43.48trixternot all
13:43.55crushermy providers do
13:44.09trixterI only do if someone doesnt send me something valid, many people are sending their extension numbers which doesnt quite work
13:44.32mutilatorwhat;re the old t1 connector dealys calleD?
13:44.34Beirdoheh
13:44.36trixterbut I also only provide free access to tollfrees where people can send their own caller id which I convert to ani
13:44.51Beirdomade me laugh the first time I saw callerid of "1006" on my cellphone
13:44.59trixtermutilator: a rj48x jack?  or something else?
13:45.03crusherhehe
13:45.14mutilatori thought they were DB something
13:45.20crusherWell, I used to put the external number in callerid, but cleaned it out my extensions.conf yesterday
13:45.24Beirdo"oops, missed a spot overriding my callerid"
13:45.28mutilatorit's a serial style thing
13:45.31trixtermutilator: well I am not sure which connector you are talking about specifically
13:45.39*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
13:45.52trixterlike v.35?
13:46.01trixterlike what a csu/dsu would plug into?
13:46.11trixterthose didnt go away they just became integrated
13:46.22mutilatornop
13:46.37trixterI am not quite sure what you are  talking about then
13:46.59mutilatordb15
13:47.24trixterthat is just a 15 pin connector ...  didnt realize that was a t1 thing specifically
13:47.36trixternor would I have ever guessed that it was t1 specifically..
13:47.38crusherMy inbound voip provider actually sets callerid "asterisk"<asterisk> when an anonymous caller rings us.
13:47.39mutilatormay not be
13:47.49trixtercrusher: asterisk does that by default
13:47.53crusherhehe
13:47.59mutilatori just know i've only dealt with it on t's
13:48.08crusherokay, obviously they use asterisk behind their sipexpress
13:48.29trixterare you sure its not your box that is doing that by default?
13:48.42trixtermine does only because I am too lazy to change it
13:49.08crusherwhere would it be set then? while handling the extension it's already set to that
13:49.19*** join/#asterisk fjean (n=fjean@201.29.122.10)
13:49.32crusherI shall test it this evening by connecting a sip phone directly
13:49.51mutilatorfound what i was lookin or
13:49.53mutilatorfor
13:49.54mutilatorhttp://www.trianglecables.com/db15f-rj45.html
13:49.56trixternot that it matters that much, but it would let you know ...
13:50.16crusheralright
13:50.17mutilatorer i need male
13:50.40mutilatorthere it is
13:50.52mutilatoreh sry typin outloud
13:50.54fjeanHello ! Anybody uses ASTSS and understands the "adjusted time" formula ?  adjtime = (((answertime - inc) + inc - 1) / inc) * inc
13:50.58fugitivo~seen coppice
13:51.07jbotcoppice <n=chatzill@199.193.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 5d 5h 19m 34s ago, saying: 'but it is the one people complain about the most :-)'.
13:51.07mutilatori think i had too much coffee
13:51.27fjeansorry not ASTSS but ASTCC
13:52.43fjeanShould I expect multiples of 6 as a result, if not, why ?  :- )
13:53.20fjeanhere is the code:  eval { $adjtime = int((($answeredtime - $numdata->{includedseconds}) + $increment - 1) / $increment) * $increment;
13:53.31[TK]D-Fender[swb] : Ok... I'm stumped....
13:55.46trixter[TK]D-Fender: did whomever stump you at least buy you dinner first?
13:56.00fugitivo[TK]D-Fender: are you using rxfax?
13:56.09X-Robfjean, isn't that saying answertime = answertime?
13:58.44X-RobNo, it's saying $adjtime = $answerdtime - $numdata->{inc..} + increment - 1
14:00.22fjeanin my case:  adjtime = int(((50 - 30) + 6 - 1) / 6 * 6;
14:01.01fjean50=talk time and 30=included seconds
14:03.03fjeanadjtime = 25 ...
14:03.25X-RobNo, it's not / 6 * 6
14:03.44X-RobIt's ) / 6 ) * 6)
14:03.57X-Robif it was / 6 * 6, it would actually be / 36
14:04.22fjeanmmmmm
14:04.32X-Robbut being that it's / 6 ) * 6), it's useless.
14:04.34X-Robyou can take it off
14:04.48X-Robso you're doing 50-30 + 6 - 1
14:05.04X-Robeg, 25.
14:05.32fjeanmmmm,cool
14:06.30fjeanlet me see...
14:06.55X-RobHowever, if they're casting int's in strange places, that's not quite right.
14:07.17*** join/#asterisk PakiPenguin (n=bah@linuxpakistan/admin/pakipenguin)
14:07.19PakiPenguinhello everyone
14:07.53*** join/#asterisk skeffling (n=Andrew_H@andrew.1ec.aaisp.net.uk)
14:08.08X-Robif it's (int (50-30+6-1)/6)*6, that's not the same, as the 25/6 will be rounded to 4, and then multipled back to 24, not 25.
14:08.08PakiPenguini'd like to ask if its better to get the Digium TDM2422B PCI Card with echo cancelation or without echo cancelation ? I'd use this card in a dual xeon machine , that would do nothing else then handling these calls
14:08.15*** join/#asterisk asteriskmonkey (n=phil@69.156.197.242)
14:08.31mikefoohey guys in general, if I have a working system, would there be any significant reason I would want to upgrade at any point in time?
14:08.38X-RobPakiPenguin, get Echo Cancellation hardware if you can afford it. Always.
14:08.40mikefooother the security issues and such.
14:08.46mikefoos/the/then
14:09.07skefflinghi,is there anywork being done to have a xfersound feature for call transfers on sip channels?
14:09.19crushermikefoo: no, or maybe new features you'd like
14:09.33iCEBrkryo yo yo
14:09.46[TK]D-Fenderfugitivo : Yes I run SpanDSP
14:10.22PakiPenguinX-Rob, if i cannot , can the cpu make up for echo cancelation?
14:11.10[TK]D-FenderPakiPenguin : As far as EC on the TDM2400 goes, if you follow the mailing lists it seems that it doesn't seem to offer much (yet) and is moderately buggy.... not sure if you're better off just stay in SW.  # of channels may have an impact...
14:11.30zoahey ho!
14:11.31X-RobPakiPenguin, not anywhere near as well.
14:11.34fugitivo[TK]D-Fender: what version of spandsp and asterisk?
14:11.39zoamorning, you fool! its almost evening!!!!
14:11.40zoa:p
14:12.18PakiPenguinX-Rob, would i be better off using the TDM 4X series?
14:12.19[TK]D-Fenderfugitivo : 1.0.9 CVS and not sure of SpanDSP
14:12.20asteriskmonkeyits 6am always somewhere
14:12.42[TK]D-FenderI'm upgrading EVERYTHING tonight.  PRI card F/W, drivers, *, Zaptel, GUI, the works....
14:13.23asteriskmonkey[TK]D-Fender : beware youll have to recomile your spandsp for 1.2 the 1.0.9 one wont work
14:13.30[TK]D-FenderPakiPenguin : Who many FXO/FXS ports do you need?
14:13.37fugitivo[TK]D-Fender: ok, because i upgraded spandsp and i'm getting errors with rxfax now
14:13.53[TK]D-Fenderasteriskmonkey : Well aware, and my vendor is the one managing the source revisions :)  I'm just doing the hardware half/
14:14.11asteriskmonkeynice
14:14.18PakiPenguin[TK]D-Fender, 6FXO and 6 FXS
14:15.47[TK]D-FenderPakiPenguin : Well TDM400 is clearly NOT the way for you to go.  Your options are TDM2400, A200, or T1 + Channel-bank.
14:16.12[TK]D-FenderWhy FXS on the card?
14:16.28PakiPenguinhmms one card solution :)
14:16.33fugitivo[TK]D-Fender: why not a sip gateway for fxs?
14:17.03fugitivoan audiocodes with 8fxs
14:17.19PakiPenguinyes , i can think about that too
14:17.49X-RobPakiPenguin, that's hugely expensive. Do what you can afford to do.
14:18.15*** join/#asterisk bas123 (n=bas@www3.datarack.nl)
14:18.18bas123hellp
14:18.23bas123hello, that is
14:18.26bas123:)
14:18.27[TK]D-FenderPakiPenguin : For FXS you could save a bundle and use ATA's.  its only 3...
14:18.30fugitivoi don't think the audiocodes is more expensive than the same solution with the tdm2400
14:18.32*** join/#asterisk saftsack (n=saftsack@p54A7DF79.dip.t-dialin.net)
14:18.46bas123anybody using polycom Soundpoint 600 phones?
14:18.50PakiPenguinyes , i was talking about the atas
14:19.02[TK]D-FenderPakiPenguin : the A200 solution would cost less than the TDM2400 for your needs while still being expanable.
14:19.06[TK]D-Fenderbas123 : Yes
14:19.12fugitivothe problem with the atas is that you'll have 3 more devices to admin
14:19.17bas123@TKD-fender
14:19.25*** join/#asterisk fishboy1669 (i=proxyuse@62.69.81.129)
14:19.28bas123have you ever had problems update the bootrom?
14:19.50[TK]D-Fenderfugitivo : True, but a fraction of the cost but that balances as well against the increased functionality they offer over TDM withing *.  Simplifies setup and reliability
14:19.57[TK]D-Fenderbas123 : Never a problem with them.
14:19.59bas123I am doing a new install at a customer site at the moment, and half of them refuse to load the new bootrom
14:20.20[TK]D-Fenderbas123: You should leave the BR at 2.6.1 if I were you....
14:20.26bas123"error updating bootROM"
14:20.28[TK]D-Fenderjust upgrade SIP.
14:20.40*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:21.13PakiPenguinhmms yeah
14:21.32bas123[TK]D-Fender: we have multiple installs and like to keep them on the same version. But your answer makes sense
14:21.44PakiPenguinbrb a sec
14:21.51[TK]D-Fenderbas123 : How do you provision them currently?
14:22.22bas123[TK]D-Fender: with scopserv
14:22.36asteriskmonkeyremember you always get what you pay for though ata's vs proper fxs gateways can be night and day in terms of quailty sometimes
14:23.02asteriskmonkeylike compare the iaxy to a audiocodes fxs gateway..
14:23.28[TK]D-Fenderbas123 : I meant what protocol (ftp/tftp (what I'm betting on for your case), http)
14:23.35mikefoowe have a few audiocode fxs/fxo gateways here
14:23.43bas123[TK]D-Fender: sorry, tftp it is
14:24.44[TK]D-Fenderbas123 : Leave them at 2.6.1 if you can and just up the SIP revision then.  The new BR is not needed for your scenario.  I'm running 1.5.3. here personally and am quite happy although I have every version handy from 1.4.3 up.
14:25.28[TK]D-Fenderincluding the BR.  I just got 1.6.5 yesterday and am following the changelogs on them regularly.  As soon as they offer something I really need or fix a bug I've encountered here, THEN I'll switch
14:25.29bas123[TK]D-Fender: The phones are loading the new SIP version now, looks good.
14:25.56[TK]D-FenderI'll have to take a look and see how ScopServ's provisioning has come along....
14:26.26bas123[TK]D-Fender: I've found the guys at scopserv very help helpful, but then again, we're reselling it too :)
14:26.45[TK]D-Fenderbas123 ..... I think we may have scrossed paths before.... Whats your real first name?
14:27.06X-Robor sab
14:28.02[TK]D-Fender<- Andrew Oulton (Montreal, QC).  I've been on a conference call with someone on your side before with Denis Trepanier over echo/fax issues.....
14:28.20[TK]D-FenderAbout 2/3 months ago
14:28.41iCEBrkr[TK]D-Fender: you're a hoser??!!?
14:29.08fourcheezeshould a user / peer specify how it wants dtmf?
14:29.12*** join/#asterisk NewSole (n=dave@d226-105-29.home.cgocable.net)
14:29.12ManxPoweriCEBrkr, perhaps you mean Canuck.  Hosers are from Indianan
14:29.16fourcheezeI mean can there be any negotiaition
14:29.24fourcheezeor is it just up to everyone to be using the same?
14:29.32*** join/#asterisk bas1234 (n=bas@www3.datarack.nl)
14:29.33iCEBrkrManxPower: Apparently you haven't watched Strange Brew, eh?
14:29.51bas1234[TK]D-Fender: sorry, my collegue tightened the firewall while troubleshooting VPN issues
14:29.55ManxPowerfourcheeze, you didn't get an answer to your question on asterisk-dev about rfc2833 DTMF and LEvel3?
14:29.56bas1234the bastard :)
14:29.57ManxPoweriCEBrkr, nope.
14:30.06NewSolestupid question anyone have experinance with sipra and asterisk
14:30.10iCEBrkrManxPower: It's a classic 80's movie
14:30.27fourcheezeManxPower: we didn't go with L3 :-)
14:30.38[TK]D-FenderiCEBrkr : Hoser?
14:30.39NewSolewe we call the sipra device it sends back busy
14:30.57ManxPowerNewSole, SIPa work fine.
14:31.05*** join/#asterisk GMsoft (n=gmsoft@gentoo/developer/gmsoft)
14:31.07iCEBrkr[TK]D-Fender: Yeeeesh! You haven't seen Strange Brew, either?!
14:31.13iCEBrkrYoung'ns.
14:31.16GMsofthi everybody
14:31.31NewSolewhy am I getting busy back
14:31.52[TK]D-FenderiCEBrkr : I'm older than you :)
14:31.54iCEBrkrNewSole: You can set verbose 9 at the CLI and watch the console
14:32.05iCEBrkr[TK]D-Fender: Maybe :P
14:32.13NewSolekind of hard
14:32.14fourcheezeManxPower: althouhg L3 inthe UK claim to use rfc2833
14:32.20[TK]D-Fender30 <-
14:32.22iCEBrkrNewSole: Why's taht?
14:32.25NewSolewe have over 100 calls going at once
14:32.26iCEBrkr[TK]D-Fender: 31 yo
14:32.27[TK]D-FenderiCEBrkr : Why start now?
14:32.30[TK]D-FenderLIES!
14:32.31iCEBrkrlol
14:32.36iCEBrkr[TK]D-Fender: 1974
14:32.42GMsoftis there a documentation regarding IAX bandwitdh calculation ?
14:32.50iCEBrkr[TK]D-Fender: and I'll be 32 this sept.
14:32.54[TK]D-Fenderok, I'll bet you've got months at best on me... I've got MILAGE!
14:33.01iCEBrkrlol
14:33.12NewSoleI was just wondering if someone has sample settings
14:33.14iCEBrkr[TK]D-Fender: Rode hard and hung up wet?
14:33.28[TK]D-FenderiCEBrkr : Think I saw that movie ;)
14:33.47iCEBrkrhaha
14:34.05iCEBrkrNewSole: I'm thinking there's something wrong with your sip.conf
14:34.31NewSolethats what I was wondering... its nat based
14:35.03iCEBrkrI wonder were my good friend sevard is?
14:35.24iCEBrkr>= )
14:36.32NewSoleiCEBrkr... anyone can help with device config
14:36.39zoaim here but i am disguised as zoa
14:36.48*** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net)
14:36.52zoaplease give me your credit card number now
14:36.57zoai forgot to write it down before
14:36.58[TK]D-Fenderbas1234 : Yeah, looks like ScopServ has improved their provisioning of the SPIP's lately.  How well does it handle multiple line keys/reg?  It LOOKS like it'll create a clash if you pick the same ext more than once..
14:37.02*** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com)
14:37.05iCEBrkrNewSole: I think you're going to have to watch the console
14:37.26iCEBrkrzoa: You need my paypal and ebay account info too?
14:37.38iCEBrkrzoa: Just to verify my info is correct..
14:37.41mikefooiCEBrkr: sup sup
14:37.44iCEBrkrmikefoo: yo
14:37.50NewSole<PROTECTED>
14:37.50NewSole<PROTECTED>
14:37.50NewSole<PROTECTED>
14:37.50NewSole<PROTECTED>
14:37.51NewSole<PROTECTED>
14:37.53NewSole<PROTECTED>
14:37.54iCEBrkrNewSole: dude.
14:37.56WinkieI'm trying to set up a queueing system, but i am a little confused, i'd like more than one call to ring in at once, because we have say 30 agents and there are always 6 or 7 calls going on at the time
14:37.57iCEBrkrNewSole: ~pb
14:38.00Winkieany ideas?
14:38.00iCEBrkr~pb
14:38.01jbotit has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
14:38.45NewSolehttp://pastebin.com/559554
14:38.50*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
14:38.51Winkiealso leastrecent doesn't seem to work, it repeatedly times out and restarts the queue, or so it seems, am i doing something stupid?
14:39.23Winkieor would using agents help instead of members?
14:40.35[TK]D-FenderWinkie : pastebin your queues.conf and the extensions.conf context that calls it
14:40.50[TK]D-FenderWinkie : and what was the last thing you changed before it stopped working?
14:40.51Winkie[TK]D-Fender: well at the moment there's virtually nothing there
14:40.55Winkieand it's not stopped working
14:41.00[TK]D-FenderWinkie : Show anyways
14:41.02NewSoleiCEBrkr... any idea
14:41.03Winkieno worries
14:41.05Winkieone min
14:41.25remiss<PROTECTED>
14:41.31remisswhy is that?
14:41.32*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
14:41.48*** join/#asterisk klerer (n=klerer@ool-44c72037.dyn.optonline.net)
14:42.05iCEBrkrNewSole: 'sip show peers'  and see if it's registered
14:42.09[TK]D-Fender~pb
14:42.11jboti guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
14:42.12NewSoleit is
14:42.40ManxPower"seeding peer" seems to indicate that the peer might not be registered.
14:43.27*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
14:43.55Winkie[TK]D-Fender: http://pastebin.ca/41941
14:44.07ManxPowerNewSole, well what is the value of DIALSTATUS after the Dial?
14:44.48Winkie[TK]D-Fender: effectively we need queues to push as many calls to phones as possible without waiting for the phones to pick up
14:45.01Winkieas we handle far too many calls to make a single call ringing at a time feasable
14:45.43Winkiethis is obviously just a test and a proof of concept but it seems weird that asterisk queues must be only one at a time
14:46.03[TK]D-FenderWinkie : not sure how to increase the number of simultaneous ringing calls.... does it abort out of queue early?
14:46.23Winkie[TK]D-Fender: not as far as i'm aware, it times out, and re-rings the same phone instead of alternating
14:46.31Winkielet me quickly check
14:47.10[TK]D-FenderI think because you are calling the queue with 5 seconds the first time with a timeout of 10 it never gets to 10 and then re-rings the same phone....
14:47.52*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfljo.dialup.mindspring.com)
14:47.53Winkieno it definately makes it past the first queue
14:48.27Winkiehttp://pastebin.ca/41942
14:48.34*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:48.34*** mode/#asterisk [+o anthm] by ChanServ
14:50.22*** join/#asterisk paulhuynh (n=paul@c-68-37-18-82.hsd1.de.comcast.net)
14:50.48paulhuynhi need little help today can someone help me!!!
14:50.56Winkiejust ask
14:51.13[TK]D-FenderActually it may keep trying to beat down that same guy if he's listed as available... you need to get them to pause or log off....
14:51.15paulhuynhMy asterisk@home iso2.4 is not taking call from from my sip provider
14:51.27wunderkinWinkie, make sure you are using a recent 1.2 release branch and see if it has the autofill option, use it
14:51.53Winkiewunderkin: 1.2.4, autofill in queues.conf and i assume that's for multiple rings?
14:51.57Winkiemultiple incoming rings*
14:52.00wunderkinim not sure if it was put into 1.2 or not but if not you may be able to patch it
14:52.21Winkiei can use CVS if neccasary, it's a test system, but obviously stability will be important when it goes live
14:52.48wunderkinthat helps fix some of the queue quirks but i dont remember what all
14:52.55Winkiealso [TK]D-Fender you reckon it's just a lack of agents? I mean leastrecent does say it will call the 'interface' least recently dialled, which wouldn't be linksys
14:52.57*** join/#asterisk burton (i=mimx@w201.ljudmila.org)
14:53.21Winkiepaulhuynh: who is your sip provider / what does sip show registry produce (use pastebin)
14:53.33Winkiewunderkin: i will google my way to finding out, ta
14:53.34[TK]D-FenderWinkie : I believe its not the least recently CALLED, its the least recently ANSWERED/.
14:53.44paulhuynhwinkie i use cvcternimation
14:53.53[TK]D-Fenderpaulhuynh .....
14:53.53NewSolethe DIALSTATUS is CONGESTION
14:53.54[TK]D-Fender~amp
14:53.56jbotfrom memory, amp is NOT supported here! people using it should join #amportal
14:53.57burtonhello, anyone has working setup with two wifi AP + asterisk + SIP wifi phones ? how about hand over roam ?
14:54.40Winkie[TK]D-Fender: ah, from voip-info it says 'leastrecent: ring interface which was least recently called by this queue' but that may be inaccurate
14:55.04[TK]D-FenderI'd bet on it...
14:55.06wunderkinpaulhuynh, oh yeah? have you used them yet, what do you think? i was thinking about using them, they are just a few racks down from me.. i don't have an account yet so i can't really help setting it up yet
14:55.20[TK]D-Fenderminor difference that makes all the difference...
14:56.28Assidheya tkd
14:56.29*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
14:56.31Assidwhats happening
14:56.48*** join/#asterisk Teeli (i=Tili@202-133-67-129-dialup.sat.net.pk)
14:56.51[TK]D-FenderAssid : just working on stuff... tonight is my works big * upgrade.
14:56.53Winkie[TK]D-Fender: indeed, also autofill and autopause, which seem to be the two interesting ones were committed in revision 8060 4 weeks ago, but i have no idea how to find out if that was included in 1.2.4
14:57.05Assidreally? what u doing in it?
14:57.15*** join/#asterisk littlejohn (n=little@host130-254.pool8263.interbusiness.it)
14:57.17[TK]D-Fenderauto-pause?  Pauses static members for not answering (like auto-logoff?)
14:57.20wunderkinWinkie, i would just use the latest release branch
14:57.23Winkie[TK]D-Fender: indeed
14:57.32paulhuynhhere are my pastebin for the sip reg
14:57.33paulhuynhhttp://pastebin.ca/41945
14:57.34Assiddidnt get you. how does that help
14:57.35Winkiewunderkin: it looks like i have to, what's asterisk's release schedule like?
14:57.36[TK]D-FenderWinkie : definately good news... I run a call center here as well..
14:58.09Assidnice... i always wanted to do atleast 1 solution for a call center
14:58.11Winkiepaulhuynh: i don't know your provider, do they require sip registration or what?
14:58.28Assideven if its around 20 users... just wanted to play with that
14:58.32Winkie[TK]D-Fender: fun isn't it? we run about 30 agents, 6000 calls a day or so
14:58.38wunderkinWinkie, not sure, it may be on the website, i think the next release is scheduled in the summer sometime
14:58.39mikefooCan ayone recommened a voip provider I can terminate on that offers concurrent calls?
14:58.40[TK]D-FenderAssid : I'm going from 1.0.9 to 1.2.4 and upgrading PRI f/w drivers, NIC drivers, and more.
14:58.41mikefooin the US
14:58.42Winkiewell i'd say nearer 8000, our phone system is holding us back
14:58.46[TK]D-FenderWinkie : WOW....
14:59.01Winkie[TK]D-Fender: how many do you handle?
14:59.14Winkiewe're looking at setting up 4 locations total with asterisk and interlinking them with leased lines + IAX trunking
14:59.16Winkiewhich will be fun
14:59.23paulhuynhmikefoo try sixtel.net or cvctermination
14:59.33mikefoopaulhuynh: thanks.
14:59.46paulhuynhwinkie this they do
15:00.09Winkiepaulhuynh: what does 'sip show registry' tell you?
15:00.27paulhuynhthat they are register
15:00.28[TK]D-FenderWinkie : 4 agents, harly more than 3 in Q at any time...
15:00.34Assidis there any perfomance hit when you have sip clients and iax interconnects?
15:00.36[TK]D-Fender(backloged)
15:00.47Winkie[TK]D-Fender: heh, we barely make 3 calls waiting unless it's really busy
15:00.55paulhuynhi contact my provider but they tell me my asterisk box is reject their call that was forward to us
15:01.01Winkieit goes from 1 guy alone on a sunday night to about 13-14 logged on at the moment
15:01.12Winkiepaulhuynh: pb your dialplan
15:01.17paulhuynhi can make call to pstn from my asterisk to though their network
15:01.20Winkieor at least, the from-pstn context
15:01.37[TK]D-FenderWinkie : He's running A@H, don't ask that!!!
15:01.46Winkieoh right
15:01.51Winkiei know nothing of this rubbish :)
15:01.57wunderkinpaulhuynh, i would look at a sip debug, you probably don't have  your dialplan correct
15:01.59[TK]D-Fenderpaulhuynh : Pastbin the FAILED CALL from CLI only please.
15:02.31wunderkinthere also aren't any codecs specified there..
