IRC log for #asterisk on 20080523

00:00.13seanbright~weather KMTN
00:00.25jameswf-homeso long and thanks for all the fish
00:02.43*** join/#asterisk anthm (n=anthm@72.60.151.190)
00:03.25rob0~weather phnl
00:13.16friezewow, so the documentation is the polycom admin guide has no obvious relationship to the firmware files they distribute does it?
00:13.24friezein the admin guide rather
00:14.06mcab~weather cyvr
00:16.15mcabfrieze: if you're setting up a boot server for polycoms, the best thing to do is get the firmware zipfile from your reseller, and use the configs from there as a base
00:18.46*** join/#asterisk Test-TDM421BF (n=Test-TDM@62.120.56.11)
00:18.55Test-TDM421BFhi all
00:20.19Test-TDM421BFi have a problem with TDM421BF card, cant dial out!
00:23.19friezemcab: thanks. actually have to wait for them to authrize me for downloads though. However I realized I was looking at the SIP application part number, not the phone part number. grr I think I can make the polycom website download work now
00:30.59SteveTotaroany way to make megaco h.248 work with asterisk?
00:31.02SteveTotaro~pb
00:31.02jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
00:32.27SteveTotaroi am tryin mgcp changed the port for megaco http://pastebin.ca/1026521
00:38.24edibraci see these warnings in my logs "channel_find_locked: Avoided initial deadlock" -- yet i have heard no complaints .. anyone familiar with this problem? One page I googled, it said that to get a better clue,  enable certain flags to make logs more verbose
00:38.52SteveTotarothus freeswitch was born
00:39.01edibraccould the warning be an indication of a future problem or I'm wondering if it's something I can ignmore.
00:39.05*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-254d5948dcdb7855)
00:39.21SteveTotaroread why freeswitch was created
00:39.31SteveTotaroit was all about the deadlocks baby
00:45.03edibraci'm on 1.2.12 though - i figure newer versions will have addressed a lot of deadlock issues, as more people have submitted backtraces?
00:45.51edibraci guess there's no way I'll know what's up, until i turn on debugging, and figure out what's going on as indicated in: http://www.voip-info.org/wiki/view/Asterisk+debugging
00:57.11*** join/#asterisk moy (n=moyhu@189.169.69.205)
00:59.18eric2what's the easiest way to compare dates in the dial plan?
01:01.47*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
01:09.54edibracSteveTotaro: i was reading http://www.freeswitch.org/node/117 which explains a lot about deadlocks -- does this mean that given any asterisk installation, if you have enough traffic you will eventually hit a deadlock situation?
01:11.39seanbrightedibrac: ideally, no.
01:12.30seanbrightedibrac: i've been running 1.4 in production with 30-40 simultaneous calls for a few months with no deadlocks
01:12.39seanbrightedibrac: not exactly a high traffic situation, but...
01:13.11edibracwell that's a lot more than me -- do you still get the warning "channel.c: Avoided initial deadlock" ?
01:13.53seanbrightedibrac: no
01:13.54*** join/#asterisk existx (i=existx@that.orgasm.made.me.s.cre.am)
01:14.41edibracah wait, i guess i still have to turn on debugging to see what it might be. and i'm on a different version.
01:14.51seanbrightedibrac: 1.2?
01:14.56edibracor i can ignore this all and pretend i didn't see it
01:14.58edibracyeah
01:15.55seanbrighti've worked with 1.2 and 1.4, and i've had more lock contention problems with 1.2
01:16.16seanbright1.2 was in a higher volume environment, however.
01:16.28seanbrightbut they have since upgraded to the latest 1.4, and have no problems at all
01:17.21seanbrightand if you are seeing 'Avoided initial deadlock' you are in much better shape than if you were actually deadlocked
01:18.04edibracha - yeah. i'm just combing through the logs so i know what i'm up against
01:21.53*** join/#asterisk Frogzoo (n=Frogzoo@124.184.33.9)
01:24.06*** join/#asterisk km2 (n=x@adsl-76-252-245-25.dsl.pltn13.sbcglobal.net)
01:25.25*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
01:35.02*** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net)
01:44.17*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
01:44.18*** mode/#asterisk [+o russellb] by ChanServ
01:48.46*** join/#asterisk sumodat (n=fred@CPE0004dc0cb5b2-CM0014e8271e96.cpe.net.cable.rogers.com)
01:49.26sumodattest
01:50.21sumodatjoin #asterisknow
01:50.44*** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
01:50.44*** mode/#asterisk [+o russellb] by ChanServ
01:54.06sumodatcan anyone take a cisco question
01:54.14*** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net)
01:54.56Strom_Mi can take it as hard as you can give it
01:55.16sumodatalrighty then
01:55.30sumodatjust installed asterisknow (new to this) and I have a 1721
01:55.47sumodatcan't figure out how to get them talking
01:55.54sumodatjust looking for someone to point me in the right direction
01:56.07Strom_Mwhat protocol is the 1721 talking?
01:56.14sumodatsip I believe
01:56.27sumodatbrand new config
01:56.43SteveTotaroh.248 megaco
01:56.45sumodatso whatever I tell it
01:57.20sumodat1721 has 2-fxs card and 2-fxo card
01:58.07Strom_Moh, it's a 1721 router
01:58.12sumodatyessir
01:58.55*** join/#asterisk mihinomenest (n=argh@cerebus.clandestineresearch.com)
01:59.52Strom_Mgod, its been forever since i dicked with my 1720
02:00.06sumodatyah well, I'm taking the economical route
02:00.44sumodatfigured it would be a good way to get started
02:01.36sumodatwhere would I tell asterisk about connecting to the 1721
02:04.17*** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net)
02:04.28*** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com)
02:05.08[TK]D-Fendersumodat, sip.conf
02:05.30sumodatthx
02:11.43*** join/#asterisk fish-bulb (n=cstewart@216.207.245.1)
02:11.46*** join/#asterisk tapirkopi (n=denmasbo@202.154.57.15)
02:12.19tapirkopihello....
02:14.02tapirkopihi is there anyone have success load balancing with asterisk ?
02:14.53[TK]D-Fendertapirkopi, usually SER or something similar sits in front and handles that.
02:15.30SteveTotaroopenser
02:15.38tapirkopiyeah...i find the solution using oepnser with asterisk
02:15.51tapirkopisomething like this :
02:16.00SteveTotarohttp://www.jeremy-mcnamara.com/2007/03/28/how-to-configure-openser-sip-registar-sip-proxy-and-far-end-nat-traversal-for-media/
02:16.14SteveTotaroit will give you a basic setup
02:16.24tapirkopi[SIP_Client]---------------------------------------->[OpenSER]-----------------------[Asterisk]--------------------
02:16.40SteveTotarohttp://www.openser.org/docs/modules/1.2.x/dispatcher.html then check this for load balancing and failover
02:16.54tapirkopiyes steve
02:17.04tapirkopilike i say yesterday
02:17.15tapirkopii've got problem "loop detected"
02:17.39tapirkopiand then you suggest me to add "canreinvite=no"
02:17.45SteveTotarooh, canreinvite=no did not work
02:18.13tapirkopiyes did not work
02:18.33SteveTotaroit should
02:18.41SteveTotaromaybe openser is setup incorrectly
02:19.00tapirkopii've already sent my config to your email
02:19.07tapirkopiif you don't mind
02:19.13tapirkopiyou could look for it
02:19.20*** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com)
02:19.40SteveTotarothe link you posted was if a call came from pstn -------------> asterisk ------------------> OpenSER ------------------ back to asterisk -------------------> back to pstn
02:21.05tapirkopiactually i just want to forward the REGISTER / INVITE to asterisk
02:21.08SteveTotarowhat are .69 and .70?
02:21.28SteveTotaro.70 is openser
02:21.32*** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net)
02:21.33tapirkopiand if this success, i'll try to balance the call to other asterisk server
02:22.06tapirkopi70 - openser
02:22.11tapirkopi69 - asterisk
02:23.25SteveTotarorewritehostport, are you sure that is right?
02:24.28*** join/#asterisk blinky42 (n=steveb@c-71-230-47-244.hsd1.pa.comcast.net)
02:25.39SteveTotaromaybe you should pb your config taking out the IPs so someone can help, i can look at it tomorrow but it is late here
02:26.28tapirkopii'm sory
02:26.52tapirkopiit is morning here
02:28.31tapirkopi<PROTECTED>
02:28.51tapirkopiopenser : 192.168.1.70
02:29.09tapirkopiasterisk : 192.168.1.69
02:29.18SteveTotarook
02:29.45SteveTotarouse_media_proxy();
02:30.01tapirkopii'll try to load balance VoIP cal, register / invite message
02:30.02SteveTotaroput that between rewrite and route(1);
02:30.17tapirkopibut first, i must be able to forward the call to asterisk server
02:31.10tapirkopiok so i must use media proxy
02:31.49SteveTotaro~pb
02:31.49jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
02:32.27tapirkopiok
02:32.51SteveTotarohttp://pastebin.com/m7893052a
02:33.54SteveTotarolines 14-19 i think you need to look at
02:35.52tapirkopithis is my conf
02:35.54tapirkopihttp://pastebin.com/m258756fd
02:37.48SteveTotarodo you see where use_media_proxy(); is used in the example, you don't have that
02:38.06*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
02:38.42tapirkopiok i see that
02:38.50tapirkopii'll try that
02:39.27SteveTotaroand the example has return too.  try to copy the examply closely
02:41.46*** join/#asterisk docelmo (n=vircuser@h59.77.75.24.cable.rstb.cablerocket.net)
02:42.15docelmoanyone know why asterisk would not pass a "ringing" when the 'r' flag is used in the dial command?
02:43.33SteveTotarodoes it pass ringing without the r flag?
02:43.39docelmono
02:43.51docelmokinda funky..  Its a Dell Dual Xeon 500
02:43.53SteveTotaroDAHDI?
02:43.57docelmoI just cant remember the model
02:43.59docelmodahdi?
02:44.04SteveTotarozaptel?
02:44.20docelmoits TDM->SIP->Asterisk
02:44.32SteveTotaroso which direction doesn't get ringing?
02:44.37docelmothen from Asterisk->Polycom
02:44.41docelmocoming to the polycom
02:44.45docelmofrom the pstn
02:45.03docelmoits weird..  I have never seen this before in my houndreds of installations of asterisk
02:45.14docelmoIm thinking its either Ubuntu 8.04 or the Dell
02:45.22docelmoIm also running Ztdummy
02:45.38SteveTotaroso what is your pstn connectivity?
02:45.42docelmoPRI
02:45.44SteveTotarowhy ztdummy?
02:45.49docelmonot TDM card
02:45.49SteveTotarowhat card?
02:46.04docelmoPRI via Interaction SIP Gateway
02:46.10SteveTotaroso a bogus pri
02:46.24docelmoI mean asterisk should at least be able to put the ring on the line and it cant do that
02:47.32docelmoI also tried injecting music on hold with the 'm' flag and it doesnt do that
02:48.30SteveTotarosip.conf
02:48.32SteveTotaro~pb
02:48.33jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
02:48.42SteveTotaroprogressinband=no
02:49.27SteveTotarohttp://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband
02:49.32SteveTotaroplay with that a bit
02:50.49docelmoI have it set to yes right now but it doesnt seem to matter.  I will change to no and see what happens
02:51.37SteveTotarohave you tried no already?
02:51.45docelmoThanks steve..   Im under the honest belief that its either the Dell or Ubuntu..   Im gonna install CentOS on it and see what happens..  if same issue time to break down and buy a box..
02:51.46docelmono
02:52.02docelmoI tried it w/o the directive which I believe defualts to no
02:52.13*** join/#asterisk BeeBuu (n=beebuu@59.38.96.217)
02:52.52BeeBuuhow can i get the num without the last digit?
02:53.04docelmocan you be more vague?
02:53.22BeeBuu${EXTEN:0:-1}?
02:53.50glazWhat number? the one calling you?
02:54.01glazthere's plenty of number in asterisk.
02:54.16BeeBuuyes,like 1234 calling me,i just want get the number 123
02:54.48glaz(${CALLERID(num):3)
02:55.36xacatecasloves asterisk.
02:55.43xacatecasnow for some sleep
02:55.44SteveTotaro<PROTECTED>
02:55.46docelmoI do when it does what I cant it to
02:57.23BeeBuuSteveTotaro: how about the callerid num longer than 3?
02:57.35SteveTotarodocelmo, funnit thing i was about to ask you if you could be more vague?
02:57.41SteveTotaroa pri is not sip
02:57.44*** join/#asterisk moy (n=moyhu@189.169.69.205)
02:57.51SteveTotaroyou are just buying marketing crap
02:59.57BeeBuuSteveTotaro: how about the callerid num longer than 3?
03:00.28mackesAhhhh. The Thursday night Asterisk IRC party
03:04.15*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:07.26*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
03:11.01*** join/#asterisk Micko113 (n=micko113@c-67-183-170-166.hsd1.wa.comcast.net)
03:11.35drmessanoIt's just Beebuu
03:13.21BeeBuuhi,drmessano
03:13.37BeeBuunice to meet you.
03:13.46*** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com)
03:13.57BeeBuuhow's the day?
03:15.09drmessanoIt's fantastic
03:15.20drmessanoDid you manage to install Asterisk yet?
03:15.37BeeBuuno.
03:15.43drmessano:(
03:16.02BeeBuumy job is not install asterisk,just using it
03:16.16drmessanoMaybe when the Windows version gets more stable you can run it on Windows Vista with an X100{
03:16.21drmessanoX100p*
03:16.51BeeBuui got your point. i don't using X100p anymore.
03:18.10BeeBuudrmessano: is there a windows version asterisk?
03:18.25BeeBuuor you are kidding me?
03:18.33drmessanoYes, there is
03:18.39drmessanoAre you interested?
03:19.03BeeBuuyeah,may i get the URL from you?
03:19.32drmessanoIn my country, that would be considered slander, but you may google for it
03:19.33BeeBuujust for other guys...
03:19.50drmessanoNot sure if you are familiar with google, but it's Lycos, but without the dog
03:19.57drmessanoIt's quite good, nonetheless
03:21.03BeeBuuis asteriskwin32?
03:21.28BeeBuudrmessano: thanks .you are a good man.
03:21.47drmessanoYes, you are quite welcome..
03:22.31BeeBuudrmessano: i know i am not lovly,but you still help me alot...
03:23.12drmessanoI do what I can
03:23.20drmessano~drmessano
03:23.20jbot[drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway, and is earning his reputation daily
03:23.28drmessanoIndeed
03:23.52*** join/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net)
03:24.07BeeBuu:-)
03:24.30BeeBuuDr. messano?
03:37.27*** join/#asterisk PepOSX (n=angeldav@200.93.28.168)
03:38.40*** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca)
03:42.15rbdhey guys, is it safe to use sleep/nanosleep inside of asterisk applications app_*.c ?