15:02.41paulhuynhok can you walk me through the cli command
15:03.06*** part/#asterisk mhnoyes (n=mhnoyes@user-2ivfljo.dialup.mindspring.com)
15:03.44PakiPenguinokay i am back
15:04.07Winkie[TK]D-Fender: the interesting thing is getting decent quality voice to canada
15:04.11PakiPenguin[TK]D-Fender,  what card do you suggest for incoming pstn ( FXO ) ? 6 lines , i am going to use ata for extensions
15:04.19Assidhey is there any perfomance hit when all your phones use sip.. and iax to interconnect.. or is it the same as sip providers instead of iax since the sip layer is the only one invoked
15:04.20Winkiei can't wait to see how much our provider's going to charge
15:04.47WinkieAssid: SIP and IAX are essentially framing, performance hits occur with transcoding afaik
15:04.55Winkieso as long as you're using the same codecs, you shouldn't have much of a problem
15:05.17*** join/#asterisk hugo-v6 (n=alterego@ns1.bundesunixminister.de)
15:05.20hugo-v6hiho
15:05.24Assidsome one told me that there is a hit even in the carrier method (iax/sip) i was like no way.. transcoding would be the only thing
15:05.45Assidbut then i started reading about SER.. where the sip layer just interconnects the sip clients
15:05.55Assidso i started thinking maybe there was something to it
15:05.57Winkiethere will be *some* performance difference but i'll emphasise it because i don't think we're talking serious differences
15:05.57fjeanx-rob:  would you know what "included seconds" means ? is it meaning the customer will get this number of seconds free ?
15:06.05*** join/#asterisk Cresl1n (n=matt@146.229.178.19)
15:06.25Winkiealso if anyone has experience in transcoding from an E1 to ulaw, let me know the stats :)
15:06.46Winkieanyway i'm off for a ciggy, thanks [TK]D-Fender and wunderkin, i'll CVS it later and try
15:06.49AssidE1 to ulaw?
15:07.01AssidWinkie: CVS is dead.. gotta use SVN
15:07.03Winkieyes, as in ISDN PRI > SIP ulaw
15:07.04[TK]D-FenderPakiPenguin : A200 w/ HWEC
15:07.07Winkiewell CVS, SVN
15:07.10Winkiethey're all TLAs to me
15:07.12Winkie8)
15:07.13Winkieciao
15:07.14PakiPenguincool , i was looking into that
15:07.15PakiPenguin:)
15:07.19ManxPowerWinkie, E-1 uses alaw
15:07.21PakiPenguinthanks , i'll be back with more questions hehe
15:07.26ManxPowerthat's all yu really need to know
15:07.32WinkieManxPower: oh really, well that will certainly simplify speccing the servers
15:07.37Winkieno dual x2s needed here then
15:08.04fjeanhi guys, anybody knows ASTCC and it's associated DB fields well ?
15:08.09paulhuynhthis is what i got when do a sip show registry
15:08.09ManxPoweralaw is the telco standard for USA/Canada (and maybe a few other countries), alaw is what the rest of the world uses.
15:08.10paulhuynhhttp://pastebin.ca/41946
15:08.17ManxPowerSo you want your SIP clients to use alaw.
15:08.40*** join/#asterisk bkw_ (n=bkw_@ppp-70-128-113-60.dsl.tulsok.swbell.net)
15:08.49Assidhrmm.. is alaw any 'clearer' than ulaw?
15:09.01ManxPowerAssid, I don't think so, just slightly different
15:09.09Assidin framing?
15:09.17Ahrimanesboth uncompressed, ulaw mainly used in the us and alaw in europe
15:09.26ManxPowerAssid, I would have to read the codec specs to say for sure.
15:09.27Assidhrmm
15:09.36Assidwell.. ulaw works.. i aint touching it
15:09.56ManxPowerAssid, you should usually generally just use the codec that your location uses.
15:10.20fugitivodoes the t110p work on a pcix slot?
15:10.21viperdudehi has anyone here provided a operator console for a solution?
15:10.29Assidwell.. mostlye looking at US anyways.. so really just following that standards
15:10.33viperdudei am looking for hardware not FOP
15:10.58Assidfugitivo: i thought pci-x is for the gfx card only
15:11.55Assidi need to learn more
15:12.09wunderkinfugitivo: yes
15:12.19fugitivohmm, why would a motherboard server come with 2 slots for gfx cards? :)
15:12.28fugitivowunderkin: great, thanks
15:12.44Assidpci-express and agp ?
15:12.46Assidlol
15:12.52Ahrimanespci-x is for all sorts of pci cards
15:13.04*** join/#asterisk kpettit (n=keith@69.15.174.114)
15:13.18Assidman.. that teaches me for not touching hardware in a year
15:13.26Assidexcept for old ones
15:14.00[TK]D-Fenderviperdude : as in to see who's on the phone and to do speed-dials?
15:14.06PakiPenguina question , would i need anything else , like a rj-11 connector panel or anything with the sangoma a200 with 6 FXO ports?
15:14.52[TK]D-FenderPakiPenguin : A200 uses RJ12 (handset style) and comes with adapter extensions.  What you'll need from there if anything depends on what you've got.
15:14.58viperdude[TK]D-Fender: yes, but also to be able to send text to to tell the operator certain info.. eg if a camped call gets returned the extension they were camped on
15:15.03[TK]D-Fenderif you're on punchdown you'll need to adapte from that
15:15.04NewSoleok anyone here good with SER
15:15.09Assidbrb
15:15.48[TK]D-Fenderviperdude : For the BLF stuff, Snom is the only current option (Polycom will be viable hopefully as of * 1.4)  as for parking indicators, not sure....
15:16.00viperdudeBLF?
15:16.11paulhuynhis anyone here use sixtel for did?
15:16.15PakiPenguin[TK]D-Fender,  i have standard pots lines ( with rj-11s at their ends ) , do i need something else then?
15:16.16[TK]D-FenderBusy Lamp Field : (in use indicator / speed-dial)
15:16.30[TK]D-FenderPakiPenguin : nope, all good out of the box...
15:16.35NewSoleok anyone here good with SER want a job... pvt me
15:16.39viperdudeok that works currently on Snom with * 1.2.1?
15:17.05[TK]D-Fenderviperdude : yup, and Snom is cheapr than the rest once you add on the sidecar.
15:17.18[TK]D-FenderAnd thats a LOT of buttons...
15:17.20Assidsnom cheaper than polycom ?
15:17.22PakiPenguincool
15:17.36*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
15:17.51viperdudedoes anyone know if Cisco 7940 support SIP MESSAGE?
15:17.57[TK]D-FenderAssid : Slightly.  but which one I'd choose depends on the application.  Typically I'd pick Polycom over everything else EXCEPT high-density receptionist.
15:18.07*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
15:18.13Assidi see
15:18.18[TK]D-Fender(best of breed selection)
15:18.31Assidthanks for the insight
15:18.36Assidwill keep it in mind
15:18.45Assidany particular model?
15:18.57jaiger[TK]D-Fender, have you looked at the add-on panel for the IP600?
15:19.00Ahrimanes[TK]D-Fender: any integration done for function buttons on polycom?
15:19.01[TK]D-FenderCisco is a great phone as well, but not for the money, and the higher end stuff supporting the sidecar doesn't do SIP yet.  So Polycom will likely beat them out for a while yet... then things may become interesting.
15:19.03*** join/#asterisk austinnichols101 (n=austinni@70.46.69.130)
15:19.23viperdudeok its just I have a cisco 7940 on my desk
15:19.31[TK]D-FenderAhrimanes : They're working on the ACD login/out/pause right now.  As for function buttons what else did you have in mind?
15:19.44[TK]D-Fenderjaiger : I have 2 on our 601
15:19.46Ahrimanes[TK]D-Fender: ACD ?
15:19.53[TK]D-FenderAhrimanes : Queues'
15:20.04jaiger[TK]D-Fender, and how does it work?
15:20.09Ahrimanes[TK]D-Fender: ah, nice.. that was most of what i had in mind.. call pickup would be nice too
15:20.29Assidhey tkd how big is this call center you run?
15:20.47jaigerI was thinking of buying one to play with
15:20.53[TK]D-Fenderjaiger : It works up until the 7-buddy watch bug....
15:21.14Ahrimanes[TK]D-Fender: know how far along the ACD stuff is with the polycoms?
15:21.23jaiger[TK]D-Fender, so what would you recommend as an alternative?
15:21.28paulhuynhmay i ask ACD?
15:21.40[TK]D-FenderAhrimanes : You can program that as a speed-dial on a line-key, as the Messages button or on one of the 2 other "loose" buttons on the 501
15:21.57[TK]D-FenderAhrimanes : so far it looks like this summer (* 1.4)
15:22.17Ahrimanes[TK]D-Fender: ok
15:22.23[TK]D-FenderAssid : 4 agents
15:22.53nextimeanyone with chan_ooh323 with latest svn trunk?
15:23.16[TK]D-Fenderpickup can be done as I just wrote above, OR once * supports SIP-B BLA draft, shared line appearances will make it look seamless.
15:24.35Ahrimanes[TK]D-Fender: yeah i've done it with speed dials on the snom190 function buttons.. but would like to leave a button lit if the queue is active or something like that
15:25.32NewSoleok anyone here good with SER and want a job... pvt me
15:25.39Winkieah man it's goddamn freezing outside, anyway an additional question, what's the easiest way to initiate a call on a gxp2000 from the manager interface? is it quite simple?
15:26.03viperdudeWinkie: "Action: originate"
15:26.22freat[TK]D-Fender: you know if Polycom has any plans on encryption support?
15:26.22[TK]D-FenderAhrimanes : Then use BRISTUFF and devstate with a script that periodically checks the queue to toggle the light.
15:26.35Winkieviperdude: i mean an outbound call to a zaptel channel but originating from a free line on a gxp2000
15:26.43Winkieor perhaps that's what you meant too :)
15:26.44freatwe've got a bunch of ip500s... in health care... HIPAA...
15:26.51[TK]D-Fenderfreat : Likely... it supports HTTPS and SFTP for provisioning so SRTP can't be too far behind.
15:27.00*** join/#asterisk Utah_Dave (n=boucha@0-1pool138-37.nas28.salt-lake-city1.ut.us.da.qwest.net)
15:27.00paulhuynhi'm on asterisk 1.2.1
15:27.15paulhuynhcould that be a problem for my did?
15:27.16viperdudeyou mean a grandstream GXP-2000
15:27.30Winkiei do viperdude
15:27.44Assidhey anyone know how much voicepulse charges for adding more chanlimits for incoming?
15:27.54Winkiealso i'll brb, i have to set up a windows share, laffo
15:28.13viperdudeso yes you do a 'Action: originate' on the manager ..see wiki for details, we use this for our click to dial on our intranet
15:28.27paulhuynhanyone?
15:28.35paulhuynhhelp please!!!
15:28.37paulhuynhi'm on asterisk 1.2.1
15:28.39paulhuynhcould that be a problem for my did?
15:28.53viperdudewhats the problem paul?
15:29.14Ahrimanes[TK]D-Fender: i tried devstate, but was never able to turn off the lights..
15:29.28[TK]D-Fenderonly ON?
15:29.47Ahrimanesyep
15:29.50paulhuynhDID from two of my carrier is not making it to asterisk
15:29.53paulhuynhbox
15:30.09wunderkinpaulhuynh, you need to do a sip debug
15:30.11paulhuynhcarrier said that my asterisk is drop the call from them
15:30.17[TK]D-Fenderpaulhuynh : is your * behind a NAT?  If so did you configure it accordingly and forward the needed ports?  You also never pastebin'd that failed call like requested.
15:30.21wunderkini always make them a peer and do sip debug peer blah
15:30.33viperdudepaulhuynh: sip debug is your friend
15:30.46paulhuynhok
15:30.58paulhuynhi got it but it flashing through alot of stuff
15:31.13paulhuynhhow can i just show sip debug for the incoming call
15:31.30Ahrimanes[TK]D-Fender: have you had devstate working turning leds on/off?
15:31.41NewSoleok anyone here good with SER and want a job... pvt me (auto timer)
15:31.43[TK]D-FenderAhrimanes : Haven't tried yet... soon...
15:31.52viperdudesip debug peer <exten>
15:32.52Ahrimanes[TK]D-Fender: please do let me know if it succeeds
15:32.58Winkieviperdude: yeah that seems to be what i need, the question is if i originate a call, how does it interact with the gxp, does it just appear as a second line, or does it ring on it? and how would i transfer one line to another via the manager interface?
15:33.24Winkiealso i need to turn LEDs on and off on the gxps too :(
15:33.32viperdudei have written a win32 app that shows the status of the exten's, busy, ringing, DND etc, sounds like that is a good start
15:33.44*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
15:34.11viperdudeWinkie: if you originate then the gxp will ring... when the handset is picked up then it dials out the number required
15:34.29Winkieviperdude: excellent, and i assume if they're already on a call it will ring on a second line?
15:34.36viperdudeyes
15:34.37Winkiei'll test it as soon as i have a chance but i'm swamped with other crap at the moment
15:34.48Winkieok, and is there a way to easily transfer the two without using the actual phone?
15:34.49paulhuynhi got it
15:34.51Winkie(long story)
15:34.55paulhuynhhere is my pastebin
15:35.02viperdudeif you want to transfer a call via the manager then use "Action: Redirect" again see the wiki
15:35.03paulhuynhplease let me know what i can do to fix it
15:35.14Winkieviperdude: the asterisk wiki or a channel specific one?
15:35.14nextimeit seem that ooh323c don't compile with latest * svn trunk
15:35.16nextimeanyone can confirm this issue?
15:35.24paulhuynhhttp://pastebin.ca/41951
15:35.29viperdudevoip-info.org/wiki
15:35.46Winkieah yes, i'm already checking out Originate on there :)
15:36.18viperdudeWinkie: the manager API allows you to do some nifty stuff
15:36.31Winkieviperdude: i just use it for a crappy call bot at the moment on our other * setup :)
15:36.40Winkie15:37.43 <@confbot> Conference meatwhore: Total connections 1
15:36.41NewSoleok anyone here good with SER and want a job... pvt me (auto timer)
15:36.47remissSIP/2.0 407 Proxy Authentication Required <-- umpf... what does this mean?
15:36.50Winkiewhat the hell is SER anyway?
15:36.54viperdudelol
15:37.04areskinestar, I am good but dont want a job
15:37.05paulhuynhSER is sip express router
15:37.05Ahrimanesremiss: it's rather self-explanatory?
15:37.06austinnichols101SER = sip express router
15:37.09Winkieah of course
15:37.12Ahrimanesareski: you suck ;)
15:37.16areskihahah
15:37.19areskisorry
15:37.29Ahrimanesareski: go do some php :)
15:37.34remissAhrimanes: yes.. i was just wondering what settings control proxy-authentication in sip.conf :-/
15:37.49areskiAhrimanes, oki.... go back to my work
15:38.01Ahrimanesareski: hehe.. anything big going on these days?
15:38.04*** join/#asterisk kippi (n=chris@untrust-gct.equinoxit.net)
15:38.05kippihi
15:38.26kippiwhat is the digium IP to analog box?
15:38.28areskiAhrimanes, lot of progress man :)
15:38.38klererHi, I’m having an experience with moving my iaxpeeers to mysql (Realtime) – the performance seems to go to hell and I get congestion.   Has anyone seen similar?
15:38.51areskiAhrimanes, I guess, next week will be start of a new age
15:39.04Ahrimanesareski: how? hehe
15:39.53areskiAhrimanes, I had speed-dial features yesterday :) funny stuff for the user
15:39.58paulhuynhdid anyone see my problem?
15:39.58*** part/#asterisk fjean (n=fjean@201.29.122.10)
15:40.06Ahrimanesareski: ah nice :)
15:41.06wasimkippi: digium make pci cards for analog FXO/FXS and digital e1 interfaces which when plugged into an * box work wonders for your phone bill
15:41.38*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
15:41.41NewSoleok anyone here good with SER and want a job... pvt me (auto timer)
15:41.53paulhuynhnewsole
15:41.54Winkieif you have an E1 interface and are worried about your phone bill there's a problem :)
15:41.59[TK]D-FenderNewSole : Turn that stupid timer off and put your request on the Wiki
15:42.00paulhuynhcontact this people
15:42.04paulhuynhmike@idv.net
15:42.11paulhuynhhe is expert on SER
15:42.17wasime1s generate big bills :(
15:42.26*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
15:42.40remissSIP/2.0 404 Not Found
15:42.47remissalright.. that one i don't understand..
15:43.00Winkieremiss: you dialling a client?
15:43.03paulhuynhso no taker on my problem
15:43.13remissWinkie: incoming call from my sip-provider
15:43.24Winkieremiss: sounds like you need a specific extension number?
15:43.34remissoh...
15:43.41Winkiesipgate requires it, last i used anyway
15:43.51remissand how do i set that?
15:43.52_Sam--anyone have any suggestions what to tweak for working on echo in meetme using digium card?
15:44.02_Sam--its all SIP clients, no PSTN
15:44.05*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de)
15:44.17remissi set context=my-number because of that, but that obviously isn't it then...
15:44.29Winkieremiss: no, you'll need an incoming context of some sort
15:44.41Winkiethen either a s exten, or an exten with the sip number your provider assigned to you
15:44.50Winkiei really can't help because i have barely a clue how to get sipgate working
15:45.34*** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net)
15:45.52viperdudeWinkie: if its sipgate in the UK THEY dont have a clue either, my account is forever timing out on registration
15:46.38*** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br)
15:46.41NewSoleok anyone here good with SER and want a job... pvt me (auto timer)
15:46.58Winkieviperdude: haha oh dear yes, they're hardly an 'enterprise' provider
15:47.23Winkiealthough i need to find a canadian company that can handle ~3000 calls a day to ~400 numbers
15:47.23viperdudeyeah won't even let you have two sip accounts on the same user account
15:47.28Winkieincoming, that is
15:48.25remissw00t :D
15:48.33remissput up an extension
15:48.40remisswith my number...
15:48.49Winkiethat working now?
15:48.53remissyeah
15:48.55Winkieexcellent
15:49.17remissbut why won't it work with just 'context=number' and 's' as extension?
15:50.16*** join/#asterisk Nix (n=Nix@81.214.255.57)
15:51.12Winkieremiss: i'll paraphrase into asterisk
15:51.16Winkiethey don't just dial SIP/remiss
15:51.21Winkiethey dial SIP/whatever@remiss
15:51.41NewSoleok anyone here good with SER and want a job... pvt me (auto timer)
15:52.00Winkieman that auto timer needs to be auto slower >:(
15:52.03remissWinkie: oh. ok
15:52.10austinnichols101somebody kick newsole
15:53.26paulhuynhanyone plase help me and take a look @ my sip debug?
15:53.34Abydos313he must not pay well since no one is biting on his job offer..heh
15:55.49*** join/#asterisk unixgeek (n=unixgeek@12.45.238.189)
15:56.22NewSoleacauly no one has msg me.... just need a ser server setup.... for 250$
15:56.30iDunnoa real unixgeek wouldn't have that nick.
15:56.30Winkieremiss: if that makes sense :)
15:56.39ManxPowerMaybe it's just that most of believe that SER is frequently the problem, not the answer.
15:56.44Winkiehe's obviously not a real unixgeek as he doesn't have rdns :)
15:56.53iDunnothat's also true ;)
15:56.57Winkiesays me
15:57.05Winkie15:58.18 [freenode] -!- Winkie [n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com]
15:57.07Winkielol
15:57.10remissWinkie: it does.. but i don't feel like it somehow :p
15:57.16iDunnohe's also using OS X, so, well ;)
15:57.23WinkieOSUX
15:57.33NewSoleManxPower... what we need is a proxy for faxes thus we are looking at ser
15:57.37NuggetOS X > *
15:57.53Nugget(* in the "all" sense, not "asterisk" sense.  I hate that ambiguity in here  :)
15:58.06remissi don't like os x anymore....
15:58.11iDunnoNugget: no, OS X has a broken BSD userspace - it is *not* greater than anything, ever. it's broken. inherently.
15:58.19Winkieapparantly you also don't understand the meaning of > and < :)
15:58.20Nuggetbroken how?
15:58.23ManxPowerI use "Asterisk" to mean "Asterisk", and * to mean wildcard
15:58.27WinkieiDunno: well it's better than windows, and os/2 :)
15:58.36Winkiei use * when i'm goddamn lazy
15:58.42iDunnoWinkie: oh, I don't know, OS/2 has it's good points ;)
15:58.48caio1982ManxPower: good one
15:59.20WinkieiDunno: it has a funky name
15:59.26Winkiei'm struggling to think of much else
16:01.13NuggetI loved OS/2.  I think it's a shame it didn't stay competetive.
16:01.26Winkieviperdude: you mentioned redirect, it's documentation is a little short, what the hell is 'extrachannel' all about?
16:01.28Abydos313still in service for some phones Nugget
16:01.35NuggetATMs too, I'm sure.
16:01.37ManxPowerhttp://store.voxilla.ca/product.php?productid=16159&cat=0&page=1  I didn't know Polycom Soundpoint supported iLBC.
16:01.39Abydos313yeah
16:01.44Winkieyep, ATMs historically run on os/2, but have started being windows
16:01.48Winkiewhich is fun when they bluescreen
16:02.00Abydos313os/2 is rock solid
16:02.20Winkieyeah but it's easy to be rock solid when it has no functionality :)
16:02.29Abydos313true
16:02.33austinnichols101did anyone happen to see the Httpanties on thinkgeek.  What a riot!  http://www.thinkgeek.com/tshirts/ladies/6792/
16:02.37iDunnowe were using OS/2 v1 (which was still MS) back in the good ol' days, it was remote booting win 3.1 workstations!
16:02.50Abydos313really? wow
16:03.19iDunno:)
16:03.21*** join/#asterisk lorinc (n=ang@caracas-2964.adsl.interware.hu)
16:05.47*** join/#asterisk tomtom_ (n=tom@83.217.70.163)
16:05.51tomtom_joe
16:05.58*** join/#asterisk salviadud (n=ralfalfa@201.137.161.198)
16:06.12salviadudgood mornin'
16:06.13WinkieiDunno: ouch, luckily i'm not that old
16:06.26iDunnoWinkie: dude - I'm only 24 ;)
16:06.32tronixheh I remember booting Warp (OS/2 3.0) on a roomie's 486/33 w/4MB... it swapped like hell, was unusable, and had 5 min long boots
16:06.36iDunnothis was while I was still at school *grin* :)
16:06.50tronixWarp on the other roomie's 486dx2/66 w/8 MB..booted in 40sec and was usable.
16:07.06WinkieiDunno: i'm 21, and the nearest i've dealt with os/2 is some legacy stuff at my old place
16:07.08tronixpretty nice OS, protected memory and stuff (back when Windows didn't have it, pre-NT)
16:07.20tronixtoo bad it didn't take off.
16:07.23Winkiea nice text novell console too
16:07.35wunderkinpaulhuynh, the call is being sent to the s exten, use that
16:08.10tronixWinkie: heh... I very vaguely remember Windows 1.0... text mode. 2.0 wasn't much better. 3.0 was a big day, nice GUI and stuff.
16:08.22Winkietronix: i found some windows 1 beta disks at my uncle's farm once
16:08.26iDunnoNovell has some "interesting" ideas, they were way ahead of their time - unfortunately it sucked management wise ;)
16:08.26Winkienow that was an interesting experience
16:08.57iDunnoand now everyone's going on about how great Active Directory is... ffs, Novell had that *years* ago ;)
16:09.04tronixindeed.
16:09.14Winkieactive directory more like ldap am i rite?
16:09.45iDunnoyeah - it is infact ldap...
16:09.53*** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com)
16:09.54iDunnobut with a slightly broken implementation.
16:10.00Winkieindeed
16:10.12Winkieldap is deprecated anyway :)
16:10.17*** join/#asterisk enemy^x (n=eqwrweqr@morpheus.dataguard.no)
16:10.39iDunnoerm - it's not ;)
16:10.40eKo1quick ?: as I understand it, every call requires two channels but when I enter 'show channels', I see an odd number of channels. Why is this?
16:10.52iDunno(we're using LDAP at work - and network booting debian machines *grin*)
16:11.01WinkieeKo1: it depends, not every call requires 2 channels
16:11.02iDunnoNIS should be deprecated, though.
16:11.48viperdudeeKo1: every BRIDGED call requires 2 channels
16:11.48WinkieiDunno: PXE?
16:11.48iCEBrkrwunderkin: I still have OS/2 Warp.  It kicked ass. :)
16:11.48iDunnoWinkie: yup :)
16:11.49eKo1viperdude: I see...
16:11.49iDunnoWinkie: and root on NFS
16:11.49eKo1hmm...all these calls should be bridged
16:11.57*** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
16:12.48iCEBrkrI've been wanting to build an OS/2 machine again.  Make it mirror all the functionality of my linux box
16:14.10wunderkini wish that os/2 succeeded, need another option to windows (for lusers)
16:14.18iCEBrkrwunderkin: Yea, me too.
16:14.22hensemaiCEBrkr: and your other hobbys include swimming with sharks and eating nails?
16:14.28iCEBrkrhensema: hehe
16:14.43*** join/#asterisk T-Squared (n=T-Square@hidden.serreyn.com)
16:14.51*** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net)
16:14.54iCEBrkrNot sure if it still exists, but Apache ran on OS/2
16:15.00enemy^xTON: National Number (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '8300'
16:15.07enemy^xwhy is the 0 stripped? the number was 08300
16:15.13enemy^xis it within zapata.conf?
16:15.20WinkieiDunno: we use PXE w/etherboot to bootstrap the boxes, and home will be on NFS sooner or later
16:15.30*** join/#asterisk mjmac (n=mjmac@pdpc/supporter/active/mjmac)
16:15.31Winkieenemy^x: is this incoming?
16:15.44WinkieiDunno: you have any experience with setting up HA NFS?
16:15.52enemy^xwinkie: outgoing.
16:16.07Winkieenemy^x: it's possible you need to pause, or so i've heard
16:16.15*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
16:16.42T-Squaredanyone have any experience with AMP?  when doing outgoing ZAP channels, I am not seeing rollover to the other ZAP trunks for outgoing calls, the 2nd person just gets a busy signal
16:16.52iCEBrkr~amp
16:16.53jbotit has been said that amp is NOT supported here! people using it should join #amportal
16:16.58Winkiebah beat me to it
16:17.01iCEBrkr:)
16:17.03enemy^xWinkie:nationalprefix=0 does this have something to do with it?
16:17.11Winkiesomeone wanna tell me what AMP actually is?
16:17.18Winkieenemy^x: I couldn't say i'm afraid
16:17.29iDunnoWinkie: nah :)
16:17.30wunderkinenemy^x, probably
16:17.32iCEBrkrWinkie: It's some 'control panel' for Asterisk.
16:17.37Winkieah right
16:17.38T-SquaredI would but it says that the channel does not exist (so not like I didn't try)
16:17.40austinnichols101winkie: brower-based admin for astrisk.
16:17.44Winkiepff control panels are for people who can't use a CLI
16:17.47iCEBrkrWinkie: I attempted to install it once, but it was bloated and required all this lame dependencies
16:18.03WinkieiDunno: i need to set up ideally some sort of striped and mirrored NFS service
16:18.18WinkieIE we have 2x200 gig in each server, and i want say 500 gig total space available :)
16:18.23*** part/#asterisk T-Squared (n=T-Square@hidden.serreyn.com)
16:18.32Winkiei'm thinking raid 0 each server and then use drbd to duplicate it
16:18.39Winkiebut as for the network side, god knows
16:18.48iCEBrkrWinkie: It's just that the learning curve for Asterisk is 3-fold.  The person needs to understand Linux, Asterisk itself and basic telephoney stuff.
16:19.02*** join/#asterisk coppice (n=chatzill@90.201.17.210.dyn.pacific.net.hk)
16:19.06enemy^xit was nationalprefix
16:19.12austinnichols101iCEBrkr: well said
16:19.36*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com)
16:19.55WinkieiCEBrkr: that's true enough but i've started out from virtually nothing into setting up an E1 connected asterisk that handles 8000 calls a day, although i haven't got it working yet :)
16:19.56iCEBrkr...on top of learning all that people just want it to WORK, cuz it's 1) cool, 2) can save them money.
16:20.00Winkiealthough i know linux :)
16:20.22Winkiewell if they want it to just work they should have bought something embedded ;)
16:20.23iCEBrkrIt's just too much stuff to learn at once.. It's all interconnected in some way or another
16:20.37redder86I've got a channel coming in to Asterisk via PSTN (TDM400) and being delivered over IAX2 to the peer.  If I make a recording using the Monitor() app in Asterisk, and if I also make a recording of the same call on the IAX2 peer, and if I then compare the two recordings ... I see that they are identical in timings and frequencies, but the amplitude of the waves (the volume) is noticeably reduced on the IAX2 peer-side.  As I've authored the IAX2 pee
16:20.41*** join/#asterisk Drew__ (n=foo@zux221-186-224.adsl.green.ch)
16:22.20Winkieredder86: wrapped i'm afraid
16:22.24ManxPowerredder86, Maybe Monitor on the system with the Zap interface gets the audio before the gains are applied.
16:22.27*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
16:22.46redder86Winkie: wrapped?
16:22.56redder86MAnxPower: which gains affect IAX2 ?
16:23.01Winkieredder86:                      amplitude of the waves (the volume) is noticeably reduced on the IAX2 peer-side.  As I've
16:23.01Winkie<PROTECTED>
16:23.01Winkie16:22.11 -!- Drew__ [n=foo@zux221-186-224.adsl.green.ch] has joined #asterisk
16:23.05Winkiewoah that's some bad pasting
16:23.09Winkiei apologise
16:23.12iCEBrkrSure sure.
16:23.27redder86As I've authored the IAX2 peer (iaxmodem) I know that the peer is not doing anything with incoming audio to trigger this.  So, could anyone explain what could be causing the volume difference between each end of the IAX2 channel?
16:23.59*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
16:24.02salviadudsquirrels man...
16:24.15salviadudi mean, i dunno
16:24.15Winkieredder86: ManxPower's idea is interesting, but it's rxgain in zapata.conf i assume he's referring to
16:24.21iCEBrkrBADGER BADGER BADGER BADGER BADGER BADGER BADGER
16:24.21Winkieor zaptel.conf, i dunno :(
16:24.40ManxPowerredder86, rdgain/txgain is only for Zap
16:24.57redder86Winkie, ManxPower: in any case, gains in Zap are at 0
16:25.44Winkieredder86: it's an interesting problem, the answer would be that i don't know. How significant is this difference?
16:25.55redder86I'd say that the volume is reduced at about 75%.  I'm getting reports of this, and as it doesn't happen to all iaxmodem users I'd guess that it's something wrong with these fellows' Asterisk configs.
16:26.54redder86It's a significant-enough decrease in volume that spandsp in iaxmodem won't even detect the audio as fax audio.
16:26.59*** join/#asterisk stoffell (n=stoffell@d51A4D720.access.telenet.be)
16:27.29Winkieuh, chan_zap does that
16:27.31Winkieiirc
16:27.37Winkieperhaps iaxmodem has something i don't know about
16:28.30redder86iaxmodem just takes the audio samples that it receives on iax2 and passes them to spandsp.  The recording is made by just saving it to disk between those two steps.
16:29.14remissuh.. i don't get it
16:29.33Winkiei'm confused as to how it could happen, have you set up a test peer and tried locally?
16:29.52remissFailed to authenticate on INVITE to '"Unknown" <sip:Unknown@10.45.0.3>;tag=as7ecbf807' <-- alright.. i don't want it to do that?
16:30.01redder86I've no problem with faxing and such with iaxmodem locally, so no I can't reproduce the problem here.
16:30.06remissdoh.. i got it.. never mind..
16:30.07*** join/#asterisk RoyK (n=roy@ti211310a080-4532.bb.online.no)
16:30.17Winkieredder86: do you have a copy of their asterisk configs?
16:30.31redder86they've sent me some of them
16:30.37redder86which ones would be suspect?
16:31.00Winkieredder86: well iax.conf and extensions.conf i'd guess, but i haven't a CLUE
16:31.02Winkieto be honest
16:31.34*** join/#asterisk RV-Dioxide (i=appleboy@ip68-231-211-153.oc.oc.cox.net)
16:31.49remissalright.. i don't get it..
16:33.06redder86okay, thanks anyway guys
16:33.08*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
16:37.00*** join/#asterisk SibRwork (i=hidden-u@fw-int.transbeam.com)
16:37.20SibRworkcalling from sip exten to sip exten, but not getting any audio to pass through...what could be the issue
16:37.23SibRwork?
16:38.45ManxPowerSibRphrek, sounds like a calssic NAT and/or firewall problem
16:40.46*** join/#asterisk brettnem (n=brettnem@nemeroff.com)
16:41.05SibRworkManxPower: my sip.conf has nat=yes
16:41.14SibRworkfor the internal extentions
16:41.23WinkieSibRwork: is there NAT between them?
16:41.40SibRworkyeah
16:41.51SibRworkwe're all on the inside of a network
16:41.58SibRwork192.168.*.*
16:42.05Winkieand this is internal to internal?
16:42.18SibRworkint extension to int extension yes
16:42.21ManxPowerSibRphrek, NAT is MUCH more complicated than nat=yes if you have Asterisk behind NAT and the SIP devices are not behind NAT or behind a different NAT
16:42.36WinkieSibRwork: why would you have NAT on an internal 192.168.0.0/24?
16:42.55SibRworkasterisk is external - in a colocation, the sip phones are internal
16:43.50Winkieah
16:43.58Winkieset up a VPN?
16:44.13SibRworkno
16:44.16SibRworkcan't do that right now
16:44.35Winkiewell given multiple internal phones it's just silly not to
16:45.31SibRworki'm just trying to run a test
16:45.35SibRworkb/c we are hooking up a new provider
16:45.41SibRworkand we are rushing to get something working
16:45.47SibRworkeven tho we have other systems that work
16:45.49SibRworklong story
16:46.16Winkieah, well there's probably a guide to NAT and STUN or whatever you need to use on voip-info.org
16:47.00*** join/#asterisk kpettit (n=keith@69.15.174.114)
16:49.01ManxPowerSibRphrek, if asterisk is on a public IP the SIP devices are behind NAT, it should work if you have nat=yes, but do NOT have localnet= or externip= in sip.conf and make sure there is no firewall active
16:49.06*** join/#asterisk w32 (n=123@adsl-70-224-74-227.dsl.sbndin.ameritech.net)
16:49.14*** join/#asterisk ms345 (n=mike_sim@64.74.198.10)
16:49.16oogledownload MP3, destroy the music industry.  download Asterisk, destroy AT&T
16:49.17ManxPoweralso make sure you DO NOT HAVE allow=all  use disallow=all and then allow=thesinglecodecyouwant
16:49.36ManxPoweroogle, your ToDo list?
16:49.42oogleyes
16:50.02*** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it)
16:50.07ManxPowerMy ToDo list just has "Become Emperor of the World"
16:50.07WinkieI have no ill feeling towards AT&T
16:50.18Winkiebut if you want to destroy Your Communications, go right ahead
16:50.27Winkieour account manager has been ignorng my texts and my workmate's emails for 3 days
16:50.29ManxPowerAT&T has been pretty good to my clients
16:50.30WinkieSERVICE!
16:50.37ms345can someone suggest a good yet not too pricey VoIP phone for a receptionist?
16:50.49Winkiegrandstream budgettone?
16:50.54ManxPowerms345, depends on what you need.
16:50.59Winkiename your features
16:51.12ManxPowerThe BudgeTone doesn't even support callerid name
16:51.32ms345lots of buttons and blinky lights - seriously - probably doesn't matter
16:51.45ManxPowerms345, There are no cheap and good phones.
16:51.46Winkiegxp2000 has more blinky lights
16:51.47Mimmus5 minutes to setup an Asterisk trunk between two sites for which company pays many many money every year... not bad... I'm pretty satisfied
16:51.48ms345they only have CIDNUM now
16:51.51WinkieManxPower: that's entirely not true
16:51.52ManxPowerTry Polycom or Cisco
16:51.59Winkiethe gxp2000 is the cheapest of our review set and it's the second best
16:52.10ManxPowerms345, no, the BT101 does not even have a display that is able to support letters
16:52.12Winkiethe sipura one is midrange and still the best :)
16:52.12ms345cool - I'll check the gxp2000  out
16:52.27oogledoes anyone know a good low-cpu-cost text to speech generator that i can stream in to asterisk besides festival?
16:52.45MimmusWinkie: is sipura now Linksys or Cisco?
16:52.47brettnemdo you guys find that a lot of providers don't give you CNAM?
16:52.49*** join/#asterisk sergeus (n=s@195.112.98.13)
16:52.49ms345any polycom or cisco model numbers to look at?
16:52.54ManxPowerWinkie, For 3 years GS has been releasing somewhat crappy hardware with some of the most bug ridden firmware in the industry.  Sorry, but I can't trust them.
16:53.03ooglei can always write the integration myself
16:53.08WinkieMimmus: i think they're linksys' and cisco own them, or something
16:53.09oogleas long as it has an api
16:53.14WinkieManxPower: no that's aastra :)
16:53.23Winkiebut yes i will agree there are issues
16:53.25IronHelixGS releases products then 1-3 years later the firmware comes to usable state :)
16:53.25brettnemms345 have you found that many providers don't provider caller id name?
16:53.30Winkieheadphone port speaker disconnection / firmware issues etc
16:53.41ManxPowerWinkie, AAstra has not even had a SIP phone for a year
16:53.50IronHelix...?
16:53.54WinkieManxPower: i have the 480i and the other one and they're appauling
16:54.06Winkieextremely expensive, don't work at all in the case of the 480i and the build quality is terrible
16:54.11IronHelix480i, 480i-ct, 9133i, 9122i...
16:54.15Winkiethe volume control on the 9133i is SO bad
16:54.16ms345no - I get CIDNAME from BellSouth and ITC, my two PRI providers
16:54.21ManxPowerbrettnem, the only providers that I know of that don't do CIDNAME are VoIP ones and only a moron would use a VoIP ITSP for production use.
16:54.23Winkieit's just a bit of plastic
16:54.38Mimmusmy boss says that 100Euro/phone limit is mandatory
16:54.43ManxPowerWinkie, Oh, there are many horrble IP phones out there, but most of them are fairly new.
16:54.45brettnemManxPower: interesting.. ;)
16:54.46Winkieright i'm off to get burger king i think
16:55.03WinkieMimmus: you won't get much for that
16:55.21brettnemManxPower: I'm trying to figure out how to sell a CNAM dipping service.. I have the program together... don't know if there is interest
16:55.22ManxPower100 Euro is, what, $140
16:55.33Winkie16:57.16 <@Shaniqua> 100 eur = 119.04 usd
16:55.34ManxPowerbrettnem, there are a couple of them out there.
16:55.51ManxPowerWow!  The exchange rate is getting slightly better.
16:56.03Winkie16:57.45 <@Shaniqua> 100 usd = 56.75 gbp
16:56.08Winkienot by much
16:56.08Mimmusehm... saying all truth: limit is lower...
16:56.09IronHelixbrett- i think theres interest, if the price is right
16:56.31WinkieMimmus: nobody produces a phone at that price i'm afraid
16:56.35Winkieso unless you want to use software
16:56.41MimmusI'm forced to buy GrandStream or 'made in china/taiwan'
16:56.46Winkiewell if they do produce a phone at that price i haven't seen it and it probably sucks
16:56.59IronHelix120 would buy you a linksys i think, maybe not one of the better ones tho
16:57.10Winkieperhaps the old 841?
16:57.27MimmusWinkie: GS 101: 65E, GS GXP-2000: 90E
16:57.32IronHelixyeah you're pretty much stuck to a sipura 841 or a grandstream 2000
16:57.38WinkieMimmus: that's pretty cheap
16:57.55Winkiecosts me 109 eur for a gxp2000 :(
16:57.56Winkiebastards
16:57.58Winkieanyway i'm going!
16:57.59Winkiebye
16:58.03IronHelixbefore you go
16:58.06MimmusI'm pushing GXP2000. Can I afford the risk to buy 50 of these or not?
16:58.08IronHelixDONT BUY A GS 101
16:58.12IronHelixbuy one
16:58.15IronHelixsee if you like it
16:58.17IronHelixif you do buy 50
16:58.23Qwell49
16:58.36WinkieMimmus: if you plan to use headsets, make sure you get the latest revision and check it out
16:58.43Winkiebecause there is a problem with the headset port on ze back
16:58.48Winkieanyway i really am gone now
16:59.24MimmusIronHelix: ok, thanks! I buyed a GS101 but there are problems, especially because there is not an alphanumeric display and microphone is low
16:59.32MimmusWinkie: bye, thanks
16:59.42IronHelixyeah the GS phones look and feel like toys
16:59.47IronHelixthus the nickname 'barbietone'
17:00.18_Sam--Mimmus :  there are only a handful of reasonably happy GXP2000 owners
17:00.19MimmusIronHelix: tried also an ATCOM AT320 (PA1688S chip), not so bad but some problems too
17:00.23_Sam--i happen to be one of them
17:00.32IronHelixme too
17:00.33_Sam--but many dissatisfied customers as well.
17:00.36IronHelixthe gxp is a good phone
17:00.38IronHelixjust not the 100
17:00.50zoahey ho sam
17:00.58zoawe have a lot of happy gxp users
17:01.01_Sam--hey there joach
17:01.15zoasold quite a big amount of those already
17:01.17_Sam--most people here do nothing but talk down the gxp2000
17:01.20zoahey ho!
17:01.28zoaive seen some issues but very few
17:01.32_Sam--it has a lot of probelms actually.
17:01.40_Sam--i wouldnt put 50 in one place, not with qualify = yes anyway :)
17:01.45IronHelixhey sam/zoa, have you found the gxp has a high failure rate (esp the handset)?  also is GS support responsive with replacement parts?
17:01.49MimmusI'm not looking for a lot of features but decent sound quality, alphanumeric LCD, MWI, call-transfer and a few others...
17:02.09zoaIronHelix: we rma'd some
17:02.09zoacompared to snom a lot
17:02.13zoacompared to the bt101 very few
17:02.17_Sam--i have 40 gxp2000s in different places
17:02.21_Sam--i have had none fail yet
17:02.27_Sam--but they are all bug filled
17:02.35zoatrue
17:02.35_Sam--if i turn on BLF, * segfaults
17:02.40_Sam--if i turn on qualify = yes, moh breaks
17:02.41zoadepends what you want to use it for, yes
17:02.41salviadudi need some help...
17:02.43salviadudhttp://pastebin.ca/41959
17:02.58salviadudi am not sure what to add to my extensions.conf file
17:03.00Mimmuswhy not "qualify = yes"?
17:03.02_Sam--they are fine if you want to check for new firmware every day
17:03.07zoa:)
17:03.20IronHelixsalviadud- iax sends context as well as exten
17:03.21_Sam--read the GXP2000 wiki page
17:03.38_Sam--http://www.voip-info.org/wiki/view/GXP-2000
17:03.40*** join/#asterisk SplasPood (n=jwb@206.252.198.100)
17:04.01_Sam--i think your customers/users/whoever could be happy with 50 of them...but its only a 60/40 chance.
17:04.03salviadudso, i should create a context called {fwdnumber}-fwd-incoming???
17:04.06_Sam--just ask justinu
17:04.08IronHelixits looking for the FWD number in [fwd-incoming]
17:04.18IronHelixno make [fwd-incoming] and put an extension s in that
17:04.38_Sam--i used to be the biggest gxp supporter here...used to be that is.
17:04.39salviadudthanx man! :D
17:04.51_Sam--i like them...but i am seeing why some people wouldnt
17:04.55Mimmus_Sam--: I read. Sic.
17:06.03_Sam--Mimmus:  what type of environment is the installation?  like, a call center, a regular office, what?
17:06.13*** join/#asterisk starwarez (n=starware@201.135.68.20)
17:06.22starwarezhi all
17:06.58zoai would recommend snom or so for a customer
17:07.11Mimmuspeople think that Asterisk is a cheap change... then you suggesto 250$ phones... and my boss dies!
17:07.30zoagive him a channel bank
17:07.30Mimmus_Sam--: an office with 60 people
17:07.32wasimMimmus: buy $50 pa168 phones
17:07.52_Sam--channel banks cost as much per port as a gxp
17:07.53wasimMimmus: or buy $85 4 port fxs ata
17:07.54_Sam--i would rather have a gxp
17:07.59austinnichols101it's cheap if you compare it to a Lucent G3
17:08.09Mimmuswasim: I have three AT320 but I'm currently unable to find them on the italian market
17:08.26_Sam--you are talking 60 PHONES>..your boss if he can afford 60 desks, should be able to afford a phone system.
17:08.27wasimMimmus: atcom ship direct as well
17:08.33_Sam--i mean, its not like you are a small shop
17:08.43_Sam--spend now, or you will end up just spending more later when you're unhappy
17:09.11_Sam--<personal opinion of course>
17:09.15Mimmuswasim: I don't like ATAs: I have no CallerID display on commonly used analog phone sin Italy
17:09.33wasimMimmus: you should, we get clid just fine on cheap $5 phones
17:09.36*** join/#asterisk paulhuynh (n=paul@c-68-37-18-82.hsd1.de.comcast.net)
17:10.03Mimmus_Sam--: I'm trying to explain that currently maintenance of our legacy PBXs is 25,000Euro/year...
17:10.06paulhuynhi'm using asterisk w/ pap2 what codec can i use that used least amount of bandwidth
17:10.18paulhuynhand that i don't have to pay the licensed from digium
17:10.23wasim60*5=300 + 60/4*85 + a small * box
17:10.36Mimmuswasim: clid in Italy is a luxury!
17:10.40wasimthats not much, enough for a cheap ass boss
17:13.41_Sam--Mimmus:  maybe not every desktop needs the same phone
17:13.58*** join/#asterisk gkchicago (n=gkchicag@66.9.120.18)
17:14.50Mimmus_Sam--: this is true even if thius could be mor emanageable
17:15.05gkchicagoDoes anyone know why `exten => s,202,MixMonitor(/tmp/recording-%d.wav)` isn't putting an incrementing number in the filename?
17:15.16Mimmus_Sam--: provisioning, configuring, training...
17:15.16*** join/#asterisk t0ke (n=kaka@194.Red-81-36-121.dynamicIP.rima-tde.net)
17:15.35paulhuynhany idea on the codec setting
17:15.40paulhuynhi'm using asterisk w/ pap2 what codec can i use that used least amount of bandwidth
17:15.43paulhuynhand that i don't have to pay the licensed from digium
17:15.49Mimmusbut too risky... better differentiating...
17:16.20MimmusI hope to buy some GXP-2000s and some AT320
17:17.06*** join/#asterisk RoyK (n=roy@ti211310a080-4532.bb.online.no)
17:18.46*** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
17:18.52Mimmusbye and thanks, I will break you in the future again...
17:19.07Winkiefor applications where you could be sending over a thousand manager commands to an Asterisk server from different client machines daily you will almost definitely have at least one crash/deadlock happen per day.
17:19.07websaeI keep getting dropped calls, how can one debug this?
17:19.10Winkieis this still true?
17:19.27websaei would greatly appreciate if anyone knows how to debug dropped calls..
17:19.35Winkiewebsae: sip?
17:19.40websaeyes
17:19.48Winkiesip debug? :)
17:20.05websaeis there a way to throw sip debug in a file?
17:20.18websaei guess that's what i am trying to figure out
17:21.24Winkienot that i'm aware of, possibly through the manager API but don't quote me on that
17:21.32NewSoleanyone good with ser
17:21.46WinkieNewSole: you've been asking all day, no, and stop it
17:22.08websaeso sib debug...no way to get that to a file?
17:25.27websaehi
17:26.16*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
17:27.53DaPrivateertrying to compile on a new box and getting undefined reference to `__h_error' any ideas?
17:28.00DaPrivateer(freebsd 5.4-stable)
17:29.07wasimwww.gentoo.org
17:29.14*** join/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0)
17:29.16salviadudguys, i got FWD working
17:29.17salviadudbut
17:29.26salviadudi can't dial toll free numbers yet...
17:29.40salviadudsomething to do with the dialplan?
17:29.49mikefoois it legal to originate a call from a terminating voip provider and have a outbound display number of say, a home phone, any not any did the voip provider assigned.
17:30.28Winkiemikefoo: it depends on your country, i'd be surprised if your voip provider's telecomms provider allows that
17:30.51mikefooAhh they might not allow a custom outbound number to go through?
17:30.59mikefooim in the US btw
17:31.33Winkiemikefoo: they almost 100% won't
17:31.39Winkieand i don't know US laws i'm afraid
17:32.09*** join/#asterisk Flauto (n=zhao@71.194.194.48)
17:32.23Flautohi all
17:33.44Flautoit is quiet here
17:33.45*** join/#asterisk Assid (n=assid@203.115.64.11)
17:33.46Assidheya
17:33.57Flautohi assid
17:34.08salviadudi need some help
17:34.09salviadudhttp://pastebin.ca/41963
17:34.17Assidumm.. can anyone suggest a provider which lets you port a number in for DID.. and lets add more incoming channels
17:34.22Assidvoicepulse has a hard limit of 4
17:34.23salviadudyou guys have fwd with toll free numbers enabled?
17:34.27PupenoLwhat character set/encoding does the manager of asterisk use/receive ?
17:34.55*** join/#asterisk _dusty (n=Dusty@64.89.118.139)
17:34.57Flautoassid, look into broadvoice
17:34.59gkchicagoIn the record app you can put "%d" in the filename to add an incrementing number.. is there a similar function for the MixMonitor app?