03:42.31tzangerthat doesn't seem like a good idea to me
03:43.16rbdtzanger: yeah...I do see calls to ast_safe_sleep
03:43.23rbdthat's probably the better one to do
03:45.04*** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320)
03:45.20*** join/#asterisk AndyGraybeal (n=AndyGray@128-177-27-78.ip.openhosting.com)
03:46.00TJNIINonono, go ahead and use nanosleep and you can send users off on wild goose chases trying to track down the source of lag in their networks when it is really just asterisk stopping.
03:46.33TJNII(even though that probably wouldn't happen)
03:50.47BeeBuuDR.messano: how can i make slient when dial?
03:56.17*** join/#asterisk SomethingISODD (n=dancole@S010600a0d1757bfb.cg.shawcable.net)
03:56.24SomethingISODDHello all
03:56.47SomethingISODDquestion does anyone know where i can find a php script thats been developed to monitor traffic. how long they are connected etc?
03:57.14*** join/#asterisk killmel8tr (n=IceChat7@c-69-244-155-174.hsd1.mi.comcast.net)
03:57.17TJNIIRealtime or as a log reader?
03:57.39*** join/#asterisk mpruett (n=mpruett@24-240-203-84.static.stls.mo.charter.com)
03:57.58SomethingISODDTJNII realtime
03:58.03SomethingISODDlike through the manager interface
03:58.14TJNII.....Manager interface?
03:58.29SomethingISODDya manager..
03:58.44TJNIIYou're using a GUI, arn't you?
03:58.58SomethingISODDno right now i do all my asterisk stuff through the configs.
03:59.29SomethingISODDI have been trying to write a script for it in php but i cant get it to print out correctly it all comes out as a big mess
03:59.36TJNIIHeh
03:59.40*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:59.46TJNIIHow are you gathering the info?
04:00.00SomethingISODDdo you know php? if so i could post my script up
04:00.25TJNIII'm not an expert, but I get by
04:00.30SomethingISODDok thanks second
04:01.04mpruettAny ideas guys (not sure if you need more info - i have it if needed) - I am trying to connect to a remote PBX and everything seems to be working fine. Calls in and out but...
04:01.37TJNII(pause for suspense)
04:01.59mpruettwhen I do a wireshark I see the remote PBX send " SIP Request: OPTIONS sip:XXX.XXX.XXX.XXX"
04:02.10mpruettmy box replies 404 Not found
04:02.28mpruett;) - I like suspense!
04:02.35TJNII(The audience gasps)
04:02.41mpruettlol
04:04.13mpruettAny ideas? If more info is needed just let me know what you need.
04:04.18SomethingISODDTJNII http://pastebin.com/d1769d5e8
04:06.39*** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com)
04:07.50TJNIISomethingISODD: Let me tinker with it and get back to you.....
04:09.02SomethingISODDok thanks TJNII
04:10.07*** join/#asterisk xpeed (n=unknown@unaffiliated/xt-9)
04:10.30SomethingISODDmpruett if you dont mind me asking what is wireshark?
04:11.30mpruettNetwork Analyzer - http://www.wireshark.org/
04:11.46mpruettPreviously ethereal
04:11.56SomethingISODDohhh ok
04:12.11SomethingISODDi was wondering why i could never find the ethereal packages lol
04:13.58jayteewireshark rocks!
04:14.56jayteeI used it to good advantage yesterday to monitor my traffice between sipX and Exchange 2007 Unified Messaging.
04:16.07SomethingISODDjaytee can you run it via command line or only through gui?
04:16.10jayteeI've even got call redirection working from Exchange UM to * clients
04:17.12jayteeI use a gui because I'm running it on a Windows Server 2003 64 bit Enterprise Edition.
04:18.09SomethingISODDoh ok.
04:19.53jayteeso I use a combination of sip debug on the console through an ssh session using Putty, wireshark for Windows, wireshark for linux on RHEL5 64 bit running * 1.4 and sipX running in a CentOS 5.1 VM
04:22.40SomethingISODDi am trying to build a web application for asterisk where i can add tracing tools like wireshark
04:22.49TJNIISomethingISODD: Output looks OK on my system.....
04:23.03SomethingISODDTJNII did you view it through the website?
04:23.09SomethingISODDor just through command line
04:23.17TJNIII did throw in a if($socket == FALSE) { echo "fsockopen failed: [" . $errno . "] " . $errstr . "\n"; exit(1);} though
04:23.21TJNIICLI
04:23.39*** join/#asterisk mpruett (n=mpruett@24-240-203-84.static.stls.mo.charter.com)
04:23.41SomethingISODDtry it from a website and you will see what i mean
04:23.45SomethingISODDits a nasty mess
04:23.46TJNIIIt would probably view odd on a website since there is no HTML
04:23.50TJNIIOkay, one sec
04:26.24*** join/#asterisk Micko113 (n=micko113@c-67-183-170-166.hsd1.wa.comcast.net)
04:28.20*** join/#asterisk Wangster (n=Wangster@S01060014bf82696b.wp.shawcable.net)
04:28.28TJNIISomethingISODD: Yea, the problem is no HTML
04:28.35TJNIILet me throw a little in there....
04:29.02SomethingISODDthank you very much TJNII i have been trying with html but i dont know array very well so i think that could be part of the problem
04:29.21km2my * won't start because i took out the PRI card, so it's complaining about lack of zap channels, but i need it to run for another reason. thoughts just to get it limping at least?
04:30.30*** join/#asterisk Kalianyia (n=firewnyd@c-68-35-189-103.hsd1.nm.comcast.net)
04:30.43km2http://pastebin.com/m502ee520
04:35.20km2fixed! i just commented out 'channel => 1-23' in /etc/asterisk/zapata.conf
04:40.10*** join/#asterisk bagc82 (n=unknown@200.114.59.173)
04:41.29xpeed:)
04:49.29SomethingISODDTJNII sorry to bug ya any luck
04:51.21TJNIINot yet
04:51.25TJNIIWas distracted
04:51.39SomethingISODDok np :-)
04:52.35mackes~mackes
04:52.35jbothttp://www.youtube.com/watch?v=P7v7uBA6LW8
04:53.33mackes~mackes
04:53.35CCFL_Man2SomethingISODD: something is odd
04:54.22*** join/#asterisk lzhang (n=lzhang@24-155-240-48.dyn.grandenetworks.net)
04:55.12CCFL_Man2from a cisco sip phone to a cisco sip gateway terminated to T1 which is terminated to a channel bank which terminates a fxs station, i get an echo if i yell, and the echo sound like crap
05:01.02SomethingISODDquestion in sip, how many host= can you have under one singal account?
05:01.20*** part/#asterisk bagc82 (n=unknown@200.114.59.173)
05:05.19SomethingISODDCCFL_Man2 did you try to adjust echo training?
05:09.46TJNIISomethingISODD: Well, I need to go to bed.  You're getting the information, but you need to parse it
05:09.58TJNIII'd suggest not keying the end of your read on EOF
05:10.05TJNIIAnd reading between commands.
05:10.12CCFL_Man2well, i guess the first step is trying to figure out where the echo is from
05:10.37SomethingISODDi tried that didnt spit out correct
05:11.05SomethingISODDCCFL_Man2 i cant help you there i dont run asterisk for stuff like that
05:11.13TJNIIYea, because you can't key on EOF between commands as the socket is still open
05:11.16SomethingISODDi use class 5 switchs for stuff like that
05:11.32SomethingISODDTJNII ya
05:11.43TJNIIPerhaps using something like socket_read or watching for newlines in the while loop
05:12.05SomethingISODDok let me read up on sockets a bit more
05:12.16SomethingISODDthanks TJNII for your recommendations
05:12.28SomethingISODDand ur time
05:12.35TJNIINP
05:12.44TJNIII'll have to look into that manager interface
05:12.47TJNIILooks handy
05:12.54SomethingISODDit is :-)
05:13.02SomethingISODDi do alot of my commands through that
05:13.30SomethingISODDi will be releasing my scrip to the asterisk community once its done
05:13.40SomethingISODDi believe it will be one  of the most advanced setups
05:13.50CCFL_Man2SomethingISODD: there is no asterisk in the system, there is a pots phone to fxs card on channel back, to T1 from channel bank to cisco voice gateway, to voip dial peer to sip phone, i suppose the echo is in the T1 interface?
05:14.33SomethingISODDya i would think so trying connecting your phone direct to that device
05:14.54SomethingISODDbasically go backwards from your device one no device each test. and see when you find the echo
05:15.15SomethingISODDi am suspecting it will be on your t1 interface as thats where i found it on our ds3`s
05:17.45CCFL_Man2ahh
05:17.50CCFL_Man2thanks
05:19.19SomethingISODDno==new btw sorry tomany things on the go and i am all fsked up because of this php script
05:19.24SomethingISODDput me behind two weeks so far lol
05:20.13CCFL_Man2lol
05:21.18SomethingISODDyou dont happen to know php do you
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05:41.57vectorSomethingISODD, I'm up late and know php if you want some help
05:43.29SomethingISODDvector i would love some help its killing me lol here is the script http://pastebin.com/d433ce398
05:43.55SomethingISODDthe problem is i cant figure out how to get each line from the socket to print out in on a browser as a new line
05:44.42vectordoes it print to the browser at all?
05:44.52SomethingISODDya let me show you that as well
05:44.57SomethingISODDhttp://www.airstarcommunications.com/billing/durations.php
05:45.20vectorah
05:45.21vectorheh
05:45.33SomethingISODD?
05:45.45vector<pre>
05:45.48vectortry that
05:45.55vectorbecause  \n means nothing in html
05:46.02SomethingISODDok
05:46.18SomethingISODDi dont have \n i have <br>
05:47.04vectorwell YOU do.. but what comes out of that socket has \n on the end of each line
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05:47.24SomethingISODDwhops :-p
05:47.29vectorso basically everywhere that you need it to drop to the next line but it's not showing that way... that's a \n from the socket :)
05:47.54vectorother way to do it is to simply add <br> to the end of each line as you get it out of the socket
05:48.35SomethingISODDi tried that
05:49.10vectoroh
05:49.12vectorI see what you did
05:49.27vectorwhen you send like "show applications" to the socket.. use \n
05:50.22vectorit's the stuff comming OUT of that socket that we want to put <br> at the end of.. OR you can enclose the entire page in a <pre></pre> tag (preformatted)
05:50.30vectorsorry I wasnt very clear about that
05:50.42SomethingISODDok let me give that a shot
05:51.08SomethingISODDdo i incluse pre with ""?
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05:51.46SomethingISODDthank you
05:51.49SomethingISODDone more question for you
05:52.05vectork
05:52.40SomethingISODDeasy to remove the headers from it. if you go refresh the durations.php script you will see how it has asterisk call manager blah blah lol and at the end the message saying good buy
05:53.05vector(http://pastebin.com/d412997fe   that's how I would use <pre>  .. just FYI)
05:53.24SomethingISODDok
05:54.32vectorok so you just want to take the part at the beginning and at the end out right?
05:55.09SomethingISODDyes.
05:55.23SomethingISODDthere is alot more to it but think this will get me in the right direction.
05:56.34vectorpm
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07:17.32lun_anyone?
07:18.21Strom_Mlun_: the answer is "buy more toner"
07:18.36lun_hehe
07:21.43patrick--Hey, can someone tell me why this call in my logfile caused the asteriskd to crash? http://phpfi.com/319019 I see its something about an unsupported format...
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07:31.27LuisTorresHi
07:31.36BeeBuuhi hi
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08:06.07jim_1Hello!
08:06.16LuisTorresHowdy
08:07.18jim_1Asterisk is really powerfull!
08:07.59LuisTorresSometimes after a Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105) I get "decline" messages from all the phones. Im using Polycoms ip550 .., does anyone experienced this too?
08:08.13jim_1Who ever worked with the function Playtones ??
08:08.14jim_1http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playtones
08:08.37jim_1No, LuisTorres
08:08.42jim_1i can't help you
08:08.52LuisTorresthankyou
08:08.53jim_1maybe set the register timeout lower
08:09.03LuisTorresits on 60secs
08:09.07jim_1ah okay
08:09.25LuisTorresalso already try with qualify =yes , but still happen
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08:09.57fluffdoes the manager-action QueueStatus ever return a Event: QueueMemberStatus?
08:09.57jim_1try to upgrade the firmware in your phones
08:12.24jim_1exten => _X.,n,Playtones(ring) & Dial
08:12.45jim_1can someone help me. I want to use Playtones and dial at the same time ...
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08:25.00BBHossjim_1: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Playtones
08:26.10jim_1Thanks BBHoss, but Playtones works but before or after dialing my mobile phone number
08:26.22jim_1i want to use it While asterisk is dialing my mobile number
08:26.39jim_1the caller hears nothing for about 4 seconds now
08:26.41BBHossdid you read the example titled Playing tones while dialing?
08:26.46jim_1yes
08:26.57jim_1using &
08:27.29jim_1i'm trying for hours now but
08:27.43jim_1i can't solve my problem ...
08:27.49BBHosstry copy/pasting the example, see if it works
08:28.05BBHossyou have the & going to a no-op context right?
08:28.16jim_1yes
08:28.27BBHosswhat version asterisk are you running?
08:28.31jim_11.2.10
08:28.40BBHossdebian?
08:28.48jim_1ubuntu
08:29.09BBHossyeah, you should really upgrade to 1.4, unless you have good reasons not to
08:29.19jim_1yeah i know but
08:29.32jim_1i have a good reason to stay with this verison
08:29.35BBHossthat is a really old ubuntu build anyways, my ubuntu asterisk is 1.4.17 with security backports and patches
08:29.53BBHosswhat would that be?
08:30.19jim_1i have a hangup problem with asterisk versions newer than 1.2.14
08:30.28jim_1i found this on the internet:
08:30.30jim_1Since version 1.2.14, * was changed so that not receiving an ACK to an OK is
08:30.30jim_1considered a FATAL error.
08:30.30jim_1The specific change that causes this problem is in sip_answer() in
08:30.30jim_1chan_sip.c:
08:30.30jim_1res = transmit_response_with_sdp(p, "200 OK", &p->initreq, 2);
08:30.31jim_1Changing the 2 to a 1 will probably fix it.  Note that this is NOT a bug in
08:30.33jim_1* but improper implementations--either caused by latency, or a software bug
08:30.35jim_1(not sending an ACK).  Perhaps it might be beneficial to have an option in
08:30.37jim_1sip.conf to change how * handles not receiving an ACK?  I know... it's
08:30.39jim_1someone else's problem, but might help those of us stuck with buggy
08:30.41jim_1implementations in production environments. :)
08:31.02jim_1Many calles a dropped when i use a newer version of asterisk
08:31.15jim_1it's a problem with my new voip host
08:31.36jim_1so i use this old version to "fix" it
08:31.46BBHossim sure if it was that serious "random calls dropped" someone would have fixed it by now
08:31.59BBHossdo you have a link to a bug on the tracker?
08:32.12jim_1i have this problem:
08:32.12jim_1http://lists.digium.com/pipermail/asterisk-users/2007-April/184553.html
08:33.29jim_1i don't use a billing system just a clean asterisk version
08:34.05oejjim_1: If a device is not sending an ACK, it doesn't understand the very basics of SIP and should NOT be used
08:34.46jim_1yeah i know
08:34.52jim_1but i can't solve the problem
08:35.16jim_1so i need to stay with asterisk 1.2.10
08:35.29jim_1many calls are dropped if i don't ...