17:35.05AssidFlauto: heard bad reviews of BV
17:35.09Flautoas i understand, incoming is not limited
17:35.23Flautoi have used it for more than a half year now
17:35.32Flautoi had problems last year in may and june
17:35.36Flautoother than that
17:35.41Flautoit has been okay
17:36.05Flautoonce in a while, one of the 5 or 6 proxies would screw up for a little bit
17:36.08Flautothat is about it
17:36.17*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
17:36.19Assidwhat about quality
17:36.40Assidvoicepulse used to give good quality
17:37.09mikefooAssid: how many concurrent calls can you make with voicepulse?
17:37.19Assidnot make.. incoming
17:37.24Assidincomng has a 4 channel limit
17:37.31mikefoooutbound?
17:37.35mikefoosdame?
17:37.37mikefoosame*
17:37.39Assidnah
17:37.41Assidpay per use
17:37.41Assidso..
17:37.56mikefoounlimited..?
17:37.56mikefooin theory.
17:38.07Assidyeah
17:38.09Flautothe call quality is not bad depending on where you are calling
17:38.10Flautohehe
17:38.11Assidso long as you have the bandwith
17:38.24AssidFlauto: shouldnt be that big of a difference
17:38.27freatam I doing the right thing, asking users to please not use those swivel jack things (that go inline on handsets to keep it from tangling up)?
17:38.36Flautothe funny thing is
17:38.38Flautoi call china a lot
17:38.44mikefooAssid: yah I will be on a 10mbit connection, need a decent provider to terminate for me.
17:38.47Flautomost of the major cities are good
17:38.55Winkiefreat: what's the problem with them?
17:39.03Flautobut some smaller ones would come cross to have problems sometimes
17:39.09Assidoutbound ive seen pretty decent feedback for VP and teliax
17:39.22Flautoi would think that they land in the major cities only
17:39.26remissFailed to authenticate on INVITE to '"Unknown" <sip:Unknown@84.48.68.129>;tag=as274d9983' <-- on outbound calls i get this.. what do i need to do to change "Unknown"?
17:39.31Assidvoipjet is in there.. but THESE guys apparetly prefer teliax over voipjet
17:39.56Assidremiss: change the username and caller id/ name registered on the sip device
17:40.14mikefooAssid: how about on outbound caller id display, can you cutomize?
17:40.15freatWinkie: seems like they are causing problems with their calls. Report is that their speakerphone works great (Polycom phones) but that their handsets are not as good. The people with the complaints all bought some cheap kind of swivel thingie for their phone cords.
17:40.25remissAssid: no way to set it in asterisk?
17:40.33Assidmikefoo: yes.. all of them let you do so
17:41.07mikefooAssid: yah there is a way just making sure they allow it.
17:41.15Winkiefreat: ah, well if they're cheap they'll probably use a copper strip or something similar to provide the connection, and i imagine it oxidises easily
17:41.24mikefooAssid: is there any legal issues I should worry about with that.
17:41.33freatWinkie: are there good ones out there?
17:41.44Assidif im not mistaken yuo are SUPPOSED to provide your caller id
17:41.47Assidhowever..
17:41.48freatWinkie: I would like to get one and test it myself
17:41.51Winkiefreat: i wouldn't know, all i can say is my mum's had one on her phone for the last 4 years and it sounded pretty good :)
17:41.58freatahh ok
17:42.02Assidif they realy wanna.. they can always trace the call back to you by seeing the company which is terminating
17:42.08Assidand reading the call records off that
17:42.34Assidfreat: polycoms shouldnt be giving oyou those problems
17:42.37Assidsipura.. yes
17:42.48mikefooAssid: yah but I will be the outsource for appoints of different companies, can I legally display the companies I dial out for, their caller id?
17:42.51Assidbut then sipura you just upgrade the firmware
17:42.57freatAssid: yeah exactly. Polycom quality is awesome
17:43.07freatAssid: got one right here on my desk (IP500)
17:43.13*** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net)
17:43.14Assidi really wish i can get a poly phone
17:43.15SpaceBasshey folks
17:43.27Assidthe guy was supposed to send me one.. but unfortunately couldnt make it
17:43.28Assidso..
17:43.33Assidim stuck with soft phone
17:43.47Assidthey are using poly 501
17:44.12Assidi was supposed to get a phone as a thank you for setting up voip for him in a real short amt of time
17:44.23Assidnot to mention.. i asked him for it
17:44.24Assidhehe
17:44.43freatAssid: keep harassing them then, if that was part of the deal.
17:44.46*** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com)
17:44.54kpettitAny ways to handle echo on IAXY's devices?  I have a confrence bridge with 6 or so Digium IAXY's connected to it
17:44.55Assidwasnt "REALLY" part of the deal
17:44.57websaeanyone know a good way to record sip debug to afile?
17:45.02Assidhe said yeah cool.. im sending it to ya
17:45.10Assidi dont REALLY REALLY need it
17:45.16kpettitafter some time I get horrid echo feedback.  And if they all hang up and call back in the previous echo is still there
17:45.17Assidbut yeah.. would be nice
17:45.18stoffellwebsae, yes, putty to the box, and let your putty capture the session.. then you have a file
17:45.24Assidsoftphones arent the best thing to play with
17:45.34kpettitIf I yell "Hello" it will start to echo, and everybody hangs up, I call back in and it's still echo'ing
17:45.36AssidWinkie: you ever used any inbound provider
17:45.50WinkieAssid: sipgate
17:45.59Assidyou in UK ?
17:46.05Winkieaye
17:46.09Flautoassid, the good think about bv is that you get a flat monthly fee
17:46.09SpaceBassdoes digium still sell a student/hobbiest (IE Low Cost!) version of their fxo/fxs board?
17:46.09Assidthought so
17:46.10Assidhehe
17:46.15SpaceBassmy clone x100p sucks a-hole
17:46.27AssidFlauto: i do need quality.. reviews of BV isnt that great
17:46.51AssidSpaceBass: how do you find it
17:46.57kpettitSpaceBass, clone x100p ??
17:47.02WinkieAssid: why what's up?
17:47.23Assidwanna give these folks a good provider
17:47.32Assidthey willl need around 6-8 channels.. and scalable
17:47.41SpaceBassso-so...used it for incoming calls mostly for a while but not using it for all LD calls for work and home... quality is better than my analog lines by far, but I do get some "break-up" as the call progresses into more than 1 hour 30 mins
17:47.51Winkieah, not got any experience of an incoming provider, really i need a top end one in canada at some point
17:47.52Assidvoicepulse isnt offering anything on that.. and well.. frankly i never tried any other provider except nufone for incoming
17:47.56SpaceBassAnd I put in a request to port my number to BV 2 months ago and they still havent done it
17:48.01SpaceBasskpettit,  yet...x100p clone
17:48.10kpettithttp://www.x100p.com/products_2.htm  like this device?
17:48.18kpettitor there PCI card/
17:48.34remiss:(
17:48.37kpettitI just bought two of there IAX FX boxes this morning.  But I hate using PayPal to buy crap
17:48.43kpettitwhere did you get yours?
17:48.55SpaceBasskpettit,  no...like this: http://cgi.ebay.com/X100P-Compatible-Card-Asterisk_W0QQitemZ5866415725QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
17:48.58remisswhy do i keep getting proxy authentication required on outbound calls?
17:49.32*** join/#asterisk chops (n=moise@207-172-221-143.c3-0.smr-ubr2.sbo-smr.ma.cable.rcn.com)
17:49.35Winkiebecause you haven't authenticated with your proxy?
17:50.10kpettitSpaceBass, ahh, I hven't used those yet.  I was excited about there FX one so I can find a replacement for the Digium IAXY's.  The really bug me
17:50.31remissWinkie: probably :-/
17:50.37SpaceBassis the IAXy done?
17:50.39Winkieremiss: you got a register statement in sip.conf?
17:50.56*** join/#asterisk juice (n=juice@mo-71-0-60-40.dyn.sprint-hsd.net)
17:51.03SpaceBasskpettit,  unless your stuck on IAX, the linksys PAP2 works well
17:51.12Winkieor the appropriate entries in the outbound definition?
17:51.30*** join/#asterisk sdali (n=FrankM@68-189-43-111.dhcp.rdng.ca.charter.com)
17:51.42SpaceBasskpettit,  that x100p clone worked GREAT in my PII 350mhz box...dropped them in my 1.7ghz box with 512mb ram and I get BAD echo
17:51.46remissWinkie: yes.. and it's registered.. incoming works ok too
17:51.53kpettitwe have alot of firewall issues with SIP, IAX seems to do the trick for the most part, but the boxes are unreliable
17:52.00remissWinkie: the outbound definition?
17:52.04kpettitknow of any clones for the FX box?
17:52.07wasimso, whats the longest iax2 call ever?
17:52.13Flautospacebass, dont' you have echo problem with x100p
17:52.19Qwell[]clones of a cheap device?
17:52.21Winkieremiss: pastebin your sip config?
17:52.30remissk
17:52.32Winkiewasim: i dunno if there's a world championship but i've left one on for a couple hours before 8)
17:52.34SpaceBassFlauto,  yeah...thats what I just said
17:52.40Assidremiss: which provier
17:52.55wasimWinkie: i'm sure its in the days, if not weeks
17:52.55mikefooAssid: sorry went to get some food..
17:53.01wasimWinkie: if not more ...
17:53.16SpaceBassFlauto,  to the point my wife is threatening to move out and get her own POTS phone service....now I'm forced to chose...asterisk...wife...asterisk...wife...(fyi, wife will win)
17:53.38kpettitI have major echo problems in my confrence bridge with the IAXY's
17:53.52kpettitwondering if it's device, asterisk or both
17:53.56Winkiewasim: i'm sure it probably is :)
17:54.05SpaceBassmine is like hearing yourself back through Kurt Cobain's guitar amp
17:54.18sdaliNewbie duh type question - Why does the Voicemail application not recognize when a caller has pressed "#"? I'm expecting it to finish recording the msg, and then fall through.
17:54.25Flautohehe, spacebass
17:54.26WinkieSpaceBass: nicely overdriven?
17:54.27kpettitAny way to get rid of that echo??
17:54.28Flautosorry to hear that
17:54.45*** join/#asterisk Givur (n=mail@Gb95f.g.pppool.de)
17:54.47GivurHi all
17:54.49sdaliHere's 3 lines from extensions.conf
17:54.53SpaceBassWinkie,  I've played with the RX and TX a lot
17:54.56remissWinkie: http://pastebin.com/559852
17:55.02sdaliexten => 3,1,Voicemail(3456)
17:55.02sdaliexten => #,1,Playback(thank-you-for-calling)
17:55.02sdaliexten => #,n,Hangup
17:55.04remissAssid: televoip.. norwegian..
17:55.07kpettitI've got audiobuffers=32 set in my meetme.conf but that dosen't seem to help
17:55.40kpettitSpaceBass, you talking fax?
17:55.48Winkieremiss: shouldn't your type be friend?
17:55.53Flautospace, i have the same echo problem or other party can not hear me
17:55.57SpaceBasskpettit,  fax is another issue I have...but no I was tlking voice
17:56.03austinnichols101is there a way to do a factory reset of an SPA3000 from the web interface?  I need to reset one at a remote site...
17:56.04Flautoeither one, x100p does not work well
17:56.06Winkiebut regardless, i'm going home and i can't see much else wrong, username/secret is fine from what i can see
17:56.06kpettitah.
17:56.06*** join/#asterisk ToTo (n=ToTo@host97-136.pool875.interbusiness.it)
17:56.15kpettitWorking on fax now.  Got it working really nice with iaxmodem and hylafax
17:56.22austinnichols101manual says nothing
17:56.26*** join/#asterisk Primer (n=vi@sh.nu)
17:57.02Flautoaustin, i guess no
17:57.06remissW`gon: read something about that... i don't think it will work with this provder...
17:57.10SpaceBassI had fax working really well using dring until I upraded to AAH 2.x...
17:57.24remissand it doesn't fix the problem either :-/
17:58.02kpettitSpaceBass, I had about 90% with spandsp and rx/tx_fax.  But I'm about 100% with iaxmodem and hylafax
17:58.06PrimerAnyone know how a provider charges for forwarded calls to PSTN when voip is unavailable?
17:58.16kpettitplus you get all the cool stuff that normally comes with hylafax, makes it nice
17:58.52stoffellkpettit, did you use an existing howto?
17:59.05kpettitno, i'm making one though
17:59.10Primerwe're looking to move our support 800 number to voip, but we need to ensure that it's always available. Seems some providers will route calls over the normal phone network if asterisk can't be reached over the internet
17:59.18kpettitIt's kind of gentoo specific sense that's what I use, but it should work for about anybody
17:59.31kpettitI'll have a good wiki on it, here in a day or two.  Just about have it done
17:59.42stoffellkpettit, ah, i see, great info is always appreciated, be it on the voip-info, or linked from there.. tnx
17:59.52kpettitInstalling it on a few more system to make sure it works like I think.  I've got this same setup on 5 machines now.
17:59.56kpettitone with 200 fax did's
18:00.07stoffellhehe :)
18:00.15remissProxy-Authenticate: Digest realm="84.48.68.129", nonce="43f610347d676648aa833a782f4dc569a3a530d3" <-- can this possibly be right?
18:00.36Flautois there anyone would help me with a2billing?
18:00.40kpettitThe guy that wrote iaxmodem has been helping us.  It's been really cool, guess he's also the maintainer to hylafax
18:01.07stoffellah, fun, guess that'll work out then ;)
18:01.09kpettitContracting him for a week is sure alot cheaper than buying one of those $5000 fax boards.
18:02.15sdaliAnyone know why the Voicemail() doesn't stop recording when a caller hits #?
18:02.18stoffellgreat!
18:02.27Qwell[]sdali: Bad dtmfmode?
18:02.44*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
18:02.45stoffellhaven't tried iaxmodem yet, but will also have look in int now I know you're using it big-scale:)
18:02.51sdaliNope. Because it "hears" the correct mailbox to use.
18:03.22sdaliQwell[]: Nope. Because it "hears" the correct mailbox to use.
18:04.08Qwell[]sdali: Don't you call it with Voicemail(12345), you said?
18:04.54sdaliQwell[]: exten => 3,1,Voicemail(3456)
18:04.55sdaliexten => #,1,Playback(thank-you-for-calling)
18:04.55sdaliexten => #,n,Hangup
18:05.09Qwell[]yeah...so...check your dtmfmode
18:05.49sdalidtmfmode=rfc2833
18:05.57Qwell[]and on the device?
18:06.01Qwell[]or whatever
18:06.20sdaliI know it's hearing the dtmf digits correctly because a message does go to the correct box
18:06.46Qwell[]no, because when you pass in digits to Voicemail...that isn't dtmf
18:07.14sdaliIs my basic understanding bad? I'm expecting it to har the #, then playback the thank you messge.
18:07.31Qwell[]No, because your dtmfmode is likely wrong, so it can't interpret any digits
18:07.39areskiFlauto, I can help u a little bit if u have straight question :)
18:07.55sdaliBut getting to the correct mailbox requires correctly interpreting the dtmf tone I selected to get to that particular mailbox.
18:08.15Qwell[]sdali: you don't use dtmf to pick a mailbox, if you pass one in
18:09.04xachenIf I do an extensions reload when there are calls going on, will that drop my calls?
18:09.18Qwell[]xachen: No, but things could get b0rked
18:09.18sdaliQwell[]: In the above code from my ext.conf, you see that they have to hit 3 to get to the vm box.
18:09.38*** join/#asterisk Dr-Linux (n=Nothing@202.125.141.8)
18:09.44Qwell[]Thank you
18:10.08*** join/#asterisk justinu (n=justin@72.18.13.34)
18:10.14salviadudhey! im trying to get toll free numbers, still no cigar http://pastebin.ca/41968
18:10.23salviadudyou guys got any ideas?
18:10.24sdaliso it seems to me that the dtmf is being interpreted correctly
18:10.39SpaceBasssdali,  you running asterisk@home?
18:10.53sdalinope, just straight asterisk 1.2.4
18:10.56*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
18:11.15SpaceBasshave # anywhere in your dialplan being used as a transfer?
18:11.36sdaliNope
18:11.44sdaliMaybe I should verify that
18:12.04SpaceBassdoes the CLI show anything when you press # in voicemail?
18:12.10SpaceBass(also is # supposed to do anything in VM?)
18:12.28Qwell[]# should say you're done recording
18:12.29Dr-Linuxhi _Sam--
18:12.39sdaliCLI doesn't hear it - or show anything
18:13.01harryvvhi spaceballs
18:13.02*** join/#asterisk afrosheen (n=test@txprotoa2.august.net)
18:13.14harryvvwho here has delt with unlimitel?
18:13.17SpaceBassHarrrry, how are ya :)
18:13.34afrosheenhas anyone had problems with DTMF via Zaptel suddenly not working very well?
18:13.36harryvvsixtel is having issues.
18:13.48SpaceBassafrosheen, not working at all, or just intermittent?
18:13.58harryvvit rings once on there then then my * rolls over to the local long distance carrier.
18:14.03afrosheenSpaceBass: intermittent, our incoming calls aren't always hitting their target extensions
18:14.14harryvvI need to put a play file name saying its out.
18:14.32SpaceBassafrosheen,  hummmm what hardware?
18:14.40afrosheenSpaceBass: a pair of tdm400's
18:15.14afrosheenI tried rearranging the modules because I had a bad one last year but the problem persists
18:15.24*** join/#asterisk LoonaTick (i=LoonaTic@office.contrabandict.nl)
18:15.36LoonaTickhi
18:15.43starwareznewbie question, Im stuck with sip clients behind nat, i read the documentation but i still have problems, using a public stun server... my question is: is posible to make a call between 2 cliens behind the same nat via an remote non-nat asrterisk server?
18:15.58justinuonly if you set reinvite=no
18:16.01sdaliThe CLI says: Executing VoiceMail("SIP/gs2-ebec", "1234") , recording the msg, then stopped after a silence of 10 seconds, despite the fact that I've hit # several times
18:16.22LoonaTickanyone know where I can find settings for zapata.conf to install my telephone line? My telephone company didn't deliver them (yet). I have a Dutch KPN ISDN30 line (PRI)
18:16.34afrosheenstarwarez: we solved that with our firewall by adding a second NIC to our * server and having one connection inside the firewall and one outside
18:16.56justinuLoonaTick: http://www.digium.com/asterisk_handbook/zapata.conf.pdf
18:16.56LoonaTick(is there perhaps a list with defaults available)
18:17.05LoonaTickthanks, will look at that one:)
18:17.35starwarezi tried nat=yes on the asterisk´s client config is not working
18:17.57afrosheenstarwarez: that works for single nat but not double nat
18:17.59justinunat=yes, canreinvite=no
18:18.22*** join/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com)
18:18.25afrosheenstarwarez: i.e. your server is firewalled/nat'ed and the client is behind a router or whatever, that's double nat and is a nightmare
18:18.31jontowawww.. wind took out the power in the CO and i lost a 225 day uptime on my voicemail machine :( :(
18:18.34justinuhe said his asterisk wasn't natted
18:18.42justinuat least thats what I gather
18:18.45warthawgmy story on asterisk on openwrt is up on newsforge
18:18.53warthawghttp://mobile.newsforge.com/article.pl?sid=06/02/09/1727256&tid=104&tid=132
18:19.00LoonaTickjustinu: Thanks, but that doesn't contain any default values to get this specific line to work.. was hoping someone here maybe had the same situation.
18:19.02starwarezno, no double nat here... * is not nated
18:19.06SpaceBassanyone have a recomendation for a french DID provider?
18:19.06warthawgit's a wonderful story
18:19.13justinuLoonaTick: euroisdn?
18:19.19LoonaTickjustinu: yes
18:19.22afrosheenoh if it's single nat then you don't have to worry about a stun server, just set nat=yes in your config on the server
18:19.27justinud channel should be on timeslot 16
18:19.32justinuswitchtype => euroisdn
18:19.44Dr-Linuxhi justinu
18:19.54LoonaTickjustinu: Thanks, and the signalling?
18:19.55Dr-Linuxi made something very good, but i a bit problem :)
18:20.03afrosheenDr-Linux: how did it taste
18:20.09justinupri_cpe
18:20.23*** join/#asterisk saftsack (n=saftsack@p54A7FEB2.dip.t-dialin.net)
18:20.30starwarezi have no audio, i enabled sip debug but i cant find the error :S
18:20.41LoonaTickthanks :)
18:20.57LoonaTickjustinu: Once the dc plugs the line back in i'll let you know if it works :)
18:20.57afrosheenstarwarez: well if you don't do nat=yes on a per-device basis then you should get one-way audio at least
18:21.08*** join/#asterisk PakiPenguin (n=bah@linuxpakistan/admin/pakipenguin)
18:21.08justinuLoonaTick: you also need zaptel.conf configured
18:21.10Dr-Linuxafrosheen: you will know when you suck it ;)
18:21.22afrosheenDr-Linux: well you said you bit a problem
18:21.23afrosheen;)
18:21.31justinuLoonaTick: you should prorbably use span=1,0,0,ccs,hdb3,crc
18:21.35justinuDr-Linux: what's up dude?
18:21.40Dr-Linuxhehe
18:21.51starwarezso, is possible to habe calls with 2 phones behind the SAME nat router?
18:21.58justinuyep
18:22.07afrosheenstarwarez: yeah
18:22.21LoonaTickjustinu: That's in the zaptel config I assume?
18:22.26justinuyes
18:22.37justinuthere's some other things that need to go in there too
18:22.40justinuthe channel definitions
18:23.04justinuLoonaTick: http://www.digium.com/downloads/configuring_zaptel.pdf
18:23.15afrosheenchannels, grouping, incoming/outgoing volume boost, etc. etc.
18:23.23warthawga question about agi:  does asterisk kill the process when it returns?
18:23.29[av]bani\o>
18:23.30LoonaTicki think i have that one correct, bchan on 1-15,17-31 and dchan on 16
18:23.30[av]bani<o/
18:23.38justinuLoonaTick: sounds good
18:23.47PakiPenguinWhat atas do you guys suggest for connecting ~6 extensions?
18:23.54LoonaTickbut I have span=1,1,0,css,hdb3,crc4
18:24.01starwarezi will remove the nat=yes to check if i have sound... :)
18:24.04Dr-Linuxjustinu: i setup this >> http://www.voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP
18:24.10justinuLoonaTick: it should be "ccs", not "css", imo
18:24.23LoonaTickccs, my typo, sorry
18:24.36Dr-Linuxeverything is very fine .. but it doesn't copy created file to /var/spool/asterisk/outgoing/ dir
18:24.54justinuLoonaTick: ah yes, your span = 1,1,0 is correct
18:24.57wunderkinLoonaTick, you need 1,1 if you are taking timing from the line
18:25.06wunderkini thought there were 2 d chans on an e1 pri
18:25.10justinuno
18:25.27afrosheenstarwarez: remove? no, you want it in there..there's no danger with it being there if it isn't needed but not the other way around
18:25.29justinuyou can ask for one, but it's kinda pointless
18:25.43wunderkini have nfi, i just thought i heard it mentioned
18:25.53justinubackup d-chans are useful on NFAS groups only really
18:26.09wunderkinthats what i figured
18:26.15LoonaTickwunderkin: I'm sorry, what does this timing mean?
18:26.35justinuyou have to sync your e1 tx source with the rx from the telco
18:26.38LoonaTicki see something with the distance of the other side
18:26.47justinuor else you'll get snap crackle and pop on the line
18:26.53LoonaTickii see
18:27.19LoonaTickthe telco delivered a hdsl modem, from there there's a line to the E1 interface
18:27.40*** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net)
18:27.48justinuyep
18:28.06LoonaTickwill it only affect the quality or can it cause the line to not work?
18:28.14justinugenerally it will still work
18:28.33justinubut it could be slipping badly enough to cause your d-channel to stay out of service
18:28.37Dr-Linuxanyone is using this wakeup call:
18:28.46justinuDr-Linux: wake up!
18:28.46Dr-Linux<PROTECTED>
18:29.09starwarezhere is my lab install :)   (lan)--net=192.168.1.x--|nat router|--net=9.9.9.x--|router|---net=1.1.1.x--(*)
18:29.30LoonaTickjustinu: Might be a really stupid question, but can you tell me the difference between B and D channels?
18:29.41W`gonstarwarez: assuming you don't own 9.9.9.x and 1.1.1.x perhaps you should use a real subnet? :)
18:29.46warthawgaccording to cnn, the whippet lost at the airport was "carried away by monkeys"
18:29.50justinub = bearer
18:29.53WinkieLoonaTick: technical or actual?
18:29.56PakiPenguinguys , can someone help me out with  fxs atas? i need 6 ports ... any suggestions with atas or some other solution?
18:30.01LoonaTickWinkie: actual
18:30.03SpaceBassarrruuggg this stupid frenchy website won't process my credit card and give me my damn sip account
18:30.04justinubearer channels bear the audio of the call
18:30.10WinkieLoonaTick: b carries signal, d carries signalling
18:30.11justinud = data (signalling) all call setup info
18:30.25LoonaTickaah that really explains thanks :)
18:31.28justinuisdn can do freaky stuff like ask for a call to be setup on 4 b-channels at once
18:31.36justinufor wideband video conference, for example
18:31.47justinuthey don't have to be carrying audio
18:32.03LoonaTickah
18:32.25justinui think there are very few people using features like that though
18:32.37Abydos313what ports do i need to open for sip users to get sound? i have 5060-5082 udp and 10000-20000 udp forwarded to * server
18:32.52LoonaTickquite old numbers, 64kbps (i think?) per channels "64k should be enough for anyone" :)
18:33.00Dr-Linuxjustinu: i'm awaking :P
18:33.06justinu64kbps actually has a purpose
18:33.14justinuit's due to the nyquist frequency
18:33.22SpaceBassAbydos313,  that should be it
18:33.31justinuyou need 8000 8bit samples/sec to get 0-4000hz frequency responce
18:33.40SpaceBassassuming 10000-20000 is what is defined in your rtp.conf.. (and you can GREATLY narrow that down)
18:34.01Abydos313then what else can i check? when i connect vpn sound works great, but thru firewall..nada but dialtone
18:34.33LoonaTicki see
18:34.46SpaceBassAbydos313, missed the issue at hand...assuming remote phone cannot connect?...do you have net=yes on that exten?
18:34.56justinuDr-Linux: hehe
18:35.41LoonaTickthat's quite beyond my current education :(
18:35.48sivanais there a background cdr process that kp wrote? Anyone know what I'm talking about?
18:36.25justinuLoonaTick: nyquist theory simple says "you need double the sampling rate for the frequency range you want to sample"
18:36.32justinu(heavily paraphrased)
18:36.36Abydos313SpaceBass yes
18:36.43Dr-Linuxjustinu: i'm looking for _Sam-- ;)
18:36.43justinuso if you want 4000hz, you need 8000 samples/sec
18:36.48SpaceBassAbydos313,  any audio? like one-way?
18:36.50justinuDr-Linux: i'm surprised he's not here!
18:36.56Abydos313nope
18:37.01Dr-Linuxjustinu: he doesn't start ... but he gets start then never stops :P
18:37.21SpaceBassAbydos313, you might want to try opening TCP ports too...crazy as that sounds, I've had success when the UDP stuff fails...maybe how the firewall processes the requrest or something
18:37.30Abydos313ok
18:37.40Abydos313it's a sonicwall.. so maybe it's alittle diff
18:37.53LoonaTickah that explains a little, thanks :)
18:38.09*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
18:38.17SpaceBassAbydos313,  I was helping some poor guy via IM that had the exact same problem with a sonicwall...never figured out what it was
18:38.21Abydos313which tcp ports? the 5060-5082?
18:38.30SpaceBassVPN helped him, but it changed the MTU and caussed jitter
18:38.37SpaceBassAbydos313,  should only need 5060 i think
18:38.40Abydos313that sucks
18:39.05Abydos313i could try the dmz but i'm testing on my work machine with vmware :)
18:40.07SpaceBassAbydos313,  ahhh it could be that!
18:40.13SpaceBassusing bridged or host networking?
18:40.44Abydos313the default.. the vmware machine has it's own ip etc
18:40.45justinuLoonaTick: 8000samples/sec at 8bits/sample = 64kbps
18:40.52Abydos313i think that is bridged
18:41.00justinuLoonaTick: and that concludes your basic telephony lesson today
18:41.06Abydos313haha
18:41.16LoonaTickhehe indeed, thanks
18:41.33SpaceBassarrruuuggg Im starting to think Asterisk hasn't made it to France yet...i just need a damn sip provider and paris DID
18:42.01justinui think that voxbone is offering paris DIDs
18:42.14SpaceBasseurika!
18:42.17SpaceBassthanks justinu
18:42.27*** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41)
18:42.45justinunp
18:43.20*** join/#asterisk Eggplant (i=No@64.233.107.198)
18:43.43SpaceBassjustinu,  what do you know about voxbone? I dont see pricing anywhere
18:44.26justinuit's not overly expensive
18:44.34justinu5 bucks a month/did? something like that
18:44.36justinuit's prepaid also
18:45.11Eggplantanyone use astlinux? The listed default login is not working, did they change it between the livecd adn live cf versions?
18:45.45starwarezWhat is the best sip client for "behind nat client" and "non-nat asterisk" installation
18:47.33Juggiestarwarez, the best sip client is common sense.
18:47.38jbalcombAny good resources out there dealing with SIPAddHeader? (please don't <tilde>docs)
18:47.41Juggieunderstanding nat can make even the worst configuratino work fine.
18:47.55Juggiei have *<-nat->internet<-nat->clients
18:47.58Juggieand they work perfect
18:49.27starwarezJuggie:I cant find the way to debug the problem... :(
18:50.25starwareztricky configurations on the sip clients are needed too?
18:50.29Dr-Linuxjustinu: i was on the phone with a US girl
18:50.56LoonaTickjustinu: I have the new configurations; got the first asterisk message, but don't know if it is good or bad:
18:50.56LoonaTickFeb 17 19:50:09 NOTICE[21264]: chan_zap.c:8171 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1
18:50.56LoonaTickFeb 17 19:50:10 NOTICE[21264]: chan_zap.c:8171 pri_dchannel: PRI got event: HDLC Overrun (7) on Primary D-channel of span 1
18:51.07justinubad
18:51.08shido6vmware and asterisk, Abydos313 ?
18:51.09LoonaTickI assume bad though
18:51.21justinuHDLC errors indicate your d-channel is not stable
18:51.29justinuDr-Linux: again?
18:51.35Juggiestarwarez, no, no special config is needed on sip clients
18:51.50Juggiedo all the work on * and the sip clients wil work easily :)
18:51.59LoonaTickjustinu: Ok, does this mean there's a problem at my telco's (or the modem's) side, or at my configuration?
18:52.03Dr-Linuxjustinu: yes she is in chat, and she wanted to talk to me and she was crying on the phone :S
18:52.08Juggiestarwarez, explain your setup and yuor problem.
18:52.19justinuLoonaTick: unknown, you will have to use loopbacks to isolate the problem.
18:52.20Juggieand do it concisely as possible please.
18:52.49LoonaTickgreat, thanks
18:52.54starwarezJuggie: sure... lemme post it..
18:53.11LoonaTick(for example I have to put a crossover cable in and let them call each other?)
18:53.45justinunot cross over
18:53.50justinuthat's for hooking up ast back to back
18:53.58justinuyou need a loopback plug
18:54.08justinuor ask the telco if they can loop up your CSU
18:54.23LoonaTickthanks!
18:55.49*** join/#asterisk JCC_ (n=john@207.41.92.131)
18:56.03*** join/#asterisk L|NUX (i=linux@203.101.160.218)
18:57.06*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
18:57.55SpaceBassarrruuggg....this echo is unbarable
18:57.57LoonaTickjustinu, correct that the HDSL modem is the DSU, and the card the CSU
18:58.15SpaceBassmy father just called and accused me of being drunk b/c I was sluring and studdering b/c all i could hear was myself
18:58.15jbalcombAny good resources out there dealing with SIPAddHeader as standard functions such as Call-Info, Alert-Info, etc?
18:58.51justinuLoonaTick: HDSL modem is called NIU
18:59.02justinuyour E1 card is called CSU/DSU
18:59.08jbalcombNetwork Interface Unit
18:59.08Qwell[]Anybody happen to know how I can access feature codes on a nortel meridian, if my phone doesn't have a feature button?  heh
18:59.16*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
18:59.21justinubut no DSU really, because you're only running channelize voice
18:59.50LoonaTickah ok
19:00.17LoonaTickand the loopback cable, is that a special cable or a regular cable with some setup?
19:00.17Abydos313s
19:00.18justinuit's usually a small plug
19:00.21justinuit takes pins 1,2 and connects them to 4,5
19:00.30justinuyou can make your own cable that does it as well
19:00.39LoonaTickah i think one of those is connected with a plastic thing to the HDSL modem
19:01.05justinucould be, they often come with adtran CSU equipment
19:01.32LoonaTickand with the loopback in instead of the cable to the NIU, it should not give those signal errors anymore? Should it give any other message?
19:01.50justinuyour d-channel will not come up to itself
19:02.07justinuwhat you need to do is plug the looback in to your end of the E1, instead of your E1 card
19:02.21justinuthen ask your telco to run BERT patterns on the line to verify the line quality
19:02.51starwarezJuggie/people:here is my setup   http://starwarez.no-ip.org/asterisko.txt   i think is lohn to paste it here :)
19:02.51LoonaTickah thank you
19:03.59LoonaTickmy telco did some testing with that small plug (labelled: PTT ISDN30 test), said there was no data coming over the line or something like that, so I think the problem is on their side then
19:04.22LoonaTickbut when the test block was plugged out he could see that the line was up upto the hdsl modem
19:04.38Juggiestarwarez, that doesnt tell me where asterisk is
19:04.45Juggieah
19:04.46Juggienm i see
19:05.03starwarez*
19:05.05starwarez:)
19:05.06Juggieyah
19:05.10Juggieso then * has a routeable ip
19:05.30Juggieso the clients are the only ones that dont have routeable ip's.
19:05.37Juggiedid you set nat=yes for the clients
19:05.49Juggieso that asterisk uses the ip from the tcpip packet rather then what the client reports
19:06.09Juggiethat really should be all you have to do.
19:06.30Juggiedoes your client hear audio but * not hear the client?
19:06.35Juggieis one direction working? but the other not?
19:06.38sdaliQwell[]: I found the problem - dialing into the FXO port of a Grandstream HT488, and forwarding that call to a VOIP extension, the * and # dtmf digits are not being handled properly. Grandstream is aware of this and have told me to wait for the next firmware upgrade.
19:06.40Juggiewhats the situation
19:06.40LoonaTickjustinu: Oh one config option I left on the default: rxwink=300. Can this matter?
19:06.44starwarezyes, nat=yes didnt did the job
19:06.59Juggiestarwarez, didnt do the job?
19:07.12Juggiestarwarez, does any audio path work?
19:07.13Qwell[]sdali: grandstream makes crap...
19:07.19sdaliI see. :)
19:07.19Qwell[]sdali: I'd suggest getting another device
19:07.27sdaliThanks.
19:07.32starwarezno sound from any client
19:07.54Juggiecan the clients hear *?
19:08.12starwarezyes, the retension music can be heared
19:08.18justinuLoonaTick: winking isn't used on PRI, so it will likely not do anything
19:08.31Juggiestarwarez, place a call and do a rtp debug
19:08.36LoonaTickah ok, thanks
19:08.37Juggieso you can see whats going on with the rtp traffic
19:08.40starwarezok
19:08.57salviadudi got it workin'!!!!
19:08.59kuku5anyone having issues with ther cisco phnes?
19:09.00salviadudpiweiwiwiewe
19:09.04kuku57960/40
19:09.16LoonaTickjustinu: If my telco tests the line with the loopback block plugged in, I should see data coming in and out, right?
19:10.24*** join/#asterisk nain (n=nain@202.125.143.66)
19:11.05nainhi
19:12.34justinuno... you're plugging the line into the loopback plug, which loops their tx signal back to them
19:12.39justinuso you get nothing
19:12.46justinuyou shouldn't have anything plugged into your E1 card
19:13.22LoonaTickjustinu: I mean the traffic lights on the HDSL modem btw, your statement still applies then?
19:13.52*** join/#asterisk Xen^ (i=linux@203.101.164.215)
19:15.31DaPrivateerIf anyone can help. I'm on FreeBSD and when compiling I keep getting utils.o: In function `gethostbyname_r':
19:15.31DaPrivateerutils.o(.text+0x3eb): undefined reference to `__h_error'
19:15.54justinuyes, but those lights are probably always going to be lit
19:16.26LoonaTickok
19:16.47LoonaTickby the way, I have it on span=1,1,0,ccs,hdb3,crc4 now
19:16.54LoonaTickand I don't get the error anymore
19:17.24LoonaTickis that good, or just a stupid idea?
19:17.36GerbilWrkis it possible to setup with a queue, to auto not send the calls to an agent if it's a certain time, or to only send calls to a specific agent at certain times without the agents logging in and out?
19:18.16GerbilWrkI have 7 agents that always need to be in the queue, and one that only needs in it a few hours a day
19:19.56starwarezJuggie here is it: http://starwarez.no-ip.org/asterisko2.txt
19:21.00nainh
19:21.02nainhi
19:21.19nainhello is every body
19:24.07*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
19:24.37*** join/#asterisk gammacoder (n=chatzill@207.67.51.233)
19:24.37LoonaTickjustinu: and by the way, even though the line still gives a network error
19:25.00warthawgyes
19:25.07warthawgyes
19:25.23justinuLoonaTick: network error?
19:26.00LoonaTickjustinu: that's when i call the phone number
19:26.30justinuLoonaTick: pri show status?
19:26.45justinuer pri show span 1
19:27.10LoonaTickPrimary D-channel: 16
19:27.10LoonaTickStatus: Provisioned, Down, Active
19:27.29justinuso that's why
19:27.31justinuno d-channel
19:28.02nainCan any body now any Predictive Dialer for Asterisk ?
19:28.14starwarezJuggie: were you able to see the sip log?
19:28.52LoonaTickjustinu: Ah thanks, didn't know that command, does this mean the problem is at the telco's side?
19:28.57stoffellnain: try here, http://www.voip-info.org/wiki/view/Predictive+dialer
19:29.38iCEBrkrWerd.
19:30.21LoonaTickjustinu: (or that the zaptel.conf is configured wrong?)
19:31.05nainStoffell: there is gnudialler for linux that is open source i am looking for windows based dialer which can be configured with asterisk
19:31.38justinuLoonaTick: unknown
19:31.43stoffellnain, there is one on that page, but don't expect it to be open source..
19:31.45justinuyour zaptel.conf is probably ok
19:32.16naini think that is sinedialer ... and that is commercial can't afford
19:32.33LoonaTickok, thank you *very* much, i will call the telco and check out their opinions :)
19:33.13*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)
19:33.49warthawgyou guys sound like a bunch of phreaks
19:34.06Abydos313phreaking went out years ago :)
19:34.06*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
19:34.07LoonaTickwarthawg: Thanks for sharing
19:34.14warthawgheh
19:34.30nainAny Open Source Predictive Dialer for Windows ??????
19:34.31warthawgi can't understand 3/4 of what is said here, it's all telephony talk
19:34.52Abydos313you need two books. voip for dummies and the asterisk book..haha
19:35.02warthawgi got the asterisk book
19:35.09warthawgi guess i need to the voip one :)
19:35.11*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:35.15Abydos313i just bought it myself two days ago. i want to learn
19:35.39Abydos313i actually was unable to find a voip for dummies at borders
19:38.11*** join/#asterisk oogle (n=jart@justin.ctlinc.com)
19:38.24starwarezbuds... here is the log: http://starwarez.no-ip.org/aster.txt
19:40.32GerbilWrkhas anyone found a way to imporve QoS with Teliax?
19:40.49iCEBrkrnain: You can buy my dialer :P
19:41.03starwarezhttp://starwarez.no-ip.org/asterisko2.txt------> full log here
19:41.09ooglehas anyone ever dealth with voip reach?
19:41.14*** join/#asterisk _Paulo_ (n=Paulo@200-168-112-132.dsl.telesp.net.br)
19:41.15iCEBrkrstarwarez: here man use this
19:41.16iCEBrkr~pb
19:41.17jbotextra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
19:41.31*** join/#asterisk ^Accie^ (n=chrake@void.leapfrog.se)
19:42.22^Accie^anyone who'd like to help a newbie... umm... managed to set up asterisk and can dial the demo from my snom190 but I can't seem to register it with asterisk even though I'm using the same username/password stated in sip.conf
19:42.40^Accie^sip debug keeps telling me 'registration failed'
19:43.14remissw00t
19:44.16*** part/#asterisk warthawg (n=warthawg@cpe-66-68-176-215.austin.res.rr.com)
19:44.45starwarezwhat means "SIP/2.0 488 Not acceptable here"????
19:45.11justinucodec problem
19:46.06*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
19:46.23starwarezjustinu: without nat it works fine that phone...
19:48.57justinugenerally 488 is telling you that the two UAs could not find a compatible codec
19:49.28_Sam--trixter...you stole someone elses idea
19:49.42_Sam--there was a guy here already doing the free toll free terminations and paying people
19:49.45_Sam--i forget who it was
19:50.39_Sam--you thief you
19:52.05nainCan any one know about windows Based Open source Predictive Dialer
19:52.33starwarezactivating nat on clients on sip .conf i get: the following message:
19:52.35starwarez<PROTECTED>
19:54.03_Sam--nain:  nobody knows because one isnt there
19:54.24_Sam--i could be wrong.
19:54.35justinustarwarez: then your phone isn't sending the right SDP message
19:54.45justinupastebin the entire invite
19:55.30starwarezjustinu: you mean the sip debug?
19:55.39[av]bani\o/
19:56.09*** join/#asterisk backblue (n=moo@87-196-46-49.net.novis.pt)
19:56.12_Sam--nain:  class 'default', on IAX2/66.250.69.13:4569-3
19:56.14_Sam--er
19:56.19_Sam--nain:  http://www.voip-info.org/wiki/index.php?page=Predictive%20dialer
19:56.19justinustarwarez: yes
19:56.50*** join/#asterisk Renacor (n=kvirc@ip21.farheap.net)
19:57.09nainSam---: Sorry i have looked for it but it has no windows based Free Dialer
19:58.21remissbah... i had it working and i managed to remove the configuration :(
19:58.32starwarezjustinu: here is it
19:58.44starwarezsorry, http://starwarez.no-ip.org/lastcall.txt
20:01.03justinustarwarez: your phone is whacked... it sends the sdp on the first invite, but not the 2nd
20:01.41starwarezjustinu: bad configuration?
20:01.42trixter_Sam-- trixter...you stole someone elses idea  -- I started investigating this back in 1997 :P
20:01.51justinustarwarez: possible
20:04.36starwarezjustinu: the weird thig is: if i remove the nat from the routers, works like a charm
20:05.18Renacoris there a way for me to send a call to VM after it's been sitting in a queue for 10 min?
20:05.35Qwell[]Renacor: Sure, put a timeout on the queue, and have the next line be voicemail
20:07.09_Sam--trixter:  took you long enough to implement!
20:07.15justinustarwarez: perhaps without nat asterisk doesn't send the proxy auth required?
20:07.17_Sam--im just busting your balls..i hope you know
20:07.27_Sam--but there is someone on this channel doing the same thing already
20:07.52[av]bani_Sam-- got any phones other than gxp2k?
20:08.16robin_zahh, at last! ... updated software for my GXP 2000
20:08.16*** join/#asterisk websae (n=icechat5@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
20:08.17_Sam--at this place the only other phone i have here is a utstarcomm wifi phone
20:08.19mikefoohey guys if I was to make a dialplan, for when someone dials in, hits a certain extension it connects to someone, its still going through my pbx yes?  is there a way I can conenct him to another number and disconnect from my pbx?
20:08.38robin_zIronHelix: you there?
20:08.43justinubani: i'm here with the snom today
20:08.49justinustill want me to listen to the tones?
20:08.49[av]banijustinu: yay! er, boo!
20:08.50websaeanyone know of a good DID provider? I tried signing up for didx.org---but never was sent the confirmation email to complete registration---any other suggestions anyone ? thanks
20:08.56[av]banijustinu: yes
20:08.59justinuk, i'll let you know
20:09.07[av]banimake an extension which goes 1,Congestion()
20:09.07_Sam--robin_z :  where is the new firmware?
20:09.18[av]baniand one which goes 1,Playtones(dial)  n,Wait(60)
20:09.34robin_z_Sam--: sorry, I was using sarcasm as a low form of wit.
20:09.42_Sam--someone has the new firmware already
20:09.43_Sam--X-rob does
20:09.46_Sam--but its not public
20:09.48robin_znot me ...
20:09.51robin_zoh good.
20:09.52_Sam--and it fixes , somewhat, your display.
20:09.53[av]baniand one which goes 1,Playtones(busy)
20:10.01_Sam--he said "it takes longer before the display breaks"
20:10.59websaeanyone have a good DID provider?
20:10.59robin_zwell, thats an improvement already
20:10.59_Sam--maybe they will release something soon, who knows
20:10.59_Sam--he said he has 1.0.2.9
20:11.06robin_zgrr
20:11.07afrosheenwebsae: commpartners is good
20:11.13[av]banijustinu: the snom native congestion sound is really bizarre. playtones(busy) sounds exactly like what i get from the PSTN
20:11.21robin_zanyway ... I solved my SIP login problem ....
20:11.41robin_zthe remote objected to "astersik PBX" as the user agent ...
20:11.50robin_zI was setting it in the sip conf for that host ...
20:12.13robin_zbut, it was handing off to a sip proxy thing .. so I had to set it in [general] or the [sip-proxy-out]
20:12.56justinui'll check it out
20:12.59*** join/#asterisk Igbothom_III (n=HiltonT@office.quarkit.com.au)
20:13.17*** join/#asterisk mischko (n=Scott@cvo-cr1-200-242.peak.org)
20:13.30Abydos313would you guys recommend latest tar.gz or cvs setup for home users
20:13.49justinutarball
20:14.04Abydos313any reason?
20:14.12justinuunless you're participating in development, you don't want to run the code out of SVN
20:14.20justinuit could be unstable
20:14.25justinuin flux, etc.
20:14.26Abydos313ok thx
20:14.32RenacorQwell: Do you have an example of this?
20:14.37Abydos313even if you sync with stable?
20:15.54justinuyou can check out a stable branch of svn i suppose
20:15.59justinubut why not just get the tarball?
20:16.03mischkoSmall company about 5-15 phones. Only 2 incoming lines currently. We're thinking of upgrading to a VOIP phone system in-house with Askerisk/Linux to be our PBX and add a couple lines. Later we want to go to straight VOIP but only want to buy VOIP phones one time. Suggestions?
20:16.38[av]banido more research.
20:16.41Abydos313ok. i did do the latest tarballs, just curious. thanks for input
20:16.48justinunp
20:18.29stoffellmischko, try a few pones, buy 1 of each you think suits the needs, ...
20:18.46stoffellphones :)
20:19.15mischkoAre there any appliance solutions that I should be looking at rather than running my own Asterisk?
20:19.27[av]banishould be? depends on your budget.
20:19.29_Sam--you could use a hosted PBX and just have your phones connect to it
20:19.36_Sam--then you wouldnt need your own *
20:19.49[av]baniyeah, use SamSoft(tm) Hosted PBX Solutions(tm)
20:20.01stoffellmischko, depends what area you are, and what you want to spend, and what you know of linux :)
20:20.02mischkoVonnage does not offer us the ability to keep our existing phone number.
20:20.03mikefoohey guys if I was to make a dialplan, for when someone dials in, hits a certain extension it connects to someone, its still going through my pbx yes?  is there a way I can conenct him to another number and disconnect from my pbx?
20:20.19[av]banimischko: if you arent a linux guru, then * is not for you
20:20.37mischkoI'm quite familiar with linux but not phone systems. I'
20:20.46mischkove run servers on Linux no problem.
20:21.05[av]banican you compile applications from source, apply patches, edit config files?
20:21.10[av]banihandle iptables?
20:21.12mischkoyes.
20:21.14*** join/#asterisk fiftyCal (n=b@69-160-145-156.ontrca.adelphia.net)
20:21.30[av]banithen the only limit is your budget
20:21.31mikefoomischko: vonage doesn't do lnp's?
20:21.33stoffellmischko, then you could try a test system with asterisk@home, and see how it goes
20:21.47[av]baniyour budget will dictate what you can and cannot do
20:21.55mischkoVonnage doesn't have a local presence in this part of Oregon so we couldn't keep our local phone number.
20:22.11mischko??asterisk@home
20:22.14[av]baniyou havent stated your requirements though, so we dont know what to recommend
20:22.18austinnichols101asterisk@home is a good way to kick the tires without a huge investment in time
20:22.33*** join/#asterisk cassio (n=cassio@c91133b9.rjo.virtua.com.br)
20:22.44austinnichols101and [av]bani is right - budget is the major deciding factor on phones
20:22.56_Sam--what is wrong with hosted pbx is you are talking about 5 phones or so
20:23.00stoffellmischko, search on google, it's a good way to get to learn the basics of telephony and asterisk
20:23.03_Sam--other than, you dont have to worry about learning *
20:23.14mischkohosted pbx would be fine if we can keep our local phone number.
20:23.23[av]baninobody serves corvallis
20:23.37Qwell[]Why go to voip?
20:23.44cassioguys, I have 5 broadvoice lines, but I am getting this error on 2 lines, the configs on sip.conf are the same, why is this happening? http://pastebin.com/560087
20:23.44[av]banicuz its neat0
20:23.45Qwell[]Just get some analog cards, and use the lines you have
20:23.46mischkoLocal ISP will be adding VOIP in about 6 months.