08:35.34oejThey should be
08:36.07jim_1how do you mean oej??
08:36.07BBHosswhat kind of tom-foolery are you using for a provider?
08:36.07oejAt some point, you have to require a basic level of interoperability with the core protocol implemented
08:36.44jim_1i'm using a dutch voip provider
08:37.53BBHossno other options?  its a pretty serious implementation flaw, especially if asterisk doesn't allow it.
08:38.13oejNo working SIP device should allow that
08:38.28oejCould also be a problem in your nat/firewall handling
08:38.30oejSo check that
08:38.37oejBefore you stop working with that provider
08:38.57jim_1no that not a nat problem
08:38.58BBHossoej, yeah it really could be that
08:39.16jim_1it's 100% sure not a firewall of nat problem
08:39.26BBHossjim_1 have you tried it with a public ip, no nat involved?
08:39.42jim_1no
08:40.02jim_1but i never had problems with my old provider
08:40.05BBHossjim_1 then you have one step left before you dump your provider
08:40.16jim_1yes i know
08:40.35BBHossspeaking of debain/ubuntu: http://www.lessaid.net/fun/apt-get-wife.png
08:42.09BBHossjim_1, also make sure you take care of that nasty openssl bug that has affected all debian/related distros
08:42.31jim_1ok
08:42.45jim_1i'm updating my server every week
08:42.47mvanbaakwhehehe
08:43.17jim_1BBHos do you think that it's possible to use Playtones
08:43.25jim_1and dial a number at the same time
08:43.33mvanbaakjim_1: did you also regenerate all ssh and dsa keys ?
08:43.39mvanbaakI know I did, it was a mess
08:43.55jim_1no i didn't
08:44.08jim_1only apt-get update && apt-get upgrade
08:44.31mvanbaakthat will only regen the hostkey
08:44.44mvanbaakyou should also regen all ssl keys, ssh keys and asterisk keys
08:44.45tengulreBBHoss: LOL! that 's good ! haha..
08:44.47mvanbaakand probably others
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08:44.49tengulrewife
08:45.21BBHossjim_1, it may only work in newer asterisk, but i would think it should work in 1.2.  However, it looks as if its a "hack" because it uses side-effects of the dial command to work
08:46.03jim_1i really would like to use a newer verion of asterisk but i can't ...
08:46.19jim_1getting this error in my log:
08:46.27jim_1maximum retries exceeded
08:46.29jim_1and then:
08:46.34jim_1hanging up call
08:48.30BBHosswe can't really help if its not an asterisk problem, sorry :(
08:49.45jim_1this works but with the silence before dialing my mobile number:
08:49.49jim_1exten => 1,3,Dial(SIP/101&SIP/106,25,Ttr)
08:49.49jim_1exten => 1,4,Dial,SIP/UITGAANG/MY MOBILE Numer
08:50.12jim_1BBHos thank you
08:50.49BBHossbest of luck finding another provider :)
08:51.20jim_1yeah i know
08:52.16jim_1my provider wants 6 euro every month for transferring calls to my mobile phone
08:52.57jim_1they told my that they're also using asterisk
08:53.09BBHossdoubt it
08:53.30BBHossif they are its probably ages older than yours
08:54.23jim_1Server: Sip EXpress router (0.9.6 (i386/freebsd)
08:54.34BBHossSERP
08:54.39BBHoss-p
08:54.56jim_1do you know what Sip Express router is ?
08:55.11BBHossyeah
08:55.32BBHosssomething asterisk providers use to allow asterisk to scale better
08:55.46jim_1okay
08:55.47BBHossthat version they are running is two years old
08:56.01jim_1may that be the problem ?
08:56.09BBHosscould be
08:56.59jim_1they told me that they are using asterisk 1.4
08:57.17jim_1i can see Server: Sip EXpress router (0.9.6 (i386/freebsd) when sip debug is on
08:57.20BBHossyeah but since they are using SER as a proxy, it doesn't really matter
08:57.29jim_1oh okay
08:57.51BBHosswhat they are doing is using SER to spread load across multiple servers
08:57.52mvanbaakjim_1: what provider are you using ?
08:58.31jim_1may i send you a pm mvanbaak?
08:58.34mvanbaaksure
08:58.50mvanbaakI'm in .nl as well, and using a bunch of dutch providers without trouble
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09:36.23Uatecmorning
09:37.01Uateccan i have a voicemail notification sent to multiple email addresses?
09:41.07jim_1hmm, Sip Express router 0.9.6 is from 2006 ...
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09:55.57Uatecopenser is the wya forward now
09:56.00Uatec*way
09:56.02Uatec#openser
09:57.31patrick--...
09:57.51patrick--is there anyone around thats fermilliar with spandsp and rxfax?
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10:01.46mockerAre priority labels supported in asterisk realtime?
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10:10.10patrick--mocker: explain that
10:10.27patrick--what do you mean by "realtime"?
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10:18.29gr0mithi Faustov
10:20.30LuisTorresHey.., any issues on the new release 1.4.19.2 ? is it safe to upgrade?
10:23.54LuisTorreslool srry didnt realize that 1.4.20.1 is out :P ...
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10:27.35RoyKhttp://www.lessaid.net/fun/apt-get-wife.png
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10:33.28Uatechmm
10:34.18Uatecpatrick--, maybe he means "not fake time", like not at 4.72pm on the 34th of Notbruary
10:35.29creativxNutbruary
10:35.50SteveTotaroRoyK:  funny
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10:37.13Kernel_Corehi all
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10:37.49Kernel_Corewhen I issue "iax2 show netstats " in local I have 14% LOSS
10:38.20Kernel_CoreI use IAX2 Trunk and I have the latest asterisk 1.4.20.1 with speex !
10:39.06kannanhello , i am having problem using a Cisco phone 7960 with *. When i dial sip peers from the phone its fine, but i cannot call zap lines. The asterisk CLI is not showing anything at all. Zap calls from Grandstram / x-lite are fine
10:39.09Kernel_CoreI have 0 percent Loss Packet ... !
10:40.38kannananyone using cisco 7960 on SIP with * ?
10:40.58Kernel_Corekannan: check your extensions.conf
10:41.15kannanKernel_Core : i have
10:41.30Kernel_Coresomething is wrong there
10:41.43kannanhmm ok will re-check and get back
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10:45.24Uateccan i have a voicemail notification sent to multiple email addresses?
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11:05.24kannanKernel_Core : thanks, the extensions was ok, but i had misconfigured Name in the Cisco SIP config menu on the phone
11:05.47kannanthanks
11:05.50kannanbbiab
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12:00.02jeremy_ggirls plz behave yourself
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12:09.49drzedlittle question on extension: how can i "exclude" a number
12:10.41SteveTotaroi trust you are pattern matching?
12:11.17drzedor is it possible to use regexp style expreession like _0[5-9].
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12:11.58drzedyes number pattern matching
12:11.58SteveTotaroare you pattern matching drzed?
12:12.14SteveTotaroi know one way
12:12.34*** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com)
12:13.00SteveTotaroyou can have a context containing the numbers you don't want to match on, and do whatever you want with them and then include another context with the pattern match you originally had
12:14.28SteveTotarothe first context will be applied and if no match it will go to the include
12:15.01drzedhm sounds like a good idea
12:15.14*** part/#asterisk Oy90 (n=ivan@213.187.111.94)
12:15.24SteveTotarodisclaimer, all my info is based on 1.2
12:15.40SteveTotaroso if 1.4 has changed this, it may not work
12:15.59drzedim also unsing 1.2 so no problem heree
12:16.01drzed-e
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12:24.13LuisTorresHi..., Any Issues known on the 1.4.20.1?
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12:26.30russellbnope, it's perfect :)
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12:31.46*** part/#asterisk Oy90 (n=ivan@213.187.111.94)
12:34.22NuggetZaroo bugs!
12:35.17russellbno blatant regressions that we are aware of, no.
12:39.55LuisTorresthanks
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12:48.06tuxx-heya
12:48.37Nuggetmoo
12:49.06tuxx-where does asterisk store the voicemail soundfiles? I can't seem to find them anywhere... I know that when u call VoiceMailmain that u can record a soundfile for your voicemail... But is there a way to store a soundfile in some dir manually?
12:52.33glaz/var/lib/asterisk/sounds iirc
12:54.09rob0no, that should be astspooldir => /var/spool/asterisk
12:54.28rob0(from asterisk.conf)
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12:57.01*** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net)
12:57.12fetcherI'm having trouble with supervised transfers between NAT'ed Polycom phones.  sip.conf already has nat=yes, canreinvite=no set.  Anything else to try?
12:57.20fetcherchan_sip.c:6930 get_refer_info: Supervised transfer requested, but unable to find callid '424ac69-87c12d67-641128b2@192.168.111.3'.  Both legs must reside on Asterisk box to transfer at this time.
12:57.57fetcherwith canreinvite=no, won't Asterisk be in the media path for any call no matter what?
13:01.14*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:01.23tuxx-ah nice, thanks rob0 :-)
13:03.38fetcherDid 1.2.13 have any known supervised-transfer bugs that were fixed later?
13:03.52*** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com)
13:04.23[TK]D-Fenderfetcher: Dial options and monitoring will keep it in the path
13:04.33iratikif i wasn't that concerned with top quality .... who is the cheapest / most reliable sip/iax trunking provider ?
13:07.01*** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz)
13:07.40fetcher[TK]D-Fender: hmm, just setting canreinvite=no isn't enough?
13:08.08fetcherI guess I could try option 'T' to see if that helps
13:08.12[TK]D-Fenderfetcher: all of those give * reason to sit in the middle.
13:08.48*** join/#asterisk imesper (n=chatzill@200.142.121.162)
13:10.39*** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz)
13:11.05*** join/#asterisk km2 (n=x@adsl-76-252-245-25.dsl.pltn13.sbcglobal.net)
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13:13.40*** join/#asterisk eXistenZ (i=existenz@unaffiliated/existenz)
13:15.21patrick--is there anyone around thats fermilliar with spandsp and rxfax?
13:15.50*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
13:15.50*** mode/#asterisk [+o lmadsen] by ChanServ
13:20.43tuxx-hmz. when i put the files unavail.gsm and busy.gsm in the /var/spool/asterisk/voicemail/default/306/ dir it still plays the 'main' voicemail file... and i cant seem to find that either in the spool directory. is there a way in the config files that you can set a global voicemail soundfile or something like that?
13:21.16yang~grandstream
13:21.16jboti guess grandstream is the Yugo of VoIP hardware.  Run.  Run away now.
13:22.11*** join/#asterisk bsaxon (n=bsaxon@12.68.234.174)
13:22.22*** join/#asterisk coppice (n=chatzill@106.198.17.210.dyn.pacific.net.hk)
13:22.59*** join/#asterisk RoyK (n=roy@fw.fortel.no)
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13:25.15Nuggetheh
13:26.29*** join/#asterisk Sajjad_Ali_Musht (n=Sajjad_A@invite36.enst-bretagne.fr)
13:26.51*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
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13:29.14a-shello
13:29.25a-sdoes speex codec of asterisk works for somebody?
13:29.44[TK]D-Fendertuxx-: You are the one responsible for telling voicmail which recording to play in your dialplan.
13:30.54a-sEven it appears loaded in `show translations`, I always get [May 23 16:21:25] NOTICE[2535]: chan_sip.c:5500 process_sdp: No compatible codecs, not accepting this offer!
13:30.54a-s<PROTECTED>
13:32.05[TK]D-Fendera-s: And you are NOT showing us the SIP debug of your failed call attempt along with your configs.
13:32.52[TK]D-Fendera-s: So clearly you could not possibly have done something wrong and * must be broken.
13:32.54NovceGuruHello, anybody have an opinion of the asterisk appliance?
13:34.26a-s[TK]D-Fender: :) ok, one moment please...
13:37.34*** join/#asterisk ThaProZac (n=DumbWebb@fw.fortel.no)
13:38.13*** join/#asterisk tobias (n=tobias@cpe-069-134-035-018.nc.res.rr.com)
13:39.06glazNovceGuru: what do you want to know?
13:40.43a-s[TK]D-Fender: that's it
13:40.55a-s[TK]D-Fender: I enabled sip debug
13:41.11a-showever, when I call, no sip packet is transmitted
13:41.31*** join/#asterisk ThaProZac (n=DumbWebb@fw.fortel.no)
13:41.40[TK]D-Fendera-s: "May 23 16:21:25] NOTICE[2535]: chan_sip.c:5500 process_sdp: No compatible codecs, not accepting this offer!"
13:41.51[TK]D-Fendera-s: how are you getting a warning without getting a packet?
13:41.52a-s*CLI> [May 23 16:36:37] NOTICE[2635]: chan_sip.c:5500 process_sdp: No compatible codecs, not accepting this offer!
13:41.52a-s<PROTECTED>
13:42.25NovceGuruglaz: if it supports presence
13:42.39a-s[TK]D-Fender: exactly. I got the NOTICE message, ans no sip packet
13:42.49a-sand
13:42.59[TK]D-Fendera-s: then debug isn't enabled properly.
13:43.10[TK]D-Fendera-s: because thats clearly a call attempt
13:43.15Slashmanis there a way to use {SSHA} or {CRYPT} in the sip.conf for the secret ?
13:43.16a-souf!
13:43.41glazNovceGuru: http://www.voip-info.org/wiki/view/Asterisk+presence
13:45.15patrick--Hey all, im using SpanDSP and rxfax to receive Faxes over my mISDN channels. But in some cases the Tiff Faxes are not viewable. Could anyone think of why this is the case?
13:45.22a-s[TK]D-Fender: I did sip debug ip ... and I enabled debug
13:45.30NovceGuruI know asterisk does, but if the asterisk appliance gui had settings for it
13:45.34a-safterwards I made the call.
13:45.57*** join/#asterisk hsv-al (n=ccvp@66.0.46.210)
13:46.01glazSlashman: you have to have mysql support, sip in mysql.
13:46.12*** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku)
13:46.33glazNovceGuru: I doubt it, I don't like/use the gui.
13:46.53Slashmanglaz : I'll wait for integrating ldap support then
13:47.24*** join/#asterisk railsmunky (n=nick@collaboration.capuk.org)
13:47.29railsmunkyback again :)
13:47.45*** join/#asterisk danlock2 (n=bean@wikipedia/danlock2)
13:48.12railsmunkyi'm getting a No application 'DigitTimeout' for extension ... any ideas?
13:49.54*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
13:50.21ZeeekGood Morning
13:51.10*** join/#asterisk Ubluzok (n=ubluzok@62.141.89.219)
13:51.35*** join/#asterisk maruz (n=maumar@88-149-241-192.dynamic.ngi.it)
13:52.15danlock2morning
13:52.17a-s[TK]D-Fender: Now it transmitted sip packets, and codecs are incompatibles...