20:24.04austinnichols101Qwell: cheap calls
20:24.08justinu31337
20:24.13mischkoWe don't like our existing Panasonic phone system and people call in getting busy signals.
20:24.22mischkoWe can't add new lines to this system.
20:24.26mischkoSo we're looking at options.
20:24.29Qwell[]So move to *, and get more lines...
20:24.32Qwell[]Don't need voip
20:25.07JCC_For what its worth, I set up an A@H box, turned off long dist calling, added voxee and a Sipura 3000 on an older 900MHz Pent III - total cost about $125 (the box was in the closet doing nothing)
20:25.11[av]banihmm.. jared likes his 7960 better thant he 7970...
20:25.16mischkoQwell, hadn't thought of that.
20:25.27austinnichols101I really like the 7960s
20:25.41Qwell[][av]bani: eh?
20:25.43stoffellmischko, indeed, like qwell says, you could do that, and in the future try voip phones, you can mix it all
20:25.46Qwell[]7970 is FAR better than a 7960
20:25.49austinnichols101but they're $300-$350 each
20:26.02[av]baniQwell[]: jared mauch. says he likes the 7960 better... weird
20:26.10austinnichols101Qwell: yeah, I have phone 7970-envy right now
20:26.11Qwell[]he's dumb then
20:26.18[av]banihe has problems with sccp
20:26.20[av]banimight be why
20:26.21Qwell[]7970 does everything the 7960 does...in color
20:26.27Qwell[]Then it's his fault, not the phones
20:26.33Qwell[]tell him I'll buy his 7970 for $100
20:26.56austinnichols101I'll trade him for a 7960 :)
20:27.18austinnichols101~seen opsys
20:27.30jbotopsys <n=opsys@68-235-141-52.miamfl.adelphia.net> was last seen on IRC in channel #asterisk, 4d 14h 46m 28s ago, saying: 'betaboi" true'.
20:27.30[av]banithe avaya phones look nice
20:27.31mischkoWhat phones work best with *?
20:27.44[av]baniavaya 4630sw...
20:27.48austinnichols101somebody was on here yesterday raving about the mitels
20:27.48mischkoI know it's supposed to be almost hardware agnostic but ?
20:27.59[av]banimischko: phones which speak SIP
20:27.59Qwell[]austinnichols101: probably Juggie
20:28.21mikefoohey guys if I was to make a dialplan, for when someone dials in, hits a certain extension it connects to someone, its still going through my pbx yes?  is there a way I can conenct him to another number and disconnect from my pbx?
20:28.46austinnichols101Qwell: sounds right.
20:28.50mikefoowhat I am trying to accomplish is connecting two people and me getting of the way.
20:28.57austinnichols101Last thing I bought from Mitel was a dialer
20:29.06_Sam--has anyone heard of ESI phones?  i was at a doctors office and was checking them out, they seem nice, but have no idea of they are
20:29.15[av]baninever heard of them
20:29.24_Sam--search google esi phones
20:29.27_Sam--first result
20:29.34[av]banihttp://search.ebay.com/search/search.dll?sofocus=unknown&sbrftog=1&catref=C6&fstype=1&from=R10&satitle=esi+phone&sacat=-1%26catref%3DC6&bs=Search&fsop=1%26fsoo%3D1&coaction=compare&copagenum=1&coentrypage=search&sargn=-1%26saslc%3D2&sadis=200&fpos=97526&ftrt=1&ftrv=1&saprclo=&saprchi=  ?
20:30.12_Sam--i think maybe their phone are only usable with their own servers
20:30.14austinnichols101anyone tried the allworx phones?
20:30.30[av]banithey dont mention sip at all
20:30.45austinnichols101found a nice comparison chart from them (the cisco retail shows way off): http://www.allworx.com/XQ/ASP/p.3406/QX/pdfs/PhoneComparisonChart.pdf
20:30.46_Sam--this one says IP:   pbx?
20:30.46_Sam--<austinnichols101> Qwell: sounds right.
20:30.47_Sam--er
20:30.53_Sam--http://www.esi-estech.com/products/systems/phones/remote/
20:31.02mischkoAnything on here: http://www.iptel.org/info/products/sipphones.php that I should definitely stay away from in hard phones?
20:31.03[av]banithey talk about integrating esi phones with microsft outlook.
20:31.13Qwell[]mischko: grandstream
20:31.24austinnichols101sam: sounds right about what?
20:31.38_Sam--that was a mess up copy/paste
20:31.41_Sam--somehow that was in my buffer
20:32.03stoffella good-looking phone: thomson st2030, but doesn't support MWI and BLF yet in SIP image..
20:32.06[av]bani_Sam--: looks like no sip
20:32.14[av]bani_Sam--: looks proprietary, like mitel
20:32.27_Sam--figured...it was a really nice office system they had.
20:32.28mischkoSo avoid mitel?
20:32.45[av]banimischko: mitel makes some SIP phones, but mitel is pretty expensive for what you get
20:32.50Qwell[]mischko: mitel is supposed to be really good
20:33.06[av]banimischko: most mitel phones are either analogue or speak proprietary mitel protocol
20:33.31afrosheenis it safe to mix zaptel versions, i.e. the latest zaptel 1.2.4 with asterisk 1.2.0?
20:33.33Juggiewrong.
20:33.33stoffellmischko, try ThomsonST2030, Aastra, Polycom, ..
20:33.36mischko"proprietary protocol" means somethings it doesn't do SIP?
20:33.37Juggiemitel makes all sip phones now.
20:33.41[av]bani~phones
20:33.43jbotit has been said that phones is at http://bani.anime.net/phones/
20:33.47Juggieall their phones support minet/sip
20:33.53Juggieanything 5215+
20:34.17Juggie5215/5220 require the dual mode model (dual firmware)
20:34.18_Sam--he's right
20:34.21_Sam--"Support for SIP and MiNET Protocols "
20:34.21[av]baniJuggie: mitel is really expensive for what you get. cisco and others are better bargain
20:34.22Juggieand everything else supports sip nateivly
20:34.28Juggieav, cisco sucks
20:34.31mischko[av]bani, ok.
20:34.34Juggiethe phones are impossible to manage
20:34.50[av]baniJuggie: mitel design their phones to be integrated into their pbx systems. friend fo mine is a mitel reseller
20:34.51mischkoWho "doesn't suck" and is reasonably priced?
20:35.00Juggiei mean common, editing like 5 files to update the firmware?
20:35.00oogledon't use the 12-button mitels
20:35.00Qwell[]mischko: polycom
20:35.11[av]baniJuggie: polycom is the same way
20:35.11*** join/#asterisk veto (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
20:35.12*** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com)
20:35.21Juggieav, i have 3 mitel phones next to me
20:35.25Juggieone is minet
20:35.29Juggietwo are on asterisk
20:35.30mischko[av]bani, "same way" = "hard to manage"
20:35.33Juggieand they work perfect
20:35.35ooglethe 24-button mitel is pretty good, and allows dhcp-tftp provisioning
20:35.35mischko?
20:35.41[av]banimischko: yes
20:35.41Qwell[]Juggie: When are you gonna send me some of the "other stuff" sitting next to you? :p
20:35.49mischko[av]bani, who's easy to manage?
20:35.52mikefoois there a reason why I shouldn't use asterisk@home on a production level?
20:36.07Qwell[]mikefoo: Because it's complete junk, primarilly
20:36.14Juggieheh.
20:36.17[av]banimischko: linksys, snom, grandstream
20:36.17austinnichols101omg: the question!
20:36.20ooglewhat's wrong with asterisk@home?
20:36.21Juggiethats the big thing yes :)
20:36.22[av]banimischko: not that i would recommend any of those
20:36.27ooglei've never used it myself, j/w
20:36.40mikefoohah, ahh ok didn't realize that, thought it was just a dumbed down version.
20:36.49Juggiemitel phones support http for config files/firmware
20:36.54mischko[av]bani, what would you recommend?
20:36.55Juggiewhich removes havnig to run tftp shit
20:36.56afrosheenfrom what I've seen on the more expensive snom's, they have decent management built in
20:36.58stoffell[av]bani; you're right, but 'what' would you recommend?
20:37.01[av]banimischko: what are your requirements and budget?
20:37.14afrosheenbut we're 100% polycom here
20:37.14[av]banistoffell: depends. define your budget and requirements
20:37.15JuggieWHICH also, ets you use apache rewrite to dynamically create your config files
20:37.18Juggiewhich like php or something
20:37.24*** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net)
20:37.26SplasPoodWhat are people generally using /w 1.2.X to hand LCR?   Generally custom development, or is there something off the shelf that's populaR?
20:37.26SpaceBasshey again
20:37.27[av]baniafrosheen: snom have nice webadmin, but the phone UI is poo
20:37.30stoffell[av]bani, when that's not an issue..  (budget)
20:37.32SplasPoods/hand/handle
20:37.34mischko[av]bani, requirements as in features? i.e. call waiting or forwarding?
20:37.42_Sam--Juggie :  are you making fun of [av]bani's php provisioner? :)
20:37.44Renacorso to send a call to vm after it's been sitting in a queue for a 2 minutes, I would have to put timeout=2 in the queues.conf for that queue and then a priority after the call is answered in the extensions.conf for that queue?
20:37.57[av]banimischko: # of lines, touchscreen, backlight, speakerphone, etc
20:37.57Juggieno its a good thing
20:37.58Qwell[]more like 120
20:38.06Juggiewho wants to have a million flat files if one php script can generate them all
20:38.07SpaceBasshey folks
20:38.10mikefooits seconds not minutes
20:38.14ooglehello SpaceBass
20:38.14SpaceBassneed some assistance configuring Voxbone as a trunk
20:38.15Juggiethat being said, i have a php tftp server too someone wrote.
20:38.20[av]baniJuggie: i have a php script to do that for snom/sipura/grandstream
20:38.29ooglepeople write the wackiest stuff in php
20:38.31[av]baniJuggie: and i will be adding polycom when i get time
20:38.34*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
20:38.43Juggieav, the one i have is was written by nuxi
20:38.52[av]baniJuggie: i wrote mine myself
20:38.54Juggieyou wrote a tftp php server?
20:38.59SplasPoodJuggie: I'd actually be interested in seeing that tftp server..
20:39.02harryvvskype was just discussed on cnn right now. Seems skype calls are 128 bit encryption.
20:39.07[av]banino, i wrote a php autoprovisioner for snom/sipura/grandstream
20:39.09austinnichols101the aastra is pretty good for a $125-$150 phone
20:39.24stoffell[av]bani; a phone that meets 'BLF/MWI/distinctive ring through SIP header' and min. 2 lines, what is a good one?
20:39.24austinnichols1019112 or 9133
20:39.32Renacoranybody got a good example on how to send a call to vm after it's been sitting in a queue for 2 min?
20:39.32harryvvaastra are nice looking design
20:39.36Qwell[]stoffell: cisco
20:39.41[av]banitake a fresh out of the box factory reset phone, plug it in and it autoconfigures totally, including creating sip extensions for astersik and reloading them
20:39.50austinnichols101harryw: and they're heavy enough that you can club an intruder with the handset
20:39.55[av]banistoffell: snom or cisco
20:39.56harryvvRenacor u running a call center?
20:40.02stoffellQwell, the 7970?
20:40.12Juggiehttp://eder.us/projects/ <- php tftp server
20:40.13Qwell[]stoffell: 7940 and 7960 also support those
20:40.16stoffell[av]bani, snom much better then polycom
20:40.18Qwell[]I don't know about the rest
20:40.19Renacorharryvv: yep
20:40.20[av]banii dont use tftp at all
20:40.25stoffellok Qwell, tnx
20:40.30harryvvaustinnichols101 :)
20:40.31[av]banistoffell: for blf/mwi/distinctive ring, yes..
20:40.45stoffellok, tnx very much..
20:40.46jontowrenacor; set the maximum in-queue time in queues.conf to 2mins
20:40.58Juggieits a shame res_php was never finished
20:41.02SplasPoodJuggie: danke
20:41.12SplasPoodso no one here does least cost routing?
20:41.34[av]baniSuitcaseAvail()
20:41.44Mavviefor zap-only interfaces of course.
20:41.54SplasPoodstoffell: Polycom.
20:41.58mikefooif I wanted to connect two parties, as in one person calls in to a dial plan, hits a prompts, calls another, I want to do this then disconnect myself from the loop, this possible?
20:42.12[av]baniSplasPood: stoffell might not like the blf issue
20:42.27stoffellSplasPood, do they also support distinctive ring via sip-header?
20:42.41stoffelli love issues ;)
20:42.42SplasPoodstoffell: yes
20:42.54SplasPood[av]bani: what issue in particular?
20:42.55[av]banistoffell: snom supports arbitrary wav URL in the sip ring header. so you can have litterally millions of distinctive rings
20:43.10[av]baniSplasPood: max 7 buddy limit. kind of takes the piss out of the sidecars :)
20:43.13stoffellnice
20:43.28SplasPood[av]: Limit?  There isn't any limit that I've encountered...
20:43.36[av]baniSplasPood: you dont have a sidecar then
20:43.39Renacorjontow: how do you set that?
20:43.40SplasPoodIP601 + Attendent Console
20:43.51jontowlook at the example queues.conf
20:43.56Renacorjontow: thanks
20:43.58jontowyou can either do it that way, or in the Queue() application
20:43.59[av]baniSplasPood: you can only blf 7 extensions max on polycom, its a hard firmware limit.
20:44.08websaeany wholesale providers in here?
20:44.09[av]baniSplasPood: its a limit a _lot_ of people complain about
20:44.25SplasPood[av]: Wait I'm confused..   They sell an add-on for the 601 that DEF has more than 7 entries...
20:44.26Juggielook at the mitel 5235
20:44.40[av]baniSplasPood: its done via a different mechanism than * uses
20:44.56Juggiehttp://www.mitel.com/DocController?documentId=14886&c=9512&sc=9517
20:45.00[av]baniSplasPood: snom, cisco, everyone else uses the mechanism * currently supports. polycom does not.
20:45.14SplasPood[av]bani: hints ?
20:45.15_Sam--there was a just thing in the asterisk-users about it...bani is right
20:45.23_Sam--it makes the sidecar thing basically useless
20:45.27[av]baniSplasPood: its partly polycom's fault and partly * doesnt support the sip extension polycom wants
20:45.29Juggiehttp://www.mitel.com/resources/5235SilverBezelforPCv24.swf
20:45.33*** join/#asterisk DrData (n=michael@p54B24BCE.dip.t-dialin.net)
20:45.42[av]baniSplasPood: but theres no good reason for polycom to arbitrarily limit buddy watch to 7
20:45.53MstlyHrmls[av]bani: dumb question, what mechanism does everyone else use with * for blf?
20:45.58[av]baniSplasPood: because it limits max blf with current * to 7
20:45.59SplasPood[av]bani: why isn't this on the wiki?
20:46.03DrDatais it possible to register asterisk AND a softphone to e.g. sipgate?
20:46.06[av]baniSplasPood: it is afaik
20:46.18Qwell[]DrData: No.  Register the softphone to *, then * to the itsp
20:46.19JuggieQwell, http://www.mitel.com/resources/5235SilverBezelforPCv24.swf
20:46.23[av]baniSplasPood: go ahead and try to blf > 7 lines with ip601 and *. you cant do it.
20:46.25stoffellto me it's new and interesting info :)
20:46.31_Sam--DrData :  you cant have two devices register with the same credentials to the same place
20:46.43SplasPood[av]: don't have one handy to test with.. I'd be all over it if I did.
20:46.52Qwell[]_Sam--: You can
20:46.53remissw00t, w00t, w00t :D
20:46.58_Sam--well you can...but only one will work
20:46.58remissi got it working :)
20:47.02SplasPood[av]bani: can you help me find documentation about this?
20:47.02[av]baniSplasPood: its a common complaint htough, you can find threads on voxilla, asterisk-users and other places about it
20:47.07DrData_Sam--,  credentials == user/pw?
20:47.11[av]baniSplasPood: about as retarded as polycom's firmware policy
20:47.28_Sam--DrData :  i will let the real experts help you
20:47.30_Sam--i am just a novice
20:47.48austinnichols101[av]bani: limits *should* be zero, exactly one or infinite.  7 is a stupid limit (should be infinite)
20:47.55SplasPood[av]bani: I'm having a lot of trouble finding any mention of this limitation
20:48.09[av]baniaustinnichols101: "should" and "polycom" ... heh
20:48.27Qwell[]Juggie: Not loading
20:48.40austinnichols101[av]bani: that's really one of my pet peeves in programming...
20:48.45[av]baniSplasPood: http://lists.digium.com/pipermail/asterisk-users/2006-February/146983.html
20:48.46austinnichols101pissed me off
20:49.07[av]baniaustinnichols101: blame management
20:49.22MstlyHrmlsaustinnichols101: the problem is "infinite" and embedded systems don't play well together. I agree that 7 is a stupid limitation, but "infinite" is unreasonable as well
20:49.26mischko[av]bani, we're using these Panasonic cordless units now.  We don't use the answering service.  Something similar would be great. http://www2.panasonic.com/webapp/wcs/stores/servlet/vModelDetail?storeId=15001&catalogId=13401&itemId=62849&cacheProgram=11002&cachePartner=7000000000000005702&surfModel=KX-TG2740S&catGroupId=25041&surfCategory=Expandable%20Systems&displayTab=O
20:49.28austinnichols101I found it (Zero-One-Infinity Rule): http://www.catb.org/jargon/html/Z/Zero-One-Infinity-Rule.html
20:50.00[av]baniMstlyHrmls: "as much as the hardware will support"
20:50.11[av]baniMstlyHrmls: usually its more work to put _in_ a limit, than to leave it open
20:50.27MstlyHrmls[av]bani: right, which should be more than 7 but less than infinity :-)
20:50.35stoffellhm, I can live with limit of 7, if one needs more, one can use snom.. but blf/mwi AND dist.ring in a polycom, sounds great
20:50.37*** join/#asterisk xyklopz (n=xyklopzi@216-91-89-21.biltmorecomm.com)
20:50.45SpaceBassanyone using voxbone?
20:51.07[av]banistoffell: configuring polycom is poo though
20:51.07austinnichols101MstlyHrmls: but the limit should be the hardware itself, not the program
20:51.09*** join/#asterisk kpettit (n=keith@69.15.174.114)
20:51.10SpaceBassI dont totally get the concept of the URL stuff...they want to forward my DID to an exten@myIP  ...
20:51.24Qwell[]austinnichols101: Would you rather it not work at all, or limit it to a reasonable number?
20:51.29[av]baniat least it is now, wait till i get my autoprovisioners working for polycom ;)
20:51.31Qwell[]"address space and memory permitting"
20:51.38SplasPood[av]bani: ugh.. I never tested to that many, I suppose...
20:51.48[av]baniSplasPood: kind of makes the sidecar pointless with *
20:51.52MstlyHrmlsaustinnichols101: yes, but you don't want non-Call features interfering with the phones primary purpose: to make calls
20:51.54austinnichols101Qwell: right - a.s and memory permitting
20:51.54stoffell[av]bani, I've heard. using GXP-2000, Cisco and Thomson ST2030 at the moment
20:52.09Qwell[]MstlyHrmls: indeed
20:52.14SplasPood[av]bani: what alternatives exist, if any?
20:52.21stoffelltoo bad aastra doesn't have distinctive ring support
20:52.36austinnichols101agreed: it's just who came up with the arbitrary limit of 7.  Why not 6 or 8...
20:52.44austinnichols101it's a lazy way to program things
20:52.51[av]baniSplasPood: snom, if you want >7 working blf. cisco.
20:53.12xyklopzHelix is afk
20:53.14Renacorhmm I don't see anything other than timeout value in the queues.conf to limit queue wait time but that doesn't do anything either
20:53.21SplasPood[av]bani: and the cisco solution is SIP?
20:53.32[av]baniSplasPood: depends on which model you buy
20:53.42stoffellSplasPood, you could also use chan_sccp
20:53.45SplasPood[av]bani: well whatever model is gonna let me show 20 lines :)
20:53.48MstlyHrmlsaustinnichols101: aye, 7 is dumb, and who knows why it was set at that.
20:53.55SplasPoodstoffell: is it really considered production ready?
20:53.58[av]baniSplasPood: 7940/7960 with sidecar then
20:54.29Qwell[]7940 with sidecar is silly
20:54.38Qwell[]in fact, I don't even know if it works
20:54.45stoffellSplasPood, depends, very active development, i'm using it with some Kirk phones, but I don't update if i don't need to :)
20:54.47[av]baniknowing cisco, probably not?
20:55.44SpaceBasscan someone help me with calling a URI? I have a service trying to call an exten@myip and I'm not sure what I need to do on my end
20:55.59[av]baniif the gxp exp. port is for sidecar, i dont see how grandstream expects the modules to stay attached
20:56.01harryvvcan anyone give me a voice quality rating comming from xo?
20:56.52stoffelldoes snom have licensing issues like cisco? (like, you pay for a license and stuff?)
20:56.57[av]banino
20:57.11stoffellok, tnx, cool
20:57.13[av]baniyou shouldnt have to pay for cisco licenses though
20:57.16SplasPoodok well this 7 limit /w polycom has depressed me greatly...   Anyone wanna cheer me up with some LCR info? :)
20:57.23[av]baniuse sccp out of the box
20:57.39stoffell[av]bani, ah, ok, you're right
20:59.02xyklopzcan someone help me with setting up a sipphone virtual #
20:59.23xyklopzi already have the sip peer setup
20:59.28xyklopzand it's registering through the firewall
20:59.42xyklopznow I want to purchase a virtual # for them to accept incoming
21:00.33De_Monxyklopz whats your question?
21:00.44De_Mons/ts/t's/
21:00.57De_Mon^_~
21:01.09xyklopzokay .. the # is 1404XXXXXXX
21:01.25xyklopzI assume they handle the routing
21:01.31xyklopzfrom PSTN to IP
21:01.50xyklopzthen it'll come on my sipphone # (my softphone account # 174XXXX..)
21:01.53De_Monxyklopz is the provider registered in asterisk?
21:02.03harryvvxo is on a roll. It is probebly going to reach over 2 billion minutes this first quarter.
21:02.14xyklopzthe peer is there
21:02.55[av]baniQwell[]: sccp can invite an rtp stream from a sip phone, or does it need to be bridged through * ?
21:04.02De_Monxyklopz ok, so someone calls that provider (any number) and it reads the config and is sent into the specified contex in extensions.conf
21:04.12[av]banihmm. the zoom fxo echo tail is 10 milliseconds... bizarre
21:04.19De_Monfrom there you told it to do whatever you wanted it to... what is your QUESTION?
21:04.54[av]baniDe_Mon: what is the airspeed of an unladen swallow?
21:05.05De_Mon[av]bani 13
21:05.06Igbothom_IIIafrican swallow?
21:05.10Nivex[av]bani: get out of my head!
21:05.41[av]baniNivex: but it was so spacious and empty
21:06.07Renacoranybody got a good example on how to send a call to vm after it's been sitting in a queue for 2 min?
21:06.32[av]baniRenacor: set queue timeout to 2 min, then have it fall to voicemail(bla)
21:06.36xyklopzanyone have a sipphone account?
21:06.42*** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
21:07.01mischkoIs Vonage proprietary or using SIP?
21:07.02Renacor[av]bani: the timeout in queues.conf right?
21:07.06De_MonIgbothom_III 13 too
21:07.07[av]baniRenacor: yes
21:07.18Qwell[]mischko: Both
21:07.26Renacor[av]bani: I tried that, and added another priority after the call was answered in the extensions.conf but it didn't work
21:07.31Qwell[]SIP with proprietary logins
21:07.32De_Monxyklopz probably most people in here...
21:07.44[av]baniRenacor: i think you need to queue(bla) with some parameters for it to timeout and fall through
21:07.48Renacor[av]bani: do you have to restart asterisk for that or can you just do a reload ?
21:07.52xyklopzI just want to see if it rings through properly
21:07.57[av]baniRenacor: reload app_queue.so
21:08.13De_Monxyklopz dial the number?
21:08.53xyklopz17476380060 if it's called from another sipphone account it's free
21:09.10xyklopzi want to make sure it's going through * correctly to my lan peer I told it to ring to
21:10.00De_MonIs sipphone some VoIP provider I've never heard of?
21:10.08xyklopzsipphone.com
21:10.14xyklopzincoming virtual #'s only $35/year
21:10.21xyklopzno fee on incoming and only 2c/min outgoing
21:10.24xyklopzpretty cheap
21:11.11mischkoI assume WiFi SIP phones are available?
21:11.16Qwell[]mischko: yep
21:11.19afrosheensorta
21:11.19De_Monoh.. in that case I retract my previous comment about most people having it
21:11.28xyklopz{{ £åügHîñg Øüt £öüÐ }} ...