13:52.22maruzin manager, Event: Newchannel trim 0 on front callerid number
13:52.33*** join/#asterisk putnopvut (n=putnopvu@216.207.245.1)
13:52.33*** mode/#asterisk [+o putnopvut] by ChanServ
13:52.36maruzCallerID: 721234567
13:52.46maruzinstead of 0721123456
13:53.03maruzbut only this Event, the others has it
13:53.39maruzcan i configure asterisk to get this 0?
13:53.57maruznationalprefix=0 doesn't fix it
13:54.16hsv-alwhy am I being spammed with this?
13:54.20hsv-al[May 23 08:53:32] WARNING[10879] config.c: Unknown directive '' at line 231 of /etc/asterisk/../zaptel.conf
13:54.32*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
13:54.51[TK]D-Fendera-s: "thats nice".
13:54.58Zeeekhsv-al: what is at line 231 ?
13:54.59danlock2hsv-al: somewhere in your zaptel.conf you have something wrong.
13:55.33[TK]D-FenderI wonder why it is there ARE even that many lines in there...
13:56.03a-s[May 23 16:49:00] NOTICE[2686]: chan_sip.c:5500 process_sdp: No compatible codecs, not accepting this offer!
13:56.03a-s<PROTECTED>
13:56.06Zeeeklines are cheap
13:56.53Zeeekoff the wall guess: a capture of a text that has a bunch of blank lines and a wacky character
13:57.19[TK]D-Fendera-s: Are you going to keep spamming that useless message over & over?
13:57.38Zeeekthe word spam has now been overused on this channel
13:57.51tzafrir_homehsv-al, that stupid warning is because something is trying to read zaptel.conf as an asterisk config file
13:58.10patrick--Is anyone using tx and rxfay?
13:58.12patrick--fax*
13:58.13tzafrir_homein asterisk everything that begons with a '#' is a special directive
13:58.26[TK]D-Fenderpatrick--: just get to your actual question.
13:58.44NovceGuruglaz: I've always steered clear of GUIs but thought the "official" one might be better
13:59.16hsv-altzafrir, ive been trying to make a custom addon, so i can manually specify a youtube URL
13:59.19tzafrir_homeThe really funny thing would be a line in the lines of zaptel.conf '#exec echo rm -rf /etc/asterisk'
13:59.24[TK]D-Fenderpatrick--: Maybe its just an incomplete transmission.  Have you tried other viewers
13:59.29Zeeekyou shouldn't use a GUI behind the wheel. CLI only
13:59.30hsv-alfor hold music.
13:59.47glazNovceGuru: I know you're a freebsd admin, I doubt you have any problems running asterisk with the CLI.
14:00.22[TK]D-FenderNovceGuru: Stop thinking and start trying.
14:00.26sp00kzthe less crap on your asterisk box, the less troubleshooting you'll do :p
14:00.35*** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130)
14:00.50NovceGuru[TK]D-Fender: well I was considering purchasing the * appliance but wanted to ask around a bit before dropping $1500 :P
14:01.06patrick--[TK]D-Fender: its weird, cause it happends only now and then
14:01.08ZeeekNovceGuru:  the appliance works great
14:01.13patrick--im using the windows tiff viewer
14:01.14[TK]D-FenderNovceGuru: What are you hoping to get out of it vs building a normal server?
14:01.22oejtzafrir_home: '#exec "echo rm -rf /etc/asterisk; echo \"#include /etc/passwd\")"
14:01.26[TK]D-Fenderpatrick--: are all faxes bad?
14:01.39Zeeekfax should be banned forever
14:02.05patrick--no
14:02.24patrick--[TK]D-Fender: not all...
14:02.40a-s[TK]D-Fender: No, I won't, but please give me an idea what to do to make speex work... :(
14:03.03[TK]D-Fendera-s: pastebin the debug & your configs or stop wasting our time.
14:03.09hsv-aldo I have to buy the $10 license, if I get a TDM411e?
14:03.26coppicepatrick--: mISDN is bad news for anything requiring an accurate audio stream
14:03.27[TK]D-Fenderpatrick--: I'd bet its just a failed fax.  I've gotten those on occasion.
14:03.59[TK]D-Fenderhsv-al: huh?
14:04.12hsv-alhttp://store.digium.com/productview.php?product_code=HPECLIC
14:04.24hsv-alqwell was telling me about this, but iwas confused what he was talking about needing this or not?
14:04.45hsv-alIm buying a 411e for home use today
14:04.54[TK]D-Fenderhsv-al: that card has HARDWARE EC, no need for the software EC
14:05.22hsv-ali mis=understood then, ok cool
14:05.59[TK]D-Fenderhsv-al: and HPEC is free for owners of Digium cards under warranty
14:06.36hsv-alfigured I get this card instead of wasting the money on some burberry ties
14:06.37hsv-al&:^)
14:07.14Zeeek[TK]D-Fender: this is for the FXO modules?
14:07.37[TK]D-FenderZeeek: Applies to any zaptel channel
14:08.11a-s[TK]D-Fender: http://asterisk.pastebin.ca/1027079
14:08.47ZeeekI ask because we don't have much trouble with echo on our old cards but I do not use the FXO modules I own on them
14:09.15[TK]D-Fendera-s: First, you can't spell : insercure = port, invite
14:09.34[TK]D-Fendera-s: second you didn't even SPECIFY your allowed codecs in there at all.  And you didn't provide the SIP debug.
14:09.40*** join/#asterisk murdock_ut (n=chatzill@70.99.184.194)
14:10.02a-s[TK]D-Fender: one moment please...
14:10.10[TK]D-Fendera-s: But before even bothering with SIP debug, you didn't bother setting your codecs in the first place.
14:10.57Zeeekwhy doesn't someone write a pre-scanner for the .conf files that outputs stuff like "Did you mean insecure?"
14:11.49jblackA linter for asterisk config files would be nice.
14:11.58Zeeek"You have extensions that start with 'n' - that won't cut it"
14:12.03railsmunkyany ideas :)
14:12.11railsmunkyGetting the same for ResponseTimeout too
14:12.44Zeeek"what's this shit about priorities not being in order the extension [sipusers] ?3
14:12.52[TK]D-Fenderrailsmunky: those 2 apps were deprecated in 1.2 and removed entirely in 1.4
14:13.01railsmunkyah :)
14:13.07[TK]D-Fenderrailsmunky: you're reading outdated docs
14:13.15[TK]D-Fenderrailsmunky: "core show function TIMEOUT"
14:13.29railsmunky[TK]D-Fender Brilliant thanks!
14:14.12patrick--[TK]D-Fender: weve gotten lots
14:14.42ZeeekI want to know who has the biggest on the channel right now.
14:14.51Zeeekasterisk installation, I mean
14:15.53MaliutaZeeek: size isn't everything
14:18.10ZeeekI know, mine is very small
14:18.49Zeeekbut in about 100 minutes we have a live conference on asterisk and higher call volumes
14:19.02Zeeekhttp://x2z.eu will get you the info
14:19.46a-s[TK]D-Fender: that's it
14:19.50a-s[TK]D-Fender: look again
14:19.52a-shttp://asterisk.pastebin.ca/1027084
14:19.52a-s<PROTECTED>
14:20.23[TK]D-Fendera-s: Wheres the call debug?
14:20.34a-sI did not provide sip debug, because I succedded to make the call and to answer; my new problem is that I hear nothing :(
14:21.04[TK]D-Fendera-s: And where is your phone relative to *?
14:21.10Zeeeka-s:  is there video though?
14:21.34a-s[TK]D-Fender: http://asterisk.pastebin.ca/1027089
14:21.47a-sI put the message from sjphone too
14:22.21a-s[TK]D-Fender: what do you mean by `phone relative to *` ?
14:22.39a-sZeeek: it's not a video call, just audio
14:22.39[TK]D-Fendera-s: what networking sits between your phone and *?
14:22.57a-s[TK]D-Fender: ah, I explain the topology:
14:23.56*** join/#asterisk merkurie (n=merkurie@192.153.163.44)
14:24.08a-sPBX <- (registered to)  *
14:24.09a-s* <- PHONE1 (ULAW)
14:24.09a-s* <- PHONE2 (SPEEX)
14:24.24a-s* registered to pbx
14:24.34*** join/#asterisk apocn (n=apo@unaffiliated/apocn)
14:24.36*** join/#asterisk axisys (n=axisys@155.70.141.45)
14:24.36*** part/#asterisk apocn (n=apo@unaffiliated/apocn)
14:24.45a-sI call from phone 1 to phone2
14:25.02[TK]D-Fendera-s: WRONG ANSWER
14:25.24[TK]D-Fendera-s: I said what NETWORKING sits between * and your phones?  that means SWITHES, ROUTERS, NAT, ETC.
14:25.25a-s[TK]D-Fender: ??? what information do you need in plus ?
14:25.36[TK]D-Fendersdsfdsfdaasd
14:26.00NuggetYour hat size and your favorite color.
14:26.18Zeeekyou can leave your hat on
14:26.37a-s[TK]D-Fender: aaaah!
14:26.42a-sto look....
14:26.57MaliutaZeeek: no, we don't like robert palmer in here ;)
14:27.07ZeeekRandy Newman
14:27.12Zeeekbut we digress
14:27.23[TK]D-FenderZeeek: Joe Cocker
14:27.24Nuggetjoe cocker did a cover of that song too.
14:27.35ZeeekSJ Phone is like Opera. Some people really like it. I've never gotten it to work on any system
14:27.35a-sphone 1 -> cisco -> switch
14:27.35a-sphone 2 sjphone -> my computer -> switch
14:27.53[TK]D-Fendera-s: what is "cisco:"
14:28.07ZeeekSeesco
14:28.13a-scisco is a sip->analog converter
14:28.29Maliutain my case the 7941 sitting on my desk
14:28.31*** part/#asterisk fetcher (n=jnh@ip-72-55-165-168.static.privatedns.com)
14:28.42Maliutacisco is all things ta all people
14:28.50Zeeekin my case, a stock that split at $100 in the heady days before the bubble burst
14:29.11Maliutagets all zen on cisco
14:29.18Zeeekand I even own a phone since they bought linksys who bought sipura
14:29.44Maliutamy cisco has Che Guevara as the logo
14:30.06Zeeekfor you big cisco phone freaks, get the second or third season of west wing
14:30.14Nuggethttp://macnugget.org/stuff/asterisk-cow-real.bmp  :)
14:30.26MaliutaZeeek: been there, seen that
14:30.37ZeeekMartin Sheen for prez
14:30.50MaliutaZeeek: and did you know that NCIS uses Logitech trackballs?
14:31.16Zeeekhaven't been watching yet but now that I have a proxy that works, I can download them free from Hulu
14:31.57Zeeekspeaking of TV, I hope to transmit some stuff from Asterisk Tag next week, maybe Mark Spencer's keynote. Like Steve Jobs does with Apple :)
14:32.33Zeeekhttp://asterisktv.com/
14:33.07Zeeekright now there's a cute model doing an asterisk install
14:33.48a-s[TK]D-Fender: I got it
14:33.52Zeeekshe's actually explaining how to find a goos SIP carrier!
14:33.58a-s[TK]D-Fender: thanks a lot for help
14:34.12a-sthe microphone from the computer did not work
14:34.30a-sand I heared nothing
14:34.48NovceGuruSorry guys busy morning, [TK]D-Fender: It just seemed like a nice all in one solution and I'm a fan of embedded hardware
14:35.03ZeeekNovceGuru:  the appliance works well
14:35.14Zeeekmine is voip-only
14:37.16NovceGuruhmm, i'll need some fxs ports
14:37.39Zeeeknaw, just buy one of these: http://x2z.eu/h
14:37.43*** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom)
14:38.01danlock2any ideas why my system would claim to see a call coming in, claim to answer it, but i just hear ringing on my phone?
14:39.12ZeeekGod I have way too many phones for two people
14:39.35*** join/#asterisk macros73 (n=cs@c-24-131-77-140.hsd1.pa.comcast.net)
14:41.45*** join/#asterisk Kobaz (n=kobaz@its.kobaz.net)
14:42.18tzangerZeeek: my neighbour has like 8 tvs for 2 people
14:42.27Zeeek"Now it is time for another test" I love this video
14:42.41JenniferAkemi[TK]D-Fender: do you have recommendations for IP in quebec?
14:43.39[TK]D-FenderJenniferAkemi: www.3menatwork.com
14:44.03tzangermen at work? isn't that an oxymoron?
14:44.14JenniferAkemi[TK]D-Fender: what do you like about them
14:44.26sp00kzyes it is tzanger
14:44.41[TK]D-FenderJenniferAkemi: Price/performance, and they wholesale to my ISP which I love.
14:45.04[TK]D-Fendertzanger: No, jsut an alternative-lifestyle 80's band ;)
14:45.32tzangerheh
14:45.33Zeeekit's raining!
14:45.49tzangerlooks out the window
14:45.50tzangerno it's not
14:46.03Zeeekis so
14:46.08JenniferAkemiblah, no prices listed.
14:46.27*** join/#asterisk rickross (n=rickross@supporter/active/rickross)
14:46.28JenniferAkemiany idea on a ballpark for 10 megs?
14:46.50Zeeekhates the endless number of voip sites with totally opaque pricing and services
14:47.06[TK]D-FenderJenniferAkemi: What sector?
14:47.17JenniferAkemi[TK]D-Fender: what do you mean?
14:47.27JenniferAkemi[TK]D-Fender: business?
14:47.27[TK]D-FenderJenniferAkemi: Geopgraphically.
14:47.31JenniferAkemi[TK]D-Fender: montreal
14:47.38[TK]D-FenderJenniferAkemi: When in town?
14:47.40[TK]D-Fenderwhere*
14:47.48JenniferAkemi[TK]D-Fender: downtown
14:47.57[TK]D-FenderJenniferAkemi: Core = $1000, outer = 1300 +/-
14:48.01Maliutaahh canuks
14:48.15Maliutamy parents live in Fort Macmurray
14:48.17JenniferAkemi[TK]D-Fender: are you affiliated with them?
14:48.37[TK]D-FenderJenniferAkemi: nope, jsut researched a lot for the day job.  We've got a T1 I'd like to abolish.
14:48.53JenniferAkemi[TK]D-Fender: thanks
14:49.00*** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net)
14:49.09JenniferAkemi[TK]D-Fender: just wanted something to go along wiht the telus and bell quotes
14:49.28*** part/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net)
14:49.34*** join/#asterisk spokra (n=spokra@host093-179-178.sea0.speakeasy.net)
14:49.47JenniferAkemithe telus and bell sales guys both give us great prices, then go back to draw up the contracts and are told, you can't sell it for that, you have to add X and Y and Z into the cost. it's getting quite annoying.
14:49.51[TK]D-FenderJenniferAkemi: www.colba.net
14:50.02[TK]D-FenderJenniferAkemi: They undercut the other guys.
14:50.34[TK]D-FenderJenniferAkemi: was considering them as well.  I'm just outside of the core so they were better for me.  Not the best customer service though... PITA to get ahold of someone IMO
14:50.48[TK]D-FenderJenniferAkemi: Bell are total fuck-offs.
14:50.51JenniferAkemiheh
14:51.04[TK]D-FenderJenniferAkemi: too me 2 weeks to get a SALES CALL.
14:51.17JenniferAkemiyeah i know what you're saying
14:51.30[TK]D-FenderJenniferAkemi: I've got fiber on-site and they can't deal for shit.
14:51.52JenniferAkemiyeah
14:51.55*** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net)
14:52.05JenniferAkemii'd like it to come in on fibre
14:52.06Zeeekspeaking of Digium, I hope y'all will be on with us in an hour at http://x2z.eu for Mike Trest's spot on "big and fast" with asterisk ?
14:52.59*** join/#asterisk rickross (n=rickross@supporter/active/rickross)
14:53.17Zeeek<PROTECTED>
14:53.51JenniferAkemido you guys ever talk about high availability solutions on teh conference?
14:54.14Zeeekyes, we hope to. I believe that Mike, today's guest could speak to that
14:54.28ZeeekTake a look at his profile: http://mike.trest.com
14:55.10ZeeekANyone who is ready to help the rest of us learn is welcome to speak. The conference belongs to everyone interested in VoIP and asterisk
14:55.19JenniferAkemicool
14:55.39Zeeekstarts in one hour, check it out
14:55.48JenniferAkemii'll try to
14:55.57Zeeekhttp://x2z.eu (spam seems to be big today, why should I be an excaption)
14:56.06Zeeekor exception even
14:56.22Maliutaanyone in .ca have a recommendation for VoIP provider doing DID's in Alberta?
14:57.06Zeeekby the way, join #voip-users-conference if you are interested in the topics, you can follow there
14:58.23*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
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15:04.51banzaikaanybody experiencing problems with Voicepulse service using digium hardware ?
15:05.18[TK]D-FenderBananaskin: How many MPG should I be getting on my Segway?
15:05.58Kobazdo de do
15:06.15Kobazokay, so i have an avaya pbx over here, just sitting minding it's own business
15:06.28Kobazi have two fxs's configured on it, one to a phone, one to an fxo on asterisk
15:06.36*** part/#asterisk imesper (n=chatzill@200.142.121.162)
15:06.56*** join/#asterisk darviria (n=darviria@87-194-177-180.bethere.co.uk)
15:06.59Kobazi call an extension mapped to the asterisk fxo, asterisk will do a pickup and play some tacks
15:07.02Kobaztracks...
15:07.13Kobazbut as far as the avaya is concerned, the circuit is still ringing
15:07.19Kobaz*but* this doesn't happen all the time
15:07.40Kobazone out of every 5-10 calls will just keep ringing even though asterisk picked up...
15:07.50Kobazso... is this an asterisk issue or an avaya issue
15:07.59arbuser(or a hardware issue)
15:08.12Kobazthis happens with a digium fxo as well as a sangoma fxo
15:08.24Kobazwith either a nortel pbx in the middle, or an avaya pbx in the middle
15:08.32arbuserah
15:08.32arbuserok
15:08.46Kobazi've duplicated the problem we had with the nortel with our avaya here
15:09.02[TK]D-FenderKobaz: And if you plug a cheap-o phone in place of *?
15:09.15Kobazgood question
15:09.47*** join/#asterisk moy (n=moyhu@nat/ibm/x-ee0be127875797dc)
15:10.00Kobazi've called a single line set many times and never noticed an issue
15:10.03ZeeekBananaskin:  I never have
15:10.36Kobazi've never specifically tried to see the behavoir on a consistant basis
15:11.24[TK]D-FenderKobaz: On that port?  Maybe the avaya port is bad
15:12.56Kobaznot specificall on that port
15:13.06[TK]D-FenderKobaz: then go test it
15:13.21Kobazaye
15:14.24*** part/#asterisk Oy90 (n=ivan@213.187.111.94)
15:15.48danlock2anyone know what "zt_handle_event: ring/off hook in strange state 6 on channel 1" means?
15:16.31spokraI've used them.. things worked fine.
15:16.49spokraVoicepulse that is
15:17.27ZeeekVoicepulse rocks
15:17.36*** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku)
15:17.46*** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net)
15:17.55spokrathere api needs a little work
15:17.57*** join/#asterisk bbryant (n=brett@216.207.245.1)
15:18.10bsdwarriorFor someone reason the pickup group on one phone does not work. Ive tried several things. anyone have any suggestions?
15:18.18ZeeekI never used it, just the macro
15:18.37Kobaz[TK]D-Fender: but here's the weird thing, this also happened on a nortel, same exact behavoir
15:18.48Kobazcompletely different hardware, etc
15:18.58[TK]D-Fenderbsdwarrior: pastebin is your friend.
15:19.02spokraah i wrote a web site to provision new numbers etc via there api..
15:19.05Kobazbasically my question is... is there something on the asterisk side that can be adjusted
15:19.40Kobazit seems like asterisk isn't doing a polarity switch when it's picking up the call, sort of thing
15:19.46Maliutabsdwarrior: solar flares
15:19.48Kobazor which ever the default behavoir of an fxo is
15:20.10Kobazwhat is the default behavoir on pickup anyway? does it just go off hook?
15:20.13Maliutabsdwarrior: based on what you have told us, definatly solar flares
15:21.51Zeeek[TK]D-Fender:  no one is your friend here
15:25.39*** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-233.usadatanet.com)
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15:27.03*** mode/#asterisk [+o drumkilla] by ChanServ
15:28.03Kobaz[TK]D-Fender: it's definitly asterisk or the server
15:28.12danlock2*sigh* anyone ever seen this error message chan_zap.c:4135 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1
15:28.13Kobaz[TK]D-Fender: i made about 50 calls phone to phone without issue
15:28.25*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)
15:28.29Kobaz[TK]D-Fender: within the first 10 of phone -> asterisk fxo, i get the problem
15:34.39Zeeekcome on over to #voip-users-conference for the call today in 1/2 hour. Right now we're talking about DECT/SIP phones though
15:35.10*** join/#asterisk mukudo (n=jgreen@58.251.97.17)
15:35.23ddunavanthow do we list voip providers in the US?
15:35.32ddunavanti know there is a way to do it...
15:35.33Kobazgoogle
15:35.40ddunavantok
15:36.45Faustovhi, is there any setting to enable indications.conf, other than having that config file in the confdir?
15:37.32*** join/#asterisk sp00kz (i=ilubj00@our.government.is.in.the.dark.bz)
15:38.19*** join/#asterisk sacitec (n=tobi@201.144.211.82)
15:39.59sacitecgood morning, i'm working with asterisk 1.2.x and Aastra ip phones 9133i with autoanswer. Now i'm looking for a polycom solutions in audioconference ip phone models (Soundpoint 4000)
15:40.28sacitecdoes anyone has tried to make autoanswer on polycom soundpoint 4000/6000 ?
15:40.28*** join/#asterisk legis (n=legis@unaffiliated/legis)
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15:48.28*** join/#asterisk kannan (n=admin@121.243.115.129)
15:48.52kannanhello, back with a fresh new troubles
15:50.17banzaikais there a way to handle invalid extension/phone numbers dialed ?
15:50.29banzaikasomething like DIALSTATUS ?
15:51.16*** join/#asterisk oej (n=olle@ns.webway.se)
15:52.19maqrbtw, why's it called Comedian Mail? did someone think that would be funny?
15:53.17*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
15:53.53banzaikavery
15:54.19NuggetJust be glad it's not called DAMHI for Digium Answering Machine Human Interface of something dorky like that  ;)
15:54.26kannanIt is doing comedy when i try to attach vm to email
15:54.50Zeeekhttp://x2z.eu VoIP Users Conference begins now
15:54.54kannansendmail is fine, but i think smtp has to be run on the localhost itself?
15:55.54Kobazmaqr: and the default voices sound very condescending
15:55.58*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
15:56.01danlock2psh... i wish i could get a call to even come through
15:56.21Kobazthere should be a "i'm better than you" track added to every prompt of the default sounds
15:56.53Kobazthe person at... 4....3...2....1... is unavailable.... because i'm better than you
15:58.02*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584529.dsl.bell.ca)
15:59.16lmadsentzanger: ping?
15:59.21tzangerpong
15:59.43maqrKobaz: yeah, i'm going to have to learn how to change those
15:59.47lmadsensee privmsg
15:59.49Nugget1 packets transmitted, 1 packets receive, 0% IRC loss
15:59.56Kobazmaqr: see vector voice
16:02.36maqrKobaz: 'voice vector'?
16:03.38Kobazhttp://voicevector.com/
16:03.40Kobazyeah that's it
16:04.00maqrKobaz: i'll probably do my own, i just have to learn more about ulaw so i can optimize the sound files for it
16:04.06Kobazwell
16:04.10Kobazthe chick they have sounds really nice
16:04.23maqrlol
16:04.23Kobazhard to beat
16:04.29maqrwhat a good business
16:05.01maqrshould hire sexy sounding women and men with british accents and start a asterisk recording business
16:05.22lmadsenI'm not female or british.. I'm damn sexy
16:05.27maqrhmm
16:05.36lmadsendon't listen to whatever tzanger says
16:12.09*** join/#asterisk oej (n=olle@ns.webway.se)
16:18.14coppicewhat about us men with sexy sounding british accents?
16:19.03JenniferAkemiswoon
16:19.33*** join/#asterisk af_ (n=getsmart@88-149-230-31.dynamic.ngi.it)
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16:21.07mking_cdhas anyone else experienced a problem with chanspy which causes the audio to be out-of-sync after the spied-on caller activates hold?
16:25.04*** part/#asterisk jivco (n=jivco@85.187.217.6)
16:25.11*** join/#asterisk smartlu (n=seblu@fw.sj.tdf-pmm.net)
16:25.16smartluhello
16:26.07smartlui have a problem on my snom phone and my new asterisk installation with hints and monitoring extension
16:26.21smartluasterisk is 1.4.20.1
16:26.33smartluand snom fw is 7.1.30
16:27.03*** join/#asterisk km2 (n=x@c-24-23-252-175.hsd1.ca.comcast.net)
16:27.16esaymhow do you install asterisk?  is it "./configure" "make menuselect" "make" "make install" or "./configure" "make menuselect" "make install"?
16:27.31esaymI guess the question is do you have to run make after you do a make menuselect or not?
16:27.33smartlufirst
16:27.37*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
16:27.56smartluto be true, i have run make -j8
16:28.03smartluafter make menuselect
16:28.31smartluthis can be revelent ?
16:28.52esaymok thanks
16:31.21smartlumy snom key have this config fkey7: dest <sip:it@192.168.3.253;user=phone>
16:32.04smartluand my asterisk
16:32.07smartlu[default]
16:32.07smartluexten => 1111,hint,SIP/test1
16:32.07smartluexten => 1111,1,Dial(SIP/test1)
16:32.07smartluexten => 2222,hint,SIP/test2
16:32.07smartluexten => 2222,1,Dial(SIP/test2)
16:32.08smartluexten => it,hint,SIP/test1&SIP/test2
16:36.07*** join/#asterisk isamar (n=isamar@200.254.219.17)
16:36.09isamarhi folks
16:36.34isamarneed help with SET(CDR(anyfield)
16:37.22isamarI've created a new field in cdr table but cannot set with SET(CDR()) :-(
16:43.36lmadsenisamar: are you using cdr_adaptive_odbc?
16:43.51lmadsenisamar: you can't just Set(CDR(anyfield)=foo) in the DB without it
16:44.21isamarlmadsen: it doesn't work
16:44.36isamarlmadsen: I am using cdr_mysql from asterisk-addons-1.4.5
16:44.43*** join/#asterisk Dabba (n=d@ipv6.mfnx.ip6net.net)
16:44.45lmadsenthat isn't cdr_adaptive_odbc
16:45.04isamarlmadsen: ok.. got you
16:45.35lmadsencdr_adaptive_odbc is in 1.6.0-beta, or you can use the backport from http://svncommunity.digium.com/view/tilghman/branches/1.4/
16:45.58lmadsenthat's the purpose of that module -- is to allow you to set custom fields in the CDR table -- otherwise, you can't
16:46.15lmadsenuse ODBC for connecting to the DB -- it's must more supported
16:46.22lmadsengoes back to work
16:46.27isamarlmadsen: okey dokey
16:46.33isamarlmadsen: thanks dude
16:47.10*** join/#asterisk Trapa (n=no@207.230.238.94)
16:47.24Dabbaanyone got any idea why two identical linksys phones spa-941's can only call between each other in one direction ? one peer can call another but not the other way round , three days of fiddling!
16:48.06Dabbaboth can call pstn via the E1 and e1>sip is ok
16:48.57*** join/#asterisk LuisTorres (n=chatzill@bl6-192-3.dsl.telepac.pt)
16:51.06smartlunobody use snom phone with this kind of setting ?
16:52.51isamarDabba: seems to be route problem in your extensions.conf or some firewall issue
16:53.04isamarsmartlu: which setting?
16:54.07Dabbaisamar, not a route issue as changing the sip device to a different brand cures the problem
16:54.17TrapaWould anyone be able to look at this log and tell me if there's anything in it that would indicate a problem such that would result in calls getting dropped when the asterisk server has been up for 8 hours http://pastebin.com/d55f90534
16:54.32*** join/#asterisk jmardonesk (n=jmardone@236-166-18.adsl.din.tie.cl)
16:55.02Dabbaall sip devices on same firmware and every setting in sip.conf same and in phone guis
16:55.48*** join/#asterisk beek (n=klinebl@65.211.106.242)
16:56.27isamarDabba: you need to give a look on the sip chat to see what's happenning..
16:56.55jmardoneskhi all, I have a litle question, can make actions in the dialplan after hangup, i.e. I make a call in an analog line, then the user in the otrer side hang up, can make an post action like obtain the call duration o the start and end of the call?
16:57.07[TK]D-FenderDabba: pastebin the CLI output of the failed attempt with SIP debug enabled
16:57.08Dabbahas done sip debug on both peers but not v helpfuleg
16:57.09[TK]D-Fender~pb
16:57.10jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
16:57.20[TK]D-Fender^^^^^^^^^^^^^
16:57.20*** join/#asterisk mmurdock (n=chatzill@mail.kimballequipment.com)
16:57.31*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
16:57.32Dabbawill do a paste debug :-)
16:57.38[TK]D-Fenderjmardonesk: to do what withe xactly?
16:57.41*** join/#asterisk SteveTotaro (n=Administ@pool-151-196-246-236.balt.east.verizon.net)
16:59.34*** join/#asterisk dlynes (n=chatzill@S01060016b68219f1.vs.shawcable.net)
16:59.40dlynesI was told that there was a version of linux called 'ATC Linux', or 'ATCLinux' that had a kernel optimised for a specific Intel chipset to run Asterisk more efficiently...would anyone happen to have a pointer to it?
17:00.18jmardoneskI was thinking in store the start/end date-time of the call, and the dialed numer in a database using dialplan and func_odbc
17:01.48smartluisamar: setting which i have posted above
17:02.06smartluisamar: a snom phone which monitor a sip extension
17:02.54[TK]D-Fenderjmardonesk: Ever heard of this wonderful thing known as CDR?