21:11.32De_Monxyklopz if you can make outgoing calls you can call it yourself
21:11.34afrosheenow my eyes
21:11.35Qwell[]mischko: If you get me a cisco 7920, I'll make sure it works with chan_sccp ;)
21:12.18mischkoQwell[], I wish.  What's chan_sccp?
21:12.23mikefooI am sitting next to four cisco as5400's  hah
21:12.45Qwell[]mischko: sccp channel driver for *
21:13.54KranZanyone messed with SRTP?
21:14.45harryvvmischko have been around for a while
21:15.01mischkoharryvv, not around VOIP.
21:15.36[av]banimischko: all the wifi phones seem to suck
21:15.48mischko[av]bani, how come?
21:15.50*** join/#asterisk NovceGuru (n=asdf@oh-71-53-48-11.dhcp.sprint-hsd.net)
21:15.51Qwell[]7920 doesn't suck
21:16.07[av]baniQwell[]: someone with a 7920 was complaining the other day its crap like a rebranded chinese clone
21:16.13*** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt)
21:16.32Qwell[][av]bani: Just like the guy who said a 7960 was better than a 7970?
21:16.33Qwell[]mmhmm
21:16.37[av]banino
21:16.48[av]banisomeone in this channel was having problems with his 7920
21:16.50*** join/#asterisk justinu (n=justin@72.18.13.34)
21:16.57[av]banisaid it was rebranded chinese clone junk
21:16.58mikefooI have six 7960's on my desk, hah
21:16.59Qwell[]so immediately it sucks...yes, of course
21:17.04PrimerI have a 7920
21:17.06Primerworks fine
21:17.15Primerusing chan_sccp
21:17.30PakiPenguindo sangoma analog cards support fax correctly?
21:17.31Primerbut I got mine from a friend that works at cisco
21:17.41*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
21:17.54[av]baniPrimer: is it worth the $550+ cisco price tag?
21:18.05[TK]D-FenderHey, need a hand here... I've got a PRI thats "Status: Provisioned, Down, Active"
21:18.05Primerdamn, I didn't realize it cost that much
21:18.10[av]baniPrimer: yep
21:18.12Primerperhaps I should ebay it
21:18.16[av]baniheh
21:18.17Qwell[]I don't think they do...not from cisco
21:18.18[TK]D-FenderHow do I cycle it so I can try to re-initializee the link
21:18.28justinud-fender: restart asterisk?
21:18.48Primer[av]bani: well, the battery doesn't seem to last long, but as far as performance, it works fine
21:18.52justinuasterisk doesn't seem to understand the concept of placing things OOS/INS
21:19.04[TK]D-Fender:/
21:19.18Primer[av]bani: could be that this one is just old, it's pretty beat up
21:19.21*** join/#asterisk jeebusroxors (n=jeebus@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
21:21.00xyklopzokay, * isn't accepting the call from sipphone.com!
21:21.01xyklopz:-(
21:21.07xyklopzit's registering
21:21.09*** join/#asterisk r0d3nt (i=nobody@wsip-24-234-241-145.lv.lv.cox.net)
21:21.48KranZsip debug ip [ipofsipphone.com]
21:22.42iCEBrkrI should just go home
21:22.48iCEBrkrI can't think anymore
21:23.23[av]baniQwell[]: they are more expensive from cisco. 550+ is street. retail is $675.99
21:23.31[av]banihttp://www.voipsupply.com/product_info.php?products_id=57
21:23.31justinu4:20 anyways
21:23.32justinu:P
21:23.37*** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk)
21:24.18Qwell[]nope...sorry...you're wrong :p
21:24.26[av]banihttp://voipstore.atacomm.com/Shops/ViewItem.aspx/27934028032-29579092224.htm
21:24.40Qwell[]$406 + $49 + $29
21:24.47Qwell[]phone + battery + charger
21:25.04*** join/#asterisk fiber0pti (n=John@invinine.com)
21:25.08[av]banioh.. $406 phone with no batter and no charger. excellent
21:25.15[av]banii stand corrected
21:25.17Qwell[]< $500
21:25.22Qwell[]500 < 550
21:25.27[av]bani$406 phone you cant use
21:25.32[av]banisuper
21:25.36Qwell[]$406 + $49 + $29
21:25.38Qwell[]phone + battery + charger
21:25.43greendiseasehow do you reload extensions?
21:25.43Qwell[]That's still < 500
21:25.46[av]baniwhere from?
21:25.48malverian[work]extensions reload
21:25.50Qwell[]greendisease extensions reload
21:25.51Qwell[][av]bani cisco
21:25.57malverian[work]asterisk -r -x "extensions reload"
21:26.03[av]banicisco sells it for less than voipsupply retails it?
21:26.08Qwell[]umm...duh?
21:26.17greendiseasei figured
21:26.17Qwell[]How else is voipsupply going to make a profit?
21:26.19greendiseasethank Qwell
21:26.22Qwell[]YOu think they sell stuff below cost?
21:26.40iCEBrkrjustinu: exactly
21:26.59[av]banidealer cost is generally < retail
21:27.07[av]baniwhy else have resellers?
21:27.27Qwell[]ask voipsupply
21:27.31[av]baniif cisco will sell direct to end users for less than resellers... sort of makes resellers pointless
21:27.35Qwell[]They're the tools who add $150 onto everything
21:27.40[av]baniQwell[]: atacomm?
21:27.44Qwell[]$200
21:28.17trixterhttp://www.thevoipconnection.com doesnt seem to rape as badly, they also dont charge what is generally listed on their webpage, or at least I have never paid that much..  and I have always gotten a price below voipsupply
21:28.24Qwell[]how much is voipsupply selling 7970's for?  or even 7960's?
21:28.26xyklopzit's not calling!!!
21:28.29Abydos313i'm editing the Makefile to compile my ztdummy driver and the line the book says to look for is not there
21:28.31trixterplus mike is a good guy, quick and very responsive ...
21:28.46xyklopzokay, configure the softphone directly to sipphone.com; call the virtual #; softphone rings
21:28.51[av]baniQwell[]: 549 for 7970, 299 for 7960
21:29.04xyklopzconfigure asterisk as sipphone.com peer; call the virtual #; get sipphone.com's vmail system
21:29.21Qwell[]cisco just raised their prices on the 7970 it seems...nice
21:29.33[av]baniexcellent
21:29.35_Sam--whats this mean:
21:29.36_Sam--Feb 17 16:29:06 WARNING[21881]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x410015f0', 10 retries!
21:29.47mischko[av]bani, I just hit both the links above for the 7920 and they show $575???
21:29.51Qwell[]270 for the 7960
21:29.59[av]banimischko: yes
21:30.03Dr-Linuxhi guys
21:30.10mischkoI just called Cisco and list on that phone is 475 (no battery/charger).
21:30.12Dr-Linuxsam i'm happy today :P
21:30.18[av]banimischko: liar!
21:30.24_Sam--Dr-Linux :  happy day to you
21:30.24mischkobattery is 75 or 95 and charger is 45. LIST.
21:30.28Qwell[]mischko: retail :)
21:30.37mischko[av]bani, I JUST called Cisco.
21:30.39_Sam--im happy every day...or at least every day when my remote asterisk gateways work
21:31.20[av]banimischko: so what is Qwell[] babbling about $406+49+29 then?
21:31.29Qwell[][av]bani: direct from cisco
21:31.29mischkoI don't know where he got those numbers.
21:31.39[av]baniQwell[]: end users?
21:31.41Abydos313nm guess it would help to open the correct Makefile..heh
21:31.55mischko[av]bani, Cisco's list price is more than I'd expect to find it for on the street.
21:31.56_Sam--Dr-Linux :  did you fix your wake up?
21:32.13[av]banimischko: i guess he's quoting dealer cost, and berating voipsupply for selling dealer cost + markup
21:32.41mischko[av]bani, for a retailer to sell that far above list is completely out of line. They should be 20-30% below list.
21:33.14[av]banimischko: i cant seem to find anyone selling the 7920 for anything lower
21:33.25[av]banimost dont even sell the 7920...
21:33.26justinuiCEBrkr: it's only 1:36 here :(
21:33.43mischko[av]bani, You can get it direct from Cisco for less than it's on those pages.
21:33.55_Paulo_here 19:33h. Time to grab a six-pack.
21:34.04[av]banimischko: maybe that explains why so few resellers list it
21:34.22mischkoI'd call people who have Cisco product line and ask 'em for a quote.
21:35.07mikefooiCEBrkr: hows the campaings going?
21:35.11Dr-Linux_Sam--: no, everything is done, but the alarm file doesn't copy to /var/spood/asterisk/outgoing dir
21:35.24_Sam--check your permissions
21:35.33mischkoI've gotta run.  Thanks for all the help [av]bani , Qwell[] , etc.
21:35.36_Sam--if asterisk is not running as root, it may not have the right permissions to write there
21:36.03*** part/#asterisk mischko (n=Scott@cvo-cr1-200-242.peak.org)
21:36.24NovceGuruhow come when dialing out some of my numbers work and some say "all circuits are busy now" ?
21:37.55*** join/#asterisk Simon- (i=fictitio@80.193.211.68)
21:38.05Dr-Linux_Sam--: you are talking to me?
21:38.07*** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net)
21:38.10GerbilWrkhas anyone found a way to imporve QoS with Teliax?
21:38.30*** join/#asterisk cassio (n=cassio@c91133b9.rjo.virtua.com.br)
21:38.33_Sam--GerbilWrk :  avbani found a way....get your terminating calls from someone else
21:38.50GerbilWrk;[
21:38.54xyklopzI can't get incoming to dial two extensions
21:39.02iCEBrkrmikefoo: Working on quota management
21:39.05_Sam--xyklopz :  DIAL(sip/1&sip/2)
21:39.05cassiodoes anyone know any provisioning software for sipura and linksys?
21:39.11xyklopzexten => s,1,Dial(SIP/1002&SIP/1005,20,r) doesn't work
21:39.33_Sam--then it only doesnt work because it doesnt know how to reach sip/10002 or 1005 (whichever one isnt ringing)
21:39.37iCEBrkrmikefoo: It paces itself and obviously records the data.  But we have quotas to meet and need to quit dialing
21:39.40_Sam--one of them probably isnt registered to *
21:39.45xyklopzbut they can talk to each other
21:39.53iCEBrkr_Sam--: Tonights the night.  I'm gonna try to start the ol' dinosaur!
21:39.54xyklopzthe dialplan seems right
21:39.54_Sam--they need to register to receive incoming calls from *
21:39.56xyklopzpm me?
21:40.01_Sam--sorry i have work to do
21:40.02xyklopzthey are both in sip show peers
21:40.07_Sam--sip show registry
21:40.10mikefooany decent answering machine detection out there?
21:40.12_Sam--er
21:40.15_Sam--sip show peers will be fine too
21:40.18iCEBrkrmikefoo: app_machinedetect.c
21:40.18_Sam--if it shows their ip
21:40.24xyklopzwhich it does
21:40.28mikefooiCEBrkr: word.. thanks
21:40.28iCEBrkrmikefoo: google it
21:40.30xyklopzi reloaded the config
21:40.32SplasPoodhas there been ANY development as to SIP-B + Asterisk?
21:40.44mikefooiCEBrkr: yah i googled before, seen a few, but bad reviews on them all.
21:40.48_Sam--iCEBrkr:  good luck!
21:40.49[av]baniSplasPood: hasnt started afaik
21:40.51mikefooiCEBrkr: later
21:40.52iCEBrkr_Sam--: :)
21:41.00SplasPoodor anything that does SIP-B that comes without a gigantic invoice...
21:41.00[av]baniSplasPood: coming summer 2006 to an * near you
21:41.09SplasPood[av]bani: ya I saw the post :)
21:41.31SplasPoodNo bounties?  Anyone we could offer some cash to to get some movement on that earlier?
21:41.40*** join/#asterisk mischko (n=Scott@cvo-cr1-200-242.peak.org)
21:41.47mischko[av]bani, http://www.mtmnet.com/CP-7920-AP-K9_New.htm
21:41.50mischko$460
21:42.01cassiodoes anyone know any provisioning software for sipura and linksys?
21:42.38_Sam--cassio:  did you try the [AV]baniMAKER (TM)?
21:42.59_Sam--<sorry couldnt help it>
21:42.59justinubani: i just tested the tones
21:43.11justinutwo comments: the snom "congestion" tone, has the wrong cadence
21:43.20justinuand it sounds brighter, or harsher
21:44.17mischkoWhat's it mean on the Cisco 7920 " without User License"?
21:44.46justinumeans you can buy it, but it's illegal to use :P
21:44.59mischkoWhat do you need a license for?
21:45.02[av]banijustinu: yes, exactly
21:45.11[av]banijustinu: snom congestion is raspy and bizarre
21:45.24*** part/#asterisk Simon- (i=fictitio@80.193.211.68)
21:45.26justinubut i wouldn't have noticed it unless you pointed it out
21:45.33[av]banijustinu: compared dialtone too?
21:45.45[av]banisame thing for me, snom dialtone is raspy and brighter
21:45.46NovceGuruhttp://pastebin.com/560231 I assume that problem isn't on my end?
21:45.46justinuno, i'll try that...
21:46.11[av]banijustinu: while playtones(dial) is smooth
21:46.19xyklopzokay, incoming sip calls aren't being displayed in the debug output
21:46.27xyklopzif i register with a softphone everything works fine
21:46.27[TK]D-FenderSplasPood : SIP-B for * 1.4
21:46.41justinubani: yeah, the difference in dial tone is more apparent
21:46.42xyklopzsomething isn't right with asterisk registering to sipphone.com as a peer
21:46.54[av]banijustinu: k, just making sure its not just me or "user error" :)
21:46.59justinuseems not
21:47.06[av]banijustinu: i'd like to get recordings somehow, since snom seems incredulous
21:47.11justinuheh
21:47.14[av]banijustinu: "you are mistaken sir"
21:47.21justinui wonder if they're sampled in the rom, or if it's a tone generator function
21:47.23[av]banijustinu: "our tones are correct, please put down your bong"
21:47.35De_MonAbydos313 are you going to use meetme with ztdummy?
21:48.00[av]banijustinu: dunno, would be silly to be sampled though.. would make it a pita to support arbitrary zones
21:48.08NovceGuruanybody?
21:48.10justinuyeah, but c'mon... it's snom
21:48.13[av]bani:)
21:48.39[av]baninice hardware, shame about the software
21:48.50justinusnom is also the only phone i have that doesn't support remote-party-id in the 180/183 responses
21:48.51jeebusroxorscan anyone recomend a cheap-o phone? its for home use....
21:49.02justinujeebusroxors: grandstream gxp2000
21:49.04justinu$85
21:49.20mischkoWhat do these "phone technology licenses" cost?
21:49.37jeebusroxorsjustinu; thanks alot - any chance of picking one of those up localy? best buy etc
21:49.38[av]banijustinu: remote party id?
21:49.44justinujeebusroxors: nope
21:49.50jeebusroxorsjustinu; figures :)
21:49.59justinubani: rpid is how asterisk can tell the callee who he's calling
21:50.04justinus/callee/caller/
21:50.05[av]banijeebusroxors: nobody sells sip phones retail, except maybe fry's
21:50.18justinui looked at fry's last time I was there
21:50.21justinuthey didn't have any
21:50.45*** join/#asterisk ambriento (n=ambrient@www.cobranet.com.br)
21:50.45jeebusroxorsthat sucks....are other VOIP phones more costly?
21:51.08justinuyeah, there is nothing at the gxp2000's pricepoint
21:51.12justinuwith the kind of features it has
21:51.19[av]banigxp2k is best price/features at that price point
21:51.20justinuyou can step down to an ATA, or budgetone 101
21:51.21SplasPood[TK]D-Fender: I know, I was wondering if some cash given to the right people might speed it up :)
21:51.34justinuyou can step up to polycom, or cisco
21:51.43[av]baniit's not a great phone, but it's great value for the money
21:51.44justinuor snom ;)
21:51.58justinuis there an RFC for SIP-B?
21:51.59*** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
21:52.01[av]banifor a home phone for playing with *, totally acceptable
21:52.30justinuone of my beta IP centrex customers loves his
21:52.37[av]banijeebusroxors: actually walmart sells a crappy soyo voip phone. they dont stock it though, you need to order it on their website to be delivered to your local store
21:52.39*** join/#asterisk Dr-Linux (n=Nothing@host202-147-168-130.lhr.dancom.net.pk)
21:52.40justinumy PRI customers hate them
21:52.49jeebusroxors[av]bani; thats cool. eventually id like to have my family hooked up on *
21:52.55harryvvhate what justinu
21:53.00justinuthe gxp2000
21:53.05[av]banijeebusroxors: but if youre going to order from the web, might as well order a gxp2000
21:53.16harryvvohh
21:53.27harryvvwhy poor quality?
21:53.27jeebusroxors[av]bani; yea ive actually seen those hanging around....
21:53.28[av]banijeebusroxors: you could spend a little more and get a polycom 301 though
21:53.59jeebusroxorsthe budget 101 is actually more than the gxp2000 heh
21:54.07justinuharryvv: poor interoperability with the PRI, i suppose
21:54.10[TK]D-FenderSplasPood : Sorry.. wish I could as I'm a Polycom user who'd benifit from it, but I'm not a coder....
21:54.11PakiPenguinanyone here is from the chicago area?
21:54.13harryvvi see
21:54.16justinujeebusroxors: budgetone 101 should be about $50
21:54.28PakiPenguini'd like to ask about any t1 provider ( voice + data both )
21:54.29jeebusroxorsi saw it at 89 heh
21:54.31afrosheenI've seen some junk sip phones at Fry's before
21:54.37[av]baniSplasPood still upset over the 7 buddy revelation?
21:54.39afrosheenAsus or someone was selling them
21:54.39[av]bani:)
21:54.42harryvvpolycom should make vidio security cat5 based and wifi cameras :)
21:54.46[av]banii just ruined SplasPood's week :)
21:54.57jeebusroxorsand the soyo is 90 bucks *rolls eyes*
21:55.03mischko[av]bani, Those licenses are about $150 apparently so the prices cited earlier may be good enough if they include a license.
21:55.10jeebusroxorslooks like gxp2000 it is
21:55.10[TK]D-Fender[av]bani : Well Cisco's 7914 only works in CCP, so that leaves SNOM and their flakeyness for presence :)
21:55.12afrosheenjeebusroxors: yeah that's the one, the Soyo
21:55.15[av]banijeebusroxors: yeah, like i said nothing comes close to the gxp2k in terms of price/features
21:55.22SplasPood[av]bani: oh yea totally.     I'm a total idiot for never trying more than 7
21:55.35jeebusroxorswell thanks alot for the advice
21:55.37[av]baniSplasPood: happy to help :)
21:55.39SplasPood[av]bani: And we were totally hoping to standardize on pcom
21:55.44Qwell[]mischko: $98 for most licenses, actually.
21:55.46JCC_the sipura 3000 ata at $99 worked out well for me as a home system, I keep the local # and use VoIP for LD
21:55.54afrosheenmight as well go with a decent phone and an iAXy if you're gonna be cheap
21:56.01JCC_and keep the portable phone
21:56.08[av]banimischko: i would imagine they do
21:56.14justinujcc: you having luck with the FXO side of it?
21:56.17afrosheenI did that with my portable phone at home
21:56.28mischkoAre these licenses a Cisco-only thing or something else?
21:56.30[av]banimischko: i checked around, nobody seems to sell it cheaper. if they list a price thats cheaper it always ends up being without battery or PS
21:56.38Qwell[]you need a license to use it with cisco
21:56.43JCC_yes, a little awkward at first, bt there are resources on the www
21:56.53justinuSplasPood: is there an RFC for sip-b?
21:56.55_Sam--anyone using any wireless headsets that can answer/hangup calls from a softphone?
21:57.12[av]banimischko: so the voipsupply and atacomm prices do not seem to be out of line with anyone else
21:57.26SplasPoodjustinu: yes I think so... I believe it's linked to on voip-info.org
21:57.38mischko[av]bani, seems like it, if the include the magic license.  Is this some sort of Cisco price gouge?
21:57.42mischko(the license)
21:57.47[av]bani_Sam--: maybe a bluetooth headset?
21:57.51Qwell[]voipsupply doesn't include licenses
21:57.56JCC_not sure what you mean, but I tried a bluetooth unit with Xten and didn't care for it, too many drops
21:58.00_Sam--i dont think they will answer/hangup the calls remotely
21:58.02[av]banimischko: cisco loves licensing.. and smartnet
21:58.06Qwell[]except maybe on the 7920
21:58.08_Sam--i have a wirless usb plantronics one
21:58.19_Sam--but it only answer/hangs up like 2 softphones remotely
21:58.33justinusearching for SIP-B rfc comes up with nothing
21:58.36_Sam--guess i have to buy eyebeam
21:58.37mischko[av]bani, so you're paying an extra $100 for a cisco phone.  That's nuts.  What do they offer that makes it worth it to go there?
21:58.38*** join/#asterisk darby_t (i=darby_t@dkt82.neoplus.adsl.tpnet.pl)
21:58.48justinumischko: name recognition
21:58.58Qwell[]mischko: If you buy it from voipsupply...you still need a license...
21:58.59[av]banijeebusroxors: its proably an optional annex to the main sip rfc
21:59.04justinuand a phone that people say works
21:59.04JCC_and good reliability, but expensive
21:59.05Qwell[]to use it with ccm anyhow
21:59.05Dr-Linuxjustinu: heyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy :P
21:59.10[av]banis/jeebusroxors/justinu/
21:59.13justinuDr-Linux: :P
21:59.14[TK]D-FenderSplasPood : I'd still go ahead if you can livewithout the full-service receptionist for a while
21:59.24justinubani: i haven't been able to find it yet, but i'd like to read it.
21:59.28mischkoQwell[], so if you use it standalone, you don't need a license?
21:59.35*** join/#asterisk Egonis (n=chultay@CPE000255fa0fde-CM00407b87dc7b.cpe.net.cable.rogers.com)
21:59.36Qwell[]as far as I'm aware
21:59.44mischkowhat's ccm provide you?
21:59.47EgonisI can't get device nodes to show for my sangoma card, although wancfg finds the card, can anyone help me/
21:59.52justinuccm is a softswitch
21:59.59Qwell[]mischko: a big hole where your bank account used to be?
22:00.02justinusimilar in functionality to asterisk, i suppose
22:00.04mischkolol
22:00.09justinui heard it runs on windows tho
22:00.12Qwell[]it does
22:00.13justinuwhich is frightening
22:00.15mischkoyuk.
22:00.32Qwell[]its funny though...even cisco direct, they won't sell you a 7920 without a UL
22:00.34mischkoSo you can use the 7920 with * just fine and it won't need the nasty license.
22:00.42Qwell[]if they did...you'd be able to get it for $308
22:00.43justinuyou still need a SIP license
22:00.48Qwell[]nah
22:00.48justinuno?
22:00.51Qwell[]keep it with sccp :p
22:00.54Primer7920 is sccp
22:00.57justinuso it's legal to buy and use with SCCP?
22:00.58Qwell[]get an $8 smartnet
22:01.03Primerthere is no official sip firware for it
22:01.04Primerafaik
22:01.05Qwell[]as far as I'm aware, heh
22:01.10mischkoSIP devices require a license? Is that some Federal tax?
22:01.14Qwell[]I think you only need a license to use it with CCM
22:01.16justinuno, it's a cisco tax
22:01.29SplasPood[TK]D-Fender: We will...   Although a receptionist console of some sort will be a requirement for some customers..
22:01.38JCC_cisco gives away nothing...
22:01.44[av]baniexcept grief
22:01.47Qwell[]So...if you can get cisco to sell you a spare 7920...you could get away with < $400
22:01.54mischkoJCC_, I'd say they charge you for nothing.  The license is nothing.
22:01.59justinu"you mean you want software for you phone? that's another 100 bucks..."
22:02.03afrosheenlol
22:02.09Qwell[]cp-7920-fc=  would be the product number, if it existed
22:02.13[TK]D-FenderSplasPood : You a reseller?
22:02.23[av]banijustinu: welcome to 1994? cisco has done that since forever
22:02.34justinuyep, xyxel was always like that too
22:02.38mischkosomeone said to stay away from Grandstream. How come?
22:02.48Hmmhesaysanyone in here ever lived in argentina?
22:02.50justinu"you bought a terminal server off ebay? that's just the hardware... the firmware image will cost you $800"
22:02.50[av]banimischko: because grandstream are cheap, like sipura used to be
22:02.57*** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr)
22:02.59SplasPood[TK]D-Fender: Well we're pushing a voip service offering.. as part of it we supply the phones, yea
22:03.02jhiver'nite all
22:03.05[TK]D-Fendermischko : Because they are cheap flimsy phones with flakey firmware....
22:03.13Qwell[]oh, wait...
22:03.19JCC_sipura used to be cheap... until cisco bought them
22:03.21Qwell[]cp-7920-fc-k9=
22:03.21[av]banias opposed to well constructed phones with flakey firmware...