17:03.51dlynesdidn't 1.4.20 just come out?  and now 1.4.20.1 is out?  another iax2 security fix?
17:04.09Qwellno
17:04.15[TK]D-Fenderdlynes: * 1.4.8.6.7.5.3.0.9!
17:04.25[TK]D-Fenderdlynes: Get yours now!
17:04.36Kobaz1.4.3.14.1.5.9
17:04.43jmardonesk[TK]D-Fender, I dont know CDR, Im realy new in this, (I work with asterisk 3 years ago... but never see CDR)
17:04.50dlynes[TK]D-Fender: 1.4.20.3.1415627
17:04.50drumkillait's a silly console cosmetic fix
17:04.52[TK]D-FenderKobaz: mmmm now I'm hungry again...
17:05.03[TK]D-Fenderdlynes: You're a little off...
17:05.14dlynes[TK]D-Fender: rounding error
17:05.36[TK]D-Fenderdrumkilla: Oil of Delay ;)
17:05.44Kobazquick quick, who can recite pi to 100 digits
17:05.59[TK]D-Fender"pi to 100 digits"
17:06.02[TK]D-Fenderwins
17:06.13*** join/#asterisk eXistenZ (i=pectic@unaffiliated/existenz)
17:06.14Qwell~pi
17:06.15jboti guess pi is 3.141592653589793238462643383279502884197169399375105820974944592307816406286208998628034825342117067982148086513282306647093844609550582231725359408128481117450284102701938521105559644622948954930381964428810975665933446128475648233786783165271201909145648566923460348610454326648213393607260249141273724587006606315588174881520920962829254091
17:06.15Kobazheh
17:06.19Qwellwins
17:06.35Dabba[TK]D-Fender: i see a few Retransmitting #1 on the INVITES ?
17:07.00dlyneshrm...I guess 1.4.20.1 is so important, they don't even say why it was released on asterisk.org :)
17:07.02[TK]D-FenderDabba: Do You?  I see nothing... where's the pastebin?
17:07.23Kobazdabba do
17:07.44Dabbahas sanitising to do
17:10.39[TK]D-FenderHCL for those tough to get out stains...
17:10.46[TK]D-Fender~h2so4
17:10.47jbot[~H2SO4] "John was here but is no more, for what he thought was H2O was H2SO4"
17:10.49[TK]D-Fender:D
17:10.53[TK]D-FenderStill there, heh
17:15.21TrapaCould somone review http://pastebin.com/d55f90534 and see if they can find any reason why we're getting calls disconnecting after the servers been up for a few hours?
17:15.34*** join/#asterisk rattler_ (n=misha@finly.sats.volia.net)
17:20.24Dabba[TK]D-Fender: http://pastebin.ca/1027211
17:21.37*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
17:22.52SteveTotaro1.4.20.1 was released to remove the g
17:22.54SteveTotaroGPL
17:23.01drumkillablinks
17:23.20SteveTotaro:-D
17:24.50[TK]D-FenderDabba: pastebin your sip.conf masking only passwords
17:27.26*** join/#asterisk darmock (n=root@c-98-211-225-216.hsd1.fl.comcast.net)
17:29.27Dabba[TK]D-Fender: http://pastebin.ca/1027220
17:31.18[TK]D-FenderDabba: What is RS, and whats the networking between them?
17:32.12Dabbaboth rs and pmr are linksys spa-941 with public ip's and on same lan
17:33.10Dabbaboth devices running same firmware and all gui settings on the endpoints are identical except username / pass
17:34.51[TK]D-FenderDabba: the retransmit is typically a network issue, be it firewall, or NAT.
17:35.14*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
17:36.00Dabbamad isnt it, phones not behind nat, in same switch, real ip's /25 mask pbx in same lan real ip etc etc
17:36.02Nuggethttp://nugget.livejournal.com/131726.html  <-- I need one of these
17:36.38QwellNugget: you know you can get custom stamps made pretty cheap?
17:36.45Nuggetyeah
17:36.53*** part/#asterisk mking_cd (n=mking_cd@pool-72-78-183-123.phlapa.east.verizon.net)
17:38.20*** join/#asterisk nny_2 (n=Scott_My@64.203.239.83)
17:39.12nny_2isn't sip show peers still the way to show sip peers? Or have our benevolent gods changed the syntax?
17:40.19nny_2nm bad config file, module isnt loaded
17:42.01*** join/#asterisk shinao1 (n=shinao1@smtp.gtbplc.com)
17:46.53*** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net)
17:47.39anonymouz666what could case ECHO between two ATAs (SIP-only)? PAP2-A calls PAP2-B. A hears echo. PAP2-A calls PSTN no echo at all. PAP2-A calls a Hardphone no echo at all.
17:48.56anonymouz666amazing.
17:50.15anonymouz666another ATA in PAP2-A position still produces echo.
17:50.22Dabbahaving just purchased the TE121andEC i wonder if digium will assist with this problem i have
17:50.53anonymouz666Dabba: why don't you call the Digium support?
17:50.57drumkillayes, please contact support@digium.com
17:51.03drumkillaor call
17:53.16anonymouz666just to finish the caes, if PAP2-A calls PAP2-C. A hears echo.
17:53.47*** join/#asterisk golumn (n=acxty@201.220.132.138)
17:55.16golumnHi guys, I want to make a Dial outside asterisk. My context is 5018989 in sip.conf so I am trying this. Dial(SIP/5018989/98781232) but it tell me Received response: "Forbidden" from .... I make a test conecting the line with xlite and it can dial outside
17:55.54smartluthanks for your help...
17:57.38plikgolumn: DIal(SIP/98781232@5018989)  number@context  should work
17:58.40golumnplik the same
17:58.43golumnresult
18:00.16Dabbadoh
18:01.03Dabbaas my issue is not digium hardware related, i need to buy consulting time, but the sales lady transferred me back even though i asked to buy some, lol
18:01.07Dabbaoh well
18:02.37golumndoes someone have that same issue before?
18:08.04*** join/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net)
18:08.34*** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
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18:11.31*** join/#asterisk davidgonzalezh (n=dgonzale@190.26.166.229)
18:11.46davidgonzalezhHi frinds from asterisk comunity
18:15.07JenniferAkemihi davidgonzalezh
18:15.43Qwelljameswf-home: 3 now
18:16.27jameswf-homequa?
18:16.31Qwellnothing
18:18.48*** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net)
18:25.16davidgonzalezhWwell guys very pleased to be on this channel
18:25.26davidgonzalezhit was long time I used IRC chat.
18:26.45davidgonzalezhI'd like to tell ypou that I'm been experimenting with Asterisk for the last yr and it's been great, ya know I hated telephony and lal that but * has opened my eyes to a whole new world
18:27.36danlock2... ok
18:29.07jameswf-homestarts humming the little murmaid song.... a whole new world...
18:30.14davidgonzalezhyeah rite the lil mermaid you don't know a thing
18:30.23davidgonzalezhthat son's from Aladdin
18:30.50jameswf-homeat least I had a disney movie,,,, arent they all the same
18:31.12cpmIt is a most elusive fish!
18:31.33davidgonzalezhAnyway this is the normal behavior I expect from guys that join these channels and you get yell at if you ask a question that's been answered a thousand times.
18:31.55davidgonzalezhanyway I'm just offering my help and experience with asterisk tto all o' those that may need it.
18:33.21JenniferAkemidavidgonzalezh: do you have any experience with making asterisk highly available? (through clustering etc)
18:35.42*** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net)
18:35.50davidgonzalezhJenniferAkemi: well we made a project on the company I worked for and it was quite big, we used PostgreSQL and ODBC to cluster all of the config files we used * realtime feature and it went quite good.
18:36.05davidgonzalezhther's load balancing between the servers and all.
18:36.13JenniferAkemiwhat did you use do do the load balancing?
18:36.34QwellJenniferAkemi: should look into dundi
18:36.57davidgonzalezhbalance I guess it was.
18:36.58JenniferAkemiQwell: yeah i have that on my list for the outbound load balancing
18:37.24JenniferAkemii'm also trying to figure out what we'll do to make the registration redundant
18:37.30JenniferAkemii was looking at using lvs and heartbeat
18:37.32Qwellfor incoming, a lot of people seem to use SER, or even just a load balancer router box
18:37.36davidgonzalezhdundi should be good, I never used it really I'd like to tho.
18:37.53*** join/#asterisk dkwiebe (n=darren@h66-112-187-16.mcsnet.ca)
18:38.54Qwell(hardware router, that is)
18:38.59JenniferAkemianother thing i'm worried about is if one of the asterisk box goes down or freezes up or something, what happens to my PRI. in order for it to work right it will have to take the PRI out of service, or else the PSTN will still send calls to that asterisk box
18:39.26Qwellor T1 failover boxes
18:39.38JenniferAkemiwell i'm not worried about hte T1 failing
18:39.47JenniferAkemibecause it's coming from a switch that is under my control
18:39.49davidgonzalezhAw tehre's no way to solve that but to have two PRI cards for redundancy..
18:39.57km2JenniferAkemi, you could get a trunk bypass switch, though i've only heard of these, never seen one
18:39.58JenniferAkemiif the T1 fails that's good - i'll just send it to the next asterisk box
18:40.02QwellT1 > magic box >> asterisk/asterisk
18:40.04Qwelllike that
18:40.14JenniferAkemii can have like 60 T1's
18:40.25JenniferAkemiand if one goes down, it can just go to the next one
18:40.43JenniferAkemithe issue is that the T1 has to go down if the asterisk box isn't ready to take the call that will show up on it
18:40.54JenniferAkemii'm worried about the sip stack freezing or something
18:41.03davidgonzalezhhmmmm
18:41.19JenniferAkemiI seem to remember i've seen random people asking about or other things in my various web meanderings
18:41.19gr0mitJenniferAkemi, that would _never happen ;-)
18:41.22JenniferAkemiok
18:42.00KobazJenniferAkemi: western telematic makes an rj48 8 conductor A/B switch
18:42.11JenniferAkemiwhat's that for
18:42.19Qwellto switch the destination of the T
18:42.28JenniferAkemihow does it know when to switch it
18:42.35Kobazyou tell it when to switch it
18:42.41Kobazit has a telnet interface
18:42.41Qwellif it's an A/B switch, it's a button you'd press
18:42.51JenniferAkemiwhen is that useful
18:42.53Kobazit can also switch on current loss
18:42.54Qwellrarely
18:42.57JenniferAkemihehe
18:43.07KobazQwell: we use them all over here
18:43.09Qwellif you're talking HA, you want 0 lost calls.
18:43.11JenniferAkemii'm not worried about the T1 going down
18:43.15Qwellso it would be automatic and immediate
18:43.20Qwellwould need to be*
18:43.29JenniferAkemiif the T1 is down, the PSTN calls won't go to it, they'll go to the next PRI in the hunt group
18:43.50KobazJenniferAkemi: we don't use it for a t1 dieing, we use it to switch over a single t1 to a failover box
18:44.14Kobazie: we need to change out hardware or bring the box down
18:44.16JenniferAkemiQwell is right though, I don't want any interruption of service at all
18:44.22gr0mitJenniferAkemi, what is the application you are considering?
18:44.33JenniferAkemiresidential voip
18:44.40Qwell5 9's?
18:44.53JenniferAkemior as close as we can get
18:45.06JenniferAkemiprobably going to be marketed as a "second line"
18:45.10JenniferAkemibut it can't suck :)
18:45.19gr0miti think you probably want a telco-grade switch then
18:45.25gr0mitif you want it scalable.
18:45.39gr0mitAsterisk is great but scalability is a bit 'meh'
18:45.41JenniferAkemiit's your opinion asterisk won't do the job gr0mit?
18:45.51gr0mithow many subscribers?
18:46.06JenniferAkemia few thousand
18:46.10drumkillai use asterisk for 1 billion subscriberes
18:46.16drumkillabut i can't spell
18:46.20JenniferAkemiheh
18:46.27JenniferAkemiwhat is the application drumkilla?
18:46.28jjshoeJenniferAkemi we have a magic t1 box switcher, it detects when the box goes down
18:46.39QwellI'd say Mike proved that Asterisk can most certainly handle that type of volume earlier
18:46.42jjshoeJenniferAkemi but it's here to try and get us to market it, i don't htink we've ever used it
18:46.57JenniferAkemijjshoe: heh
18:47.00Kobazjjshoe: what do you use?
18:47.12*** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca)
18:47.15JenniferAkemii don't think i need a magic t1 box switcher at all
18:47.20jjshoeKobaz if our pri's go down we automatically roll to a voip provider.
18:47.25adeelJenniferAkemi, a lot of people use a hybrid setup to do what you're looking for...typically handle the sip registrations through openser and then let * handle the media transcoding/voicemail/etc
18:47.36*** join/#asterisk sbingner (n=john@pdpc/supporter/sustaining/sbingner)
18:47.46gr0mitJenniferAkemi, I really don;t know if it can handle thousands of subs.  most people use SER in front to load balance
18:47.48JenniferAkemiyeah i've seen a lot of talking about using openser to handle sip registrations. why use openser for that over asterisk though?
18:47.50jjshoeKobaz we have the ability to directly point our number on the pri's to the voip. <3 our telco.
18:48.06gr0mitand use asterisk as a glorified media gateway
18:48.09adeelJenniferAkemi, openser's implementation can scale better than *'s
18:48.17JenniferAkemiwhat if you just add asterisk boxes
18:48.17QwellJenniferAkemi: SER is a SIP proxy.  It is not a PBX, so it doesn't have to handle any of that type of thing
18:48.19JenniferAkemito scale
18:48.20Kobazjjshoe: ah
18:48.31JenniferAkemihow many registrations can asterisk handle
18:48.34gr0mitadding asterisk boxes is problematic
18:48.34QwellJenniferAkemi: that's perfectly acceptable, as long as you have some way to get the packets between the boxes
18:48.56JenniferAkemidedicated gigabit ethernet apart from the voice path?
18:49.18gr0mitbut even then you are not really load sharing
18:49.35JenniferAkemii was hoping one asterisk box could handle registrations, and store them in a realtime database
18:49.36gr0mitbecause all your traffic then has to go through 2 or more boxes
18:49.46jjshoehonestly? making money by running a reseidential voip service is almost impossible
18:49.48JenniferAkemiincoming calls come in through multiple pris
18:50.08gr0mitresidentail voip is a mugs game.
18:50.11JenniferAkemiwhich go to multiple asterisk boxes which look up the ip's of the registered sip things
18:50.17JenniferAkemiwhy gr0mit. what is the roadblock
18:50.18jjshoeat $10 a month it takes one 5 minute phone call from the customer before you've spent the $10 they paid you that month
18:50.20gr0mitcustomers expect perfection with zero cost
18:50.25QwellJenniferAkemi: that's what most people use SER for
18:50.27*** join/#asterisk kannan (n=kann@123.201.60.110)
18:50.35Qwellmost?  many?  some?  whatever
18:50.38JenniferAkemii'm just wondering why SER instead of asterisk though?
18:50.47QwellSER is only a SIP proxy
18:50.47gr0mitnot instead of, as well as
18:51.16JenniferAkemii understand they use asterisk for voicemail and for the voice path etc, but SER for the registration, but what i dont understand is why
18:51.21kannanhello, i am not able to attach vm to email, i get in /var/log/maillog , connection refused at 127.0.0.1
18:51.44gr0mitJenniferAkemi, the problem is load
18:51.44JenniferAkemijjshoe: i guess we have an advantage
18:52.07kannanwhat about server loads, isnt SER supposed to be very much efficent? also it can switch TCP SIP?
18:52.09JenniferAkemijjshoe: we have pstn minutes available to us an incredibly cheap rates.
18:52.31adeelJenniferAkemi, from my readings, the max number of sip registrations i've seen on an * box is around 4000-5000 (i may be wrong) but i've seen SER boxes (similar in specs or worse) that can handle 10,000-20,000 registrations
18:53.23gr0mitand once a client has registered on box A, you need to know that client z is registered on box A
18:53.27adeelkannan, you haven't setup your postfix/sendmail/mail relay properly
18:53.46*** join/#asterisk banzaika (n=banzaika@rrcs-208-105-66-210.nyc.biz.rr.com)
18:53.48JenniferAkemiusing realtime i was under the impression it didn't matter which box the client regisered on.
18:54.16gr0mitso if a call comes in for client z on an E1 on box B, then box B has to send it to Box A over IAX or SIP
18:54.22*** part/#asterisk drzed (n=drzed@synflood.homelinux.org)
18:54.31QwellJenniferAkemi: to be honest, I wouldn't use realtime in a setup like that
18:54.47JenniferAkemican't box b just look it up in the sip table in realtime and send it to the ip
18:54.58JenniferAkemiQwell: do you have a reason?
18:55.10KobazJenniferAkemi: asterisk doesn't work like that
18:55.12gr0mitJenniferAkemi, have you used Asterisk yet ?
18:55.21Qwellit adds several extra layers of potential failure
18:55.23JenniferAkeminot in the scenario i'm describimg :
18:55.26JenniferAkemi:)
18:55.32JenniferAkemii am in the middle of setting it up though
18:55.38Qwellrequires extra database servers, etc, etc
18:55.42gr0mitwell,
18:55.47JenniferAkemii've  got one asterisk box using realtime attaching to 2 database servicers which are mirrored master master
18:55.57JenniferAkemii'm just setting up the other
18:56.01Strom_Mservicers? sigh
18:56.06Kobazrealtime doesn't gain you much of anything
18:56.07JenniferAkemiand am going to attemp to call from one to the other
18:56.08kannanadeel : how to install the sendmail?
18:56.15JenniferAkemiservers sorry typo
18:56.23gr0mitrealtime only helps with automateing adding sip clients
18:56.34kannan/usr/sbin/sendmail -d0 < /dev/null | grep -i version gives me the version, and the X package manager shows sendmail is installed
18:56.35Kobazyou can do an insert into a db rather than generating config files, that's about it
18:56.35gr0mitdoes not help you scale an operation.
18:56.46*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:56.52JenniferAkemithe thing i've liked about realtime is that there is only one config for allthe boxes
18:56.56JenniferAkemithey all access it
18:57.08QwellJenniferAkemi: yeah, but you could do that with rsync and the likes
18:57.11gr0mitand you can achieve the same with a big file and nfs export to multiple asterisk boxes
18:57.12Kobazyou can have a master configuration server that rsyncs your configs to your dest
18:57.15Kobazyeah exactly
18:57.27JenniferAkemihm.
18:57.47adeelJenniferAkemi, i don't mean to be rude, but if it was as simple as you expect it to be, everyone would be doing it
18:57.58JenniferAkemicoming from a non linux background i am at a disadvantage as I didn't even know what rsync was
18:58.09jjshoeif you have super cheap pstn minutes available sell them to other voip provider.
18:58.12JenniferAkemiadeel: which? providing residential voip, or using realtime to provide HA
18:58.13jjshoeproviders.
18:58.15adeelkannan, yeah, that can take a lot of work, and the best way to get the answer is to google it...
18:58.26KobazJenniferAkemi: so you're comming from a non-linux background and you want to use asterisk to sell phone service to 1000+ people?
18:58.35KobazJenniferAkemi: i suggest you try and different business
18:58.39JenniferAkemijjshoe: once we get this voip stuff running that's another market to tap
18:58.43Kobazs/and/a
18:58.45gr0mitJenniferAkemi, I don't want to put you off
18:58.47JenniferAkemiwell i'm coming from a telco background
18:58.52gr0mitasterisk is GREAT
18:58.54gr0mitbut.....
18:59.14adeelJenniferAkemi, providing residential voip isn't too hard, but scaling it efficiently is the hard part...if you have 200 * boxes to service 10,000 clients, i don't think you'll be able to cover your expenses while keeping your prices low
18:59.17gr0mityou don't use an MD110 as an internaltional SS7 gateway,
18:59.33JenniferAkemiasterisk boxes are cheap though
18:59.34gr0mitand you don't use an AXE10 for a 100-seat pbx/call centre
18:59.35adeelJenniferAkemi, and using realtime * typically involves some custom app's and whatnot...
18:59.42JenniferAkemithe big cost is the digium card
18:59.43adeelJenniferAkemi, but electricity and connectivity isn't
18:59.52JenniferAkemiconnectivity is cheap
18:59.56KobazJenniferAkemi: and you don't want digium either
18:59.59JenniferAkemiif you already have it
19:00.04gr0mitJenniferAkemi, i dont thing anyone here is saying 'No'
19:00.14QwellKobaz: back that statement up with facts.
19:00.28gr0mitbut I think you should consider playing in a small-scale depoloyment before you scale up
19:00.31KobazQwell: i have a higher pile of dead digium cards than any other
19:00.34Strom_MQwell: watch these facts all be from five years ago
19:00.41JenniferAkemigr0mit which is what i'm doing right now
19:00.49JenniferAkemiobviously we'll open up for beta testing first
19:00.50gr0mitJenniferAkemi, fine!
19:00.56[TK]D-FenderQwell: fact : * isn't what most telcos would consider "stable", this relying on that to get to the PSTN = not so great
19:01.02Kobazi find sangoma much more solid
19:01.14gr0mituses Sangoma cards
19:01.25JenniferAkemiwe're going to get sangoma card to test
19:01.28adeelJenniferAkemi, my biggest cost right now is the bandwidth i need
19:01.33JenniferAkemiwe have a digium and a t100p clone right now
19:01.38JenniferAkemiadeel how much do you pay?
19:01.39gr0mitJenniferAkemi, which country is this for?
19:01.41JenniferAkemiadeel i was pricing that out today
19:01.42JenniferAkemiCanada
19:01.50QwellJenniferAkemi: avoid the clones
19:01.52gr0mitok
19:02.02gr0mitwell Sangoma are in Toronto
19:02.12kannanis x100p.com a clone?
19:02.13*** join/#asterisk A500mg (n=x@ACaen-156-1-2-126.w90-17.abo.wanadoo.fr)
19:02.17gr0mittheir support is great
19:02.20adeelJenniferAkemi, depends...i'm in california, and depending on where i go, i get different rates....t1 from 300 bucks to a t3 for just under 3,000 or so....could get into a carrier hotel, but the costs are still pretty high
19:02.27[TK]D-Fenderkannan: No, its just shit :)
19:02.35kannanoh ok
19:02.54JenniferAkemii figured i'd start with business dsl for beta testing
19:02.54[TK]D-FenderYou wanna run an ITSP, use serious back-end gear like AudioCodes gateways.
19:03.06gr0mitor Teles
19:03.07kannanhmm , is there a friendly GUI fror sendmail heh?
19:03.14gr0mitmy wholesaler uses Teles kit
19:03.23[TK]D-Fenderkannan: webmin
19:03.29JenniferAkemiyeah i just noticed sangoma is local
19:03.32jjshoeum hahaha
19:03.37jjshoeif you think the cards are expensive
19:03.41[TK]D-Fenderkannan: For as "frienldy" as that can be.  when in doubt, JFGI
19:03.42*** join/#asterisk grantm (n=grant@68.142.138.4)
19:03.43gr0mit'local' as in Canada, !
19:03.44jjshoeyou're in the wrong line of work
19:03.46adeelkannan, unless you need a FULL MTA, don't use sendmail...just setup SSMTP for  relaying
19:03.48kannan[TK]D-Fender : thanks
19:03.48*** part/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net)
19:03.50gr0mit(tis a big place!)
19:03.51JenniferAkemii'm about 30 minutes from markham
19:03.56gr0mitah ok
19:04.05kannanadeel : thanks , whats that? heh
19:04.07A500mgmmh
19:04.15A500mgI've a question
19:04.16kannani just need to attach the vm to email,
19:04.16adeelkannan, a simple mail relay
19:04.19*** join/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net)
19:04.24gr0mitare you with an ISP JenniferAkemi
19:04.24JenniferAkemijjshoe: the computers are cheaper than the cards. minutes are pracitally free
19:04.39JenniferAkemilong distance provider
19:04.41[TK]D-Fenderkannan: You shouldn't HAVE to configure anything for sendmail for that to work in most cases
19:04.42kannanwe have to replace the cmd for sendmail in voicemail.conf after building that?
19:04.44A500mgwhen I do a "core show channels" during a call is ringing
19:04.44gr0mitaaaah ok
19:04.53kannanor can we hust build and link to sendmail
19:04.55gr0mithence 'free' minutes ;-)
19:05.02JenniferAkemiwe have 20 ds3s around the country
19:05.05A500mgI've a "ring" and a "ringing"
19:05.07JenniferAkemiso yeah :)
19:05.18A500mgwhat's the difference between "ring" and "ringing" ?
19:05.25jjshoeJenniferAkemi that's it?
19:05.32[TK]D-Fenderkannan: jsut install * like normal, * just uses it to send the message.  shouldn't have anything to configure there unless your ISP forces you to do SMTP through their gateway
19:05.33kannan[TK]D-Fender : i thought so, it worked in other boxes automatically, now it is not working, dunno howto figure it at all
19:05.38gr0mitwell, iiwy I would look at getting some telco-grade media gateway
19:05.54JenniferAkemilike a cisco something?
19:05.59gr0mitnah
19:06.04gr0mitTeles
19:06.07[TK]D-FenderJenniferAkemi: AudioCodes Mediant series
19:06.09gr0mitor Audiocodes
19:06.10kannani am not getting any, any at all, messages at root@localhost
19:06.27jjshoeKobaz you're %100 correct on your recommdantion btw.
19:08.19A500mgI think "ring" is the caller and "ringing" the callee, for example:
19:08.20A500mgSIP/tech02-09210c70!context-tech02!701!1!Ringing!AppDial!(Outgoing Line)!701!!3!1!(None)
19:08.20A500mgSIP/tech01-091eca00!context-tech01!701!2!Ring!Dial!SIP/tech02|60|tT!700!!3!1!(None)
19:08.31A500mgtech01 is calling tech02
19:09.02JenniferAkemiwhat sorts of rates would you guys pay for voip minute
19:09.06JenniferAkemis
19:09.14Qwell2c/min is common
19:09.23JenniferAkemito us ?
19:09.32kannanlol
19:09.34Qwellus, canada, some places in europe, mexico
19:09.51JenniferAkemiwhat sort of quality?
19:10.03Qwelltoll?
19:10.04kannanoh to USA , i thought to JenniferAkemi
19:10.14JenniferAkemi:P
19:10.15Strom_Mfor 2c/minute, I expect toll quality
19:10.19QwellStrom_C: yeah..
19:10.21JenniferAkemiwhat's toll quality
19:10.26JenniferAkemiis that the best?
19:10.36Strom_MJenniferAkemi: i thought you said you came from a telco background :P
19:10.37delparnelLike a normal phone line
19:10.42JenniferAkemiyeah but we just buy the best
19:11.00Strom_Mtoll quality is the same quality you expect when placing a standard toll call over the PSTN
19:11.04JenniferAkemiok
19:11.07Qwellwell, what's the best?  surely you aren't offering 44khz stereo audio
19:11.18JenniferAkemii mean
19:11.44JenniferAkemiwhen a company says they have "gold rates" "platinum rates" or sorry to say what they sometimes call "voip rates" we only buy the platinum minutes
19:12.00JenniferAkemiit hasn't been worth the customer service nightmares to buy the cheapest
19:12.10Strom_MJenniferAkemi: uh, so exactly what kind of "telco background" do you have?
19:12.18jjshoeI'm beginning to think this is one very good troll
19:12.31JenniferAkemithe kind that you get from working at a telco for the last 10 years
19:12.35[TK]D-FenderJenniferAkemi: I use cubic zirconium minutes.  The cost much less but last just as long
19:12.43JenniferAkemia troll?
19:12.52delparnelbahahaha D-Fender
19:13.20Strom_MJenniferAkemi: so wait, how can you work at a telco for a decade and not know what "toll quality' means
19:13.22JenniferAkemii'm offended even though you made me laugh
19:13.28A500mglol d-fender
19:13.29jjshoeJenniferAkemi so what do you plan to do with 10 years of experience of mopping the floors of a telco for 10 years?
19:13.43jjshoeooo that was an awful sentence.
19:13.51delparnelyes, yes it was.
19:13.58[TK]D-FenderJenniferAkemi: pwned
19:14.08Qwellgirl + irc + troll.  impossible
19:14.26JenniferAkemitoll quality could mean what people commonly accept on a long distance call - as in a toll call, vs what one might expect on a local call - which in my opinion is a little different
19:14.41JenniferAkemiyou expect the person you call on the other side of the world to sound a little fuzzier than the person next door
19:14.42jjshoeQwell hahaha +10 points to you
19:14.51Qwellno, I meant, she really is a girl :p
19:14.54jjshoeJenniferAkemi no I don't.
19:15.00JenniferAkeminot anymore
19:15.04JenniferAkemibut historically people did
19:15.10delparnelI think for 2c a minute you're going to get toll quality or better if your connection is good.
19:15.11*** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca)
19:15.30JenniferAkemii'm not disputing your 2c/minute number for toll quality
19:15.38jjshoeQwell that would explain a lot
19:15.47QwellJenniferAkemi: this is true.  these days "toll" doesn't really apply
19:15.52delparnelSometimes you can get even better, depending on the volume of calls you do.
19:15.58JenniferAkemiare you being a little sexist here jjshoe?
19:16.06QwellJenniferAkemi: ignore him
19:16.14JenniferAkemiwhat about for inbound minutes on a DID
19:16.15delparnelI have received as low as $0.015/min
19:16.15JenniferAkemisame rate?
19:16.27JenniferAkemithanks Qwell.
19:16.28QwellJenniferAkemi: pretty much
19:17.17JenniferAkemiit's true that the telco world is a little overrun by men.
19:17.51JenniferAkemii've only ever talked to one or two other women at other telcos
19:17.57delparnelJenniferAkemi: Some ITSP's offer unlimited incoming for a fixed rate per DID.. something like $5.95 in the USA or $6.95 in Canada
19:18.17Qwelldelparnel: those aren't actually unlimited though
19:18.21Qwellit's "unlimited*"
19:18.36delparnelOr alternatively $0.99/DID + $0.01/min
19:18.38[TK]D-Fender(tm)(r)(oac)(apr)
19:18.40Qwell(where "unlimited" means x minutes * xc/min)
19:18.57A500mgwhat's the difference between "ring" and "ringing" when I do "core show channels" ?
19:19.03QwellA500mg: incoming vs outgoing
19:19.04delparnelIt usually ends up being more profitable though if you do a high volume.
19:19.07QwellI don't know which is which though
19:19.29A500mgmmh
19:19.35A500mgI think this also
19:19.54A500mgbut i've test an inbound call on misdn
19:20.23A500mgand I've ringing for both (misdn line and sip/phone)
19:20.30A500mgmaybe a little difference with misdn ?
19:20.33jjshoeany res. voip company that gets big enough is just going to get slapped around by the major telco's like vonage did anywyas. it's a stupid market to enter imho.
19:20.42A500mgI will test with zap channel for see ...
19:20.51*** join/#asterisk Hydrant (n=aj@CPE0011950c737b-CM0012c90d1420.cpe.net.cable.rogers.com)
19:20.52Qwellvonage was stupid on their own
19:21.20Qwellyou don't spend $150 per customer in advertising for a customer you're only going to have for 3 months
19:21.22Strom_Mis that barrel of monkeys still around?
19:21.35*** join/#asterisk M1s3ry (n=M1s3ry@216.207.245.1)
19:21.41Strom_Mhey M1s3ry
19:21.51M1s3ryhowdy
19:22.16*** join/#asterisk axisys (i=iqbala@otaku.freeshell.ORG)
19:22.17JenniferAkemii don't get the big price difference between like vonage and vbuzzer
19:22.32JenniferAkemiones $20 a month and ones $2 a month
19:22.38JenniferAkemiand they seem to offer the same service pretty much
19:22.44Qwelllike I said - you don't spend $150 per customer :)
19:22.50A500mgmmh or this is inbound/outbound and not callee/caller, call is inbound on misdn and inbound also on the phone who is ringing
19:23.15M1s3ryif the quality and everything else is the same, I guess the difference would only be $18
19:23.23JenniferAkemiheh M1s3ry
19:23.24Strom_MI would like to strangle whoever thought up the word "callee"
19:23.30kannanyippee You have new mail in /var/mail/root
19:23.31Strom_Mit's "called party"
19:23.39Qwelland caller?
19:23.43Strom_M"calling party"
19:23.44Qwell(calling party)
19:23.48jjshoeStrom_M I'd agree.
19:23.50Qwellbut, are you okay with caller?
19:24.04kannanheh, after a thorough googling , i had to do a chmod a+x rc.sendmail for that one, heheh
19:24.04Strom_MQwell: moreso than I am with 'callee'
19:24.17Strom_Mbut i don't like it in that context
19:24.19davidgonzalezhwhich would be a good voip provider tthat goves me free 8000 calls
19:24.22JenniferAkemiwhy is it scln and scdn
19:24.25M1s3ryor what about "calling agent" and "agent called"?
19:24.35A500mgmmh
19:24.37kannanbut at least it gave me a start on MTA
19:24.40*** join/#asterisk angom (n=angom@201.170.65.143)
19:24.40A500mg3 test:
19:25.18A500mgSIP/phone1 place an internal call on SIP/phone2: SIP/phone1 is "ring" and SIP/phone2 is "ringing"
19:25.53A500mgSIP/phone1 place an external call on misdn/(number): SIP/phone1 is "ring" and misdn-e54ezf54 is "ringing"
19:26.00Hydrantanyone have experience using a USB phone?  I want to setup a USB phone with my laptop so that when I'm travelling and on wireless I can have my laptop phone on a ring group and I can get calls
19:26.36jjshoeHydrant never used one. I'm not sure I see the dificulty though. I'm sure things like x-lite might have support for it.
19:26.54A500mginbound call from external (isdn line), on SIP/phone1: misdn-ezf251ze5 is "ringing", SIP/phone1 is "ringing"
19:27.03Hydrantjjshoe: just used to the good ol' Linux days where the word "USB" made you cringe
19:27.07lmadsenwhat am I doing wrong with this STRPTIME() function?
19:27.08lmadsenhttp://www.pastebin.ca/1027321
19:27.12A500mgthe last example trouble me :)
19:27.17QwellHydrant: USB "phones" are just soundcards
19:27.42A500mgmisdn can be the "caller" but is "ringing"
19:27.48HydrantQwell: cool... so they should be fine with Linux then... I'll check to see what I can do
19:27.56lmadsennevermind
19:27.58lmadsenTYPO!
19:28.25A500mgif we think "misdn receive a call from external" and "SIP/phone1 receive also a call from misdn" , "ringing" for both is logic
19:28.28lmadsen${start_time} != ${start_date}
19:28.49A500mgI just need to understand :)
19:29.05jjshoeHydrant it still does, what os are you using?
19:29.22Hydrantjjshoe: Linux
19:29.26Hydrantjjshoe: 2.6.x
19:31.17jjshoeHydrant oh, best of luck.
19:31.26davidgonzalezhAw cool that'¡s agood list of Voip providers very comperhensive
19:31.42jjshoeHydrant why not use your existing sound card?
19:31.54Hydrantjjshoe: I just like the feeling of a phone
19:32.04Hydrantjjshoe: although I guess I could get a USB headset or something
19:32.12davidgonzalezhhehe, the old days of a phone.
19:32.21davidgonzalezhget over it use xlite
19:32.52Hydrantonly other nice thing is a ringer
19:33.03Hydrantalthough I can setup something on my screen or something else
19:33.07*** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net)
19:34.54HydrantI guess the best bet is to get a headset of some sort after all
19:35.03Hydrantresearching it seems to be less of a headache
19:37.07JenniferAkemii have a headset that i use that works well when i'm not at home. it's not usb though, it just has two jacks, one for the mic and one for the headphones
19:37.50*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
19:39.57HydrantJenniferAkemi: I think that's the way I'm gonna go
19:40.23HydrantI just have to figure out a way to setup something that will work with Asterisk, so that I can setup my laptop as a SIP device or something...
19:40.42HydrantMaybe setup something with openvpn for incoming calls, outgoing shouldn't be a problem
19:40.50JenniferAkemii just use it with xlite connected to asterisk as davidgonzalezh suggested.
19:42.12*** join/#asterisk niZon (n=niZon@tande.voinetworks.net)
19:43.02JenniferAkemiwell damn :( you guys were right that didn't work
19:43.36JenniferAkemii just setup the second * box sharing the sip_buddies table using realtime and registered to one and tried to call an account on the other one
19:48.33JenniferAkemimaybe i need to use DUNDI to make this part work too
19:52.15kannanhello, i just realize sendmail enables me to set the from address as anything i desire, isnt that a risky thing in general?
19:52.15*** join/#asterisk didz_ (n=hoje@201.19.207.130)
19:52.15didz_the counter of "Bipolar Viol" is increasing on zttool... anyone knows what it means?
19:52.30kannani mean there is no sanctity for a email adress?
19:52.56*** part/#asterisk rattler_ (n=misha@finly.sats.volia.net)
19:53.04kannani know this snt the rrom for it, just set me wondering a lot
19:54.49lanningnot really a sendmail issue.  it extends from SMTP
19:54.53*** join/#asterisk kai4711 (i=psybnc@h1395155.stratoserver.net)
19:55.19[TK]D-Fenderkannan: Seriously... get a clue.
19:55.42lanningyou can telnet to TCP port 25 (SMTP port) and send anything you want.
19:55.46Bananaskinhey has anyone got iaxmodem/hylafax working on a sip trunk would like to carry out a test of faxing over sip ?
19:55.55[TK]D-Fenderkannan: And you WONDER why your "friends" send you ads for Viagra.  Guess what, it isn't "them", they we spoofed
19:56.11*** join/#asterisk A500mg (n=x@ACaen-156-1-104-54.w90-17.abo.wanadoo.fr)
19:56.12A500mgraaaaaaaa
19:56.13[TK]D-FenderBananaskin: .... WGLWAT
19:56.21A500mgsorry disconnected and pseudo in use, grrrr
19:56.21Bananaskinhey [TK]D-Fender huh ?
19:56.23[TK]D-Fenderwere*
19:56.27[TK]D-Fender~wglwat
19:56.28jbot[wglwat] well, good luck with all that
19:56.30[TK]D-Fender:p
19:57.09[TK]D-FenderBananaskin: the send you say "SIP over internet", thats akin to saying "automatic fax transmission failure"
19:57.21*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:57.21[TK]D-FenderBananaskin: for faxing that is...
19:57.35A500mgQwell have you try to answer my question ? ("ring" and "ringing")
19:57.36Bananaskinlol, well tbh, I tested an outgoing last night to an actual fax wired to pstn and accidentally sent it over sip and it was as good if not better than the one I sent out over the zap channel
19:58.05[TK]D-FenderBananaskin: fax is digital... should have absolutely no impact
19:58.07Bananaskinironically the one over zaptel took longer as well
19:58.24*** join/#asterisk golumn (n=golumn@201.220.132.138)
19:58.30BananaskinDigital until it meets the telco
19:59.24golumnHi guys, I am getting handle_response_invite: Received response: "Forbidden" From "user"....... every time I try to make an outbound call
20:01.08Bananaskinseems there was a bit of jitter logged on the outbound over sip but nothing which affected the fax transmission
20:09.52*** join/#asterisk grantm (n=grant@66.29.180.194.static.utahbroadband.com)
20:18.41TrapaHas anyone here had major issues with XLITE?
20:18.43TrapaI'
20:18.50nny_2using it at home Trapa
20:19.07TrapaI'm getting a problem that the phone will get a half-ring and then it gets a hangup
20:19.24nny_2is this on the local network?
20:19.36TrapaYeah
20:20.05nny_2hmm not sure, have you tried Ekiga or other to confirm it isn
20:20.16nny_2't an issue with the dialplan?
20:20.41TrapaSo the calls come into the asterisk server and then go to the xlite phones which are local to the asterisk server
20:20.52nny_2sounds about right
20:20.54TrapaI don't know that it's NOT a issue with the dialplan .. I'm using queue's...
20:21.05Trapaand i thought i tested .. and when i test i can't re-create the problem
20:21.20TrapaAnd of course i have some agents .. who are giving me not very useful information about what is happening ...
20:21.57TrapaThis is a pastebin of the verbose output from asterisk.    This was from last night, until this morning ...
20:22.12TrapaThis morning the agents said that we needed to restart the server, and when we did it was resolved http://pastebin.com/d55f90534
20:22.28TrapaBut i don't see any errors in the log there. So i am pretty confused as to whats happening
20:23.32*** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net)
20:23.51nny_2yeah being able to not recreate the issue is a PIA eh?
20:24.05Trapayeah no kidding
20:24.10nny_2personally never had that issue unless asterisk suddenly thought the sip peer was gone
20:24.25TrapaI mean i kinda wonder if people are just hanging up
20:24.32nny_2ha
20:24.37nny_2check the CDR?
20:24.46TrapaCDR?
20:24.56*** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net)
20:25.00hmmhesayswhat up folks
20:25.20nny_2/var/log/asterisk/cdr-custom/Master.csv
20:25.51Trapadon't have that log
20:25.57nny_2that and the asterisk logs should at least tell you some more info about the past
20:25.58nny_2hmm?
20:26.07nny_2strange
20:26.20nny_2is on all my installs by default, without any extra input
20:26.52outtolunccheck in the cdr-csv (vice the cdr-custom you had him look in)
20:27.10nny_2yeah heh
20:27.58nny_2the whole "they could just be hanging up" thing is possible, although i hate guessing myself.. is it frequent or once in a while?
20:29.41*** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za)
20:29.54TrapaWe only get calls like once in a long while
20:31.38nny_2well the cdr records may help with tracking down who what etc, but (correct if i am wrong, peeps) i would think the event would show up as ANSWERED once your IVR or if an Answer() is in before it goes to an agent, so tracking it down may be a bit hard... alternatively, if the sip client was doing the hang-up vs the caller, I don't think it would show up differently in the cdr
20:31.53nny_2i am a bit green myself, so take all that with a grain of salt :)
20:37.57JenniferAkemidoes anyone know what the sip field fullcontact is for?
20:38.44*** join/#asterisk datachomper (n=russ@75.146.194.59)
20:39.08datachomperIs there a "dialtone" sound file? If I want to play a dialtone to a user, how would I normall go about doing it?
20:39.18Strom_Mplaytones()
20:39.51datachompergreat, thanks
20:43.00kannanbye all , c u tomorrow
20:43.32bbryantJenniferAkemi: it stores the contact header sent when a sip call is registered
20:45.07Ritzeriskis there a good gui to install for like doing a easy configuration
20:45.21Ritzeriskon top of the core asterisk already
20:46.31Trapanny_2 thanks for the help i'm just busy i'll be back in a momne
20:46.34JenniferAkemilooks like i could do a mysql lookup for fullcontact, and then send the call directly via sip instead of going through the original sip registration server
20:46.41JenniferAkemii guess that's too much overhead though
20:46.49nny_2Trapa: heh not sure if its help, but i try
20:47.18*** join/#asterisk ddunavant (n=David@75.145.240.14)
20:51.20TrapaSo ... My queue has several seconds of silence before it says anything to the customer .. Is there some way of changing that?
20:51.43TrapaI would like the customer to always hear somthing .. be that the "Your next in line" or the music on hold, or ringing or anything other than silence ..
20:53.23*** join/#asterisk war59312 (n=war59312@unaffiliated/war59312)
20:53.50*** join/#asterisk unstable (i=unstable@tor/regular/sid)
20:57.01*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:58.19JenniferAkemiso i'm beginning to understand that asterisk is more like a pbx and not a tandem switch
20:58.24JenniferAkemiit's good at the features
20:58.36*** join/#asterisk revengervn (n=test_tes@static-96-226-59-205.dllstx.dsl-w.verizon.net)
20:58.39JenniferAkemibut not so good at the rock solid huge volume just switching calls
20:58.45revengervnhi
20:58.45JenniferAkemiis that correct?
21:00.28unstableyes
21:00.55*** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)
21:00.56unstablerevengervn: what is your question?
21:01.37revengervndoes anyone know what extension MP3Player application support ?
21:01.37revengervncan it play .wma or .asf
21:01.51revengervn?
21:01.56revengervnanyone can help me?
21:02.56revengervnhi
21:02.59revengervnnice to see you here
21:03.04revengervnbecause
21:03.18revengervnI want asterisk to play streaming media
21:03.19Ritzeriskis there a simple way to configure a wildcard T1 i tried doing a ztcfg -vv
21:03.24revengervnso I wonder
21:03.25Ritzeriskbut it saw no channells
21:03.43revengervnif MP3 player can handle other extensions like .asf
21:03