22:03.26[TK]D-FenderSplasPood : Ok, well there are PC based optiosn if they can live with it for a bit...\
22:03.29*** join/#asterisk FuriousGeorge (n=Brian@ool-44c5a9b8.dyn.optonline.net)
22:03.37SplasPood[TK]D-Fender: like FOP?
22:03.42[TK]D-FenderSipura was always AFFORDABLE, not "cheap".
22:03.44mischko[av]bani, what brand do you recommend?
22:03.47afrosheenwell I love our polycoms, the 38 out of 40 that don't reboot mysteriously
22:03.54[TK]D-FenderSplasPood : Yeah or any of the other AMI based options.
22:03.57JCC_better put, you're right
22:03.59[av]banimischko: depends on your requirements and budget
22:04.16EgonisHas anyone else had /dev problems with Sangoma Cards?
22:04.20*** join/#asterisk bhima (n=gf2e@i13pc168.ilkd.uni-karlsruhe.de)
22:04.23[av]bani[TK]D-Fender: the 841 was cheap
22:04.27jhivermischko, I like my Fritz!FonBox integrated DSL / VoIP device
22:04.37[av]bani[TK]D-Fender: which is why everyone crreamed their pants over hte 941
22:04.41jhivertalks SIP, no NAT issue, just connect some phones on it
22:04.43SplasPood[TK]D-Fender: Do you have a personal fav?
22:04.46[TK]D-Fender[av]bani : Ok, barring that one :)  It's nearly the same boat as the GXP :)
22:05.15[av]bani[TK]D-Fender: spa3k is also cheap
22:05.21bhimaI'd like to make an asterisk call go through with audio in one direction only. Is there an easy way to do that?
22:05.24[av]bani[TK]D-Fender: and now, basically unsupported :/
22:05.40Qwell[]okay, they do exist.  $309 for a 7920, plus the battery/charger
22:05.46jhiverbhima, yes, use SIP + NAT :))))
22:05.53[av]baniQwell[]: cisco refurb?
22:05.53Qwell[]with no license
22:05.56[av]baniew
22:05.57Qwell[]brand new
22:06.01jhiverjust kidding really :)
22:06.02bhimajhiver: SIP and NAT were working for me. :P
22:06.05ManxPowerhttp://www.ashlux.com/?postid=14
22:06.22[av]baniQwell[]: cisco direct? i cant find any cisco resellers who will sell for that low
22:06.29[av]banineither can mischko
22:06.30Qwell[][av]bani yeah
22:06.40[av]banimakes you woinder why cisco bothers with resellers then
22:06.47[av]baniif they are just going to undercut their retail channel
22:06.56Qwell[]list is about $550
22:07.24mischkoQwell, list is $475 for part# CP-7920-AP-K9
22:07.38[av]banithats with license right?
22:07.41[av]bani-ap-
22:07.42Qwell[]mischko: change AP to FC...that'll work in the US
22:07.44*** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox)
22:07.50Qwell[]no, with the license is -ch1
22:07.53PakiPenguini  was wondering what sort of t1 i'd need to carry both voice and data if i go for a t1 from some provider and what card / hardware would i need ?
22:08.11mischkoQwell[], fellow at Cisco knew I was calling for US usage and gave me the wrong part #. Nice.
22:08.17Qwell[]heh
22:08.26Qwell[]I don't know what the difference is, but...pretty sure you want -fc
22:08.35Qwell[]fc=fcc
22:08.41Qwell[]must be fcc approved...dunno
22:08.50Qwell[]or follows fcc regs
22:08.57Qwell[]ap is for eu iirc
22:09.07mischkoOk. I'm off again.
22:09.22ketanpjhiver: does that fritz box provision over https?
22:09.49jhiverketanp, I don't think it does support provisioning
22:09.53jhiveryou have to configure it
22:10.02jhiverand I don't think you can lock it either
22:10.05ketanpjhiver: ok, thanks
22:10.07Qwell[]brb
22:10.11ketanpalways a catch... haha
22:10.18jhiverbut it's such a sweet device ):
22:10.19jhiver:)
22:11.40bhimaso, anybody know how to mute one side of a call?
22:11.45JCC_are they available in the US?
22:13.13[TK]D-FenderPakiPenguin : You'd just need a mixed mode T1 and any of the std T1 cards supported by *
22:13.39PakiPenguin[TK]D-Fender, i see , how'd i get data out of it?
22:14.10PakiPenguini mean the hardware needed for that
22:14.19PakiPenguinwould be provided by the teleco?
22:16.56*** join/#asterisk [hC] (i=turnerd@66.199.130.40)
22:17.07[TK]D-FenderPakiPenguin : the T1 card does the work...
22:17.15PakiPenguinawesome!
22:17.16[TK]D-Fenderbe it Digium or Sangoma
22:18.12ManxPowerI prefer doing the muxing/demuxing outside of Asterisk.  That way the internet and frame relay doesn't go down everytime I run ztcfg or reboot the Asterisk server, or stop Asterisk.  Expecially if you are SSHing into the system via that internet link
22:20.14MavvieASTERISK_FILE_VERSION(__FILE__, "$Revision$")
22:20.15MavvieWonder why this isn't expanded by SVN.
22:20.15PakiPenguinManxPower, what can i use to do that? i mean the hardware?
22:20.54ManxPowerPakiPenguin, Adtran TA 850 will do it, and there are several Adtran TSUs that will do it.
22:21.15ManxPowerLemme check a min.
22:21.23ManxPowernaw, I'm too lazy for that.
22:21.41*** join/#asterisk kpettit (n=keith@69.15.174.114)
22:21.50ManxPowerAdtran makes several boxes that will do it.  We got one that take a T-1, splits out channels to a DXS-1 port (to Asterisk) and V.35 port (to the Cisco router)
22:22.32*** join/#asterisk file[laptop] (n=jcolp@206.13.96.155)
22:22.48iCEBrkreh?
22:24.01ManxPowerTSU120 is what I think we have.
22:24.18PakiPenguini see
22:24.27*** join/#asterisk Libila (n=vye@ip68-8-174-154.sd.sd.cox.net)
22:24.30ManxPowercome to think of it a TA750 should be able to do it too, but I've not tried that.
22:25.43harryvvI wonder if its possible to force a extention to ask for a 1800 number and dial it on a usa bound termination point. lots of 1800 us bound numbers dont work up here in canada.
22:26.01*** join/#asterisk Tamarisk (n=adrian@user-6887.lns5-c11.dsl.pol.co.uk)
22:26.35*** join/#asterisk Ahrimanes (n=michael@aronsen.dk)
22:27.08ManxPowerharryvv, trivia
22:27.10ManxPowertrivial
22:28.10ManxPowerYou just have to make sure that users can't call USA toll free numbers that bill your phone number
22:28.16*** part/#asterisk JCC_ (n=john@207.41.92.131)
22:28.31LibilaI've got this chunk in my extensions.conf (http://rafb.net/paste/results/HEA11x69.html) and when I call a normal 7 digit phone number I hear all this cracking and buzzing, asterisk says:  -- Executing Dial("SIP/user1-dd98", "Zap/g1/1234567|20|t") in new stack, then says -- Called g1/1234567. So why do I not hear the number I called ringing? Then when I hang up the phone starts ringing and I answer it to hear the dial tone.
22:28.44ManxPoweri.e. "Call 1-800-BLOW-JOB, $2.99/min will be billed to your phone."
22:28.54bhimamanx: those exist?
22:29.23Eggplantbhima, oh yes they exist, and they are evil
22:29.36Qwell[]Libila: Are you calling through the right type of ports?
22:29.42EggplantAOL's 1800 dialup number does that
22:30.47LibilaQwell[]: I'm pretty new to all this. I think your talking about the FXO ports I have on my TDM04B? I have the first port hooked up to my normal telephone line.
22:30.57Qwell[]heh
22:31.07Qwell[]What is the phone plugged into?
22:31.19LibilaLAN
22:31.41Libilait's registered with asterisk, I can make normal extension calls to another phone.
22:32.13TamariskHi can someone tel me what libpri-1.2.2.tar.gz provides?
22:33.45ManxPowerTamarisk, it provides support for PRI lines
22:34.56robin_zmeep!
22:35.06TamariskManxPower. Thanks so for no ISDN cards I should not need it?
22:35.32ManxPowerTamarisk, I always install it anyway, but you should not need it unless you are using ISDN PRI
22:36.10TamariskOK just trying to figure why I have make errors and no package at the end.
22:36.34robin_zhey, watch out for this scam thats going around the supermarkets at the moment .. two girls offer to pack your bags and carry them in exchange for a lift to another nearby shop, once in your car one of them gives you a blow-job, afterwards you are so distracted, you fail to notice they steal your shopping whne they get out. A nasty trick indeed.
22:36.39TamariskI am just trying to install the asterisk 1.2.4 No Zaptel or any add-ons
22:36.42robin_zI fell for it last Tuesday
22:36.46robin_zwednesday
22:36.49robin_zfriday
22:36.53robin_zand twice on saturday
22:37.07bhimahttp://www.ftc.gov/bcp/conline/pubs/tmarkg/tollfree.htm
22:37.31mishehurobin_z: if they're hot and you bought some cheap ass shopping, it'd pay off.
22:37.42bhimaBilling you without a prior agreeement and a security method of some kind is illegal.
22:38.47mishehurobin_z: REALLY.  I didn't know that.
22:38.57bhimarobin_z: Given how much I think blow jobs probably cost, and how much my groceries cost, I'm not sure that's a bad scam.
22:39.38mishehurobin_z: but alas, if it was a joke and you need to label as a joke, it obviously wasn't a ha ha funny type of joke.
22:40.27robin_zsee!
22:40.39bhimayeah, I thought it was funny too.
22:41.07crusherThanks for the advice robin_z. Too bad I don't own a car....
22:41.35robin_zanyway ... ;)
22:42.04*** part/#asterisk ms345 (n=mike_sim@64.74.198.10)
22:42.13TamariskYou can always try a push bike and have one sit on the handlebars!
22:42.43robin_zI am reasonably certain I wold crash ...
22:43.30Tamariskstabilizers!
22:44.20mikefooHow would I be able to call into a pbx, then fwd to another party, and completely disconnect from the two callers, hence not be billed for their talk time. this possible?
22:44.29*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
22:44.50Qwell[]mikefoo: No
22:45.05mikefooahh any way that I can make that happen to some degree?
22:45.12crusherUnless your SIP provider offers free calls
22:45.13Qwell[]and not be billed?  no
22:45.21jhivermikefoo, if you're using SIP and reinvite it could be ok
22:45.28jhiverif you're using PSTN, no way
22:45.38jhiveryou'll have to pay for both ways
22:45.51Qwell[]SIP will still charge you
22:45.56jhiverguys have you tried out hamachi?
22:46.06mikefoomy pbx would be * on sip, with a providers doing the terminating for me.
22:46.23jhiverit looks like quite interesting because it's able to establish NAT <=> NAT VPN sessions and that would be pretty handy for VoIP
22:46.28mikefooso no way huh...
22:46.34Mavviehmmm... wonder which dialplan variable is holding the context on which the calls comes in.
22:46.44Qwell[]${CONTEXT} ?
22:47.10MavvieQwell[]: nope, that one holds the current include or macro.
22:47.23Mavvie<PROTECTED>
22:47.39Qwell[]${MACROCONTEXT}
22:47.54ManxPowerMavvie, SetVar(SAVED_CONTEXT=${CONTEXT})
22:48.17ManxPowerREADME.variables is your friend
22:49.10jhiverI wonder how they do it though
22:49.41NovceGuruhttp://pastebin.com/560231 I assume that problem isn't on my end?
22:51.05MavvieManxPower: yes, but then I have to add it to all numbers before they jump into the macros.
22:52.01ManxPowerMavvie, I'll bet MACRO_CONTEXT  would work, but again, README.variables is your frien
22:52.01ManxPowerd
22:54.05Dr-Linuxjustinu
22:54.09NovceGuruit gives me the "all circuits are busy now" error
22:54.28MavvieManxPower: that one still gives me the calling context, not the initial context.
22:55.12jhiverhttp://en.wikipedia.org/wiki/Hamachi
22:55.24Mavviethere is no such a thing yet.
22:55.44ManxPowerMavvie, Yup.  Just do this, make the context the call comes into very limited, i.e. an exten => line to match any call, do the setvar, then go a goto ${EXTEN} in the real context to do everything else.
22:56.22ManxPowerYou can pretty much to any processing before call routing doing it that way
22:56.55ManxPowerto == do
22:57.07Mavvieaha, that's a nice workaround. thanks.
22:57.07*** join/#asterisk rogier (n=rogier@16-65-dsl.ipact.nl)
22:57.37ManxPowerMav have you ever used variable "subscripts" in extensions.conf?
22:57.58MavvieI'm not familiar with the term.
22:58.34Mavvie(but that doesn't mean that I might not have used it)
22:59.39*** part/#asterisk austinnichols101 (n=austinni@70.46.69.130)
23:00.39ManxPowerMavvie, http://pastebin.ca/42002
23:00.59ManxPowerMavvie, they are NOT subscripts, but they look like them
23:01.56Dr-LinuxManxPower: whats path of README.variable ?
23:02.04Mavvieto me it looks like a macro, but my guess is that it's different.
23:02.33ManxPower<PROTECTED>
23:02.54Dr-Linuxhhm.. cool
23:02.54Mavvieyou call this with "include => auto-attentdent" ?
23:03.05ManxPowersorry, that's not a good paste for the example
23:04.30*** join/#asterisk tuxinator_linux (n=tuxinato@70-32-106-248.ontrca.adelphia.net)
23:04.36ManxPowerThis is much more complex: http://pastebin.ca/42003
23:04.38*** join/#asterisk santiago (n=santiago@63.245.86.219)
23:04.51justinuwhat does this message mean?
23:04.52justinuFeb 16 08:47:06 ERROR[21778] app_dial.c: Could not stop autoservice on calling channel
23:05.09*** part/#asterisk santiago (n=santiago@63.245.86.219)
23:05.40*** join/#asterisk TuckerAdelaide (n=TuckerAd@58.160.196.17)
23:05.46TuckerAdelaide#thebook
23:05.51Dr-Linuxjustinu: did you google it?
23:05.54justinuno
23:06.09*** join/#asterisk bkw_ (n=brian@ppp-70-128-113-60.dsl.tulsok.swbell.net)
23:06.12*** part/#asterisk bkw_ (n=brian@ppp-70-128-113-60.dsl.tulsok.swbell.net)
23:06.14*** join/#asterisk thazza (n=thazza@229.9.233.220.exetel.com.au)
23:06.16*** join/#asterisk bkw_ (n=brian@ppp-70-128-113-60.dsl.tulsok.swbell.net)
23:06.18Dr-Linuxlooks new to me
23:06.59thazza~thebook
23:07.01jbothmm... thebook is Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
23:07.38Winkieis that the ololly one?
23:08.19thazzaif you mean oriely.. Yep.
23:08.33Winkieexcellent i didn't know if it was easily available online or if i'd have to google
23:08.37Winkiei'm going to hardcopy it at some point
23:09.14*** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net)
23:09.35TuckerAdelaide~thebook
23:09.36jbot[thebook] Asterisk: The Future of Telephony, released under the Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
23:09.46tzafrir_laptopWhat is "the creative commons license"?
23:10.17Winkieit's a licensed designed by web 2.0 retards
23:10.18thazza~google
23:10.20jbotsomebody said google was a search engine found at http://www.google.com/
23:10.34Mavviehttp://en.wikipedia.org/wiki/Creative_Commons
23:10.41Mavviejust had it open for somebody else.
23:10.49thazzalol
23:10.50tzafrir_laptopThere are plenty of CC licenses. That specific one actually does not allow commercial distribution (e.g.: to include it in a CD you sell for the cost of burning)
23:11.19tzafrir_laptopNot that I don't consider the release of the bok under that license very generous and helpful
23:12.09tzafrir_laptophow do I edit a jbot item?
23:12.29MikeJ[Laptop]jbot: help?
23:12.31jboti guess help is /msg jbot help
23:12.39MikeJ[Laptop]heheh\
23:12.51*** part/#asterisk TuckerAdelaide (n=TuckerAd@58.160.196.17)
23:12.55MikeJ[Laptop]jbot tzafrir is a guy who needs help
23:12.56jbot...but tzafrir is already something else...
23:12.58tzafrir_laptopdidn't get me very far last time I tried it
23:13.10MikeJ[Laptop]jbot: tzafrir
23:13.11jbotextra, extra, read all about it, tzafrir is http://tzafrir.org.il/
23:13.26Dr-Linuxwhos owner of jbot ?
23:13.28tzafrir_laptop~help
23:13.29MikeJ[Laptop]jbot: no tzafrir needs help
23:13.52MikeJ[Laptop]jbot: tzafrir
23:13.53jbotfrom memory, tzafrir is http://tzafrir.org.il/
23:13.57MikeJ[Laptop]jbot: no tzafrir needs help
23:14.13MikeJ[Laptop]jbot: tzafrir
23:14.15jbotsomebody said tzafrir was http://tzafrir.org.il/
23:14.19MikeJ[Laptop]grrr
23:14.22MikeJ[Laptop]oh
23:14.23[av]banihttp://www.vonmag.com/images/issue/jul05/depts/tjo-avaya.jpg
23:14.26[av]banilooks like a nice phone
23:14.27MikeJ[Laptop]jbot: no tzafrir is needs help
23:14.29jbotMikeJ[Laptop]: okay
23:14.33MikeJ[Laptop]there we go
23:14.36MikeJ[Laptop]jbot: tzafrir
23:14.38jbotit has been said that tzafrir is needs help
23:14.43MikeJ[Laptop]:P
23:15.14FuriousGeorgeanyone using mog's asterisk-xmpp?
23:15.32*** part/#asterisk Cresl1n (n=matt@gateway.digium.com)
23:15.53Dr-Linuxanyone is using wake up call?
23:16.09tzafrir_laptopjbot, forget tzafrir
23:16.39tzafrir_laptopjbot, no, tzafrir is http://tzafrir.org.il/
23:16.41jbottzafrir_laptop: okay
23:16.47TamariskHelp with install from source file. I get: -  checking for tgetent in (various) Configure error termcap support not found
23:17.21TamariskI have termcap libs installed but can not see what provides tgetent?
23:17.29WinkieTamarisk: do you have termcap devel libs?
23:18.03Mavviencruses-devel.
23:18.04TamariskWinkie. I will check what I have in yast please hold!
23:18.22Mavviencurses-devel
23:19.20TamariskSorry is ncurses-devel what i need?
23:19.21*** part/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl)
23:19.27Mavvieyes
23:19.28harryvvyes
23:19.32tzafrir_laptop~thebook
23:19.34jbotit has been said that thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
23:19.37Mavviesee the README in the root directory.
23:19.40TamariskOK looking for that as well
23:19.47tzafrir_laptop(note the "a")
23:20.40tzafrir_laptopMavvie, you probably know that "ncurses" is "new curses".
23:21.09TamariskI have the termcap libary and termcap debug info installed
23:21.15tzafrir_laptopWhich probably comes natural with the build process
23:21.19TamariskI am now installing ncurses-deval
23:21.35Tamariskand debug info?
23:21.49tzafrir_laptopNot needed for building
23:21.50klasstekWhy does IAX -> SIP -> ZAP suck so much when SIP -> SIP -> ZAP works great?
23:22.21tzafrir_laptopklasstek, is there transcoding in the SIP>SIP?
23:22.29Tamariskthe debug info not needed?
23:22.36tzafrir_laptopno
23:22.53klasstekno transcoding but the media path is not released for other reasons
23:23.32TamariskOK may be a bit late to stop now but should not hinder.  I hope
23:25.19Tamariskgetting further then ever before
23:26.18Tamariskplenty of missing prototypes but still going
23:28.04Mavviehey... looks like the dial command has become smarter.
23:28.25Mavvieyou can now do more than one trunk there.
23:28.33*** join/#asterisk st3v (n=spj@netblock-66-218-41-231.dslextreme.com)
23:29.29TamariskBummer!  Cannot find -lssl collect2: id returned 1 exit status
23:30.10MavvieTamarisk: check the README, search for ncurses-devel and see what other libraries you need to install.
23:30.12justinubani: it's weird that the snom internal ring tone sounds just like the asterisk generated tones
23:30.19justinubut not the dial/busy/congestion
23:30.31TamariskOK
23:31.05tzafrir_laptopTamarisk, openssl-devel or something in the neibourhood
23:32.39*** join/#asterisk jijgeh (n=luken@static-66-182-95-76.bbsc.net)
23:33.01Tamarisk<PROTECTED>
23:34.28*** join/#asterisk ctooley (n=ctooley@c-67-187-102-122.hsd1.tx.comcast.net)
23:34.39ctooleyAnyone in the Dallas area looking for work?
23:35.54Tamariskmissing the openssl-deval had openssl but not the libs  thanks will now install and try again
23:36.19*** part/#asterisk p0g0 (n=p0g0@madwifi/support/p0g0)
23:36.24FuriousGeorgeis there any way to check the version of the zaptel module that is laoded
23:37.07tzafrir_laptopTamarisk, generally if there is one missing prototype you're missing some headers and usually that means a missing -devel package (or an rpm-based system)
23:37.23backblueklasstek: i have sip-iax-zap and it works great
23:37.36klasstekother way
23:37.41klasstekiax-sip-zap
23:37.47tzafrir_laptopklasstek, what version of *?
23:37.51klasstekzap is pri on t400p
23:37.51backbluei dont use iax phones
23:38.00backbluemy zap is bri
23:38.02klasstekhrmmm....  one moment
23:38.19klasstekiax-sip is stable 1.2.4
23:38.25backbluei even do sip-iax2-sip-zap
23:38.27TamariskOk thanks for that information I will try and file away in the gret matter.  Suse10 provides 1.09-4 all docs or articles relate to 1.2.4 hence trying to install from source
23:38.28backblueand it works great
23:38.52klassteksip-zap shows  CVS-v1-0-06/04/05-21:26:28
23:39.30klasstekiax-sip works fine and sounds great
23:39.39klassteksip-sip-zap works fine and sounds great
23:40.00klasstekiax-sip-zap sounds terrible to the zap side. sounds fine on the iax side
23:40.31*** part/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net)
23:40.52*** join/#asterisk saftsack (n=saftsack@p54A7FEB2.dip.t-dialin.net)
23:40.53saftsackhi
23:41.02*** join/#asterisk robbie2 (n=rob@howzat.dsl.onthenet.net)
23:41.09robbie2mornings
23:41.10Tamarisktzafrir_laptop.  just reading your comment through again when you say a rpm based system. Suse is I think is that an issue
23:41.53robbie2i have a TE110P, everything is working excellent, except people tell me there is a background crackling when i call them
23:42.05robbie2its connected to an ISDN10 in australia
23:42.10robbie2using asterisk 1.2.4
23:42.15tzafrir_laptopTamarisk, get the asterisk 1.09 srpm and make sure you have all the build dependencies installed
23:43.11TamariskI can and have installed 1.09 from Suse it does install. I then removed it all to start fresh with 1.2.4
23:43.17tzafrir_laptopThat is: when you run: rpmbuild -t   (is that the right flag for extracting only) it will not fail
23:43.41tzafrir_laptopThis is something that can be done by a normal user in a different dir.
23:44.14tzafrir_laptop(At least the equivalent in Debian is the recommended practice even for non-package builds)
23:44.45saftsackhi is somebody here how has a hfc chipset based isdn card?
23:45.28Tamariskrpmbuild, I can just about remember that from RH many years ago. Yast is the automatic installer in Suse
23:45.39*** part/#asterisk Utah_Dave (n=boucha@0-1pool138-37.nas28.salt-lake-city1.ut.us.da.qwest.net)
23:46.00ctooleyIf anyone's interested in the Dallas area job email me at ctooley@gmail.com
23:46.31TamariskIt manages the download off the web for me etc and make life very simple and I need it at this rate
23:51.49*** join/#asterisk anderiv (n=anderiv@207-67-87-34.gen.twtelecom.net)
23:52.57*** join/#asterisk kietlak (n=kietlak@11-mo3-6.acn.waw.pl)
23:54.20*** join/#asterisk ome3 (n=ome@69.90.135.67)
23:54.38ome3Anyone know of any cheap providers which would allow me to move my lines from BS to them?
23:55.05FuriousGeorgeanyone using bristuff?
23:55.41TamariskWhooop  Whooop!!
23:55.51TamariskAsterisk Installation complete
23:56.11ome3anyone here moved phone numbers to a provider? i juts asked in #nufone although nobody seems to be there
23:56.37FuriousGeorgei changed zaptel-1.2.3 to 1.2.4 in the bristuff install script, and eveyrthing went ok with the patching, but i got to wondering if it was a bad idea

Generated by irclog2html.pl by Jeff Waugh - find it at freshmeat.net! Modified by Tim Riker to work with blootbot logs, split per channel, etc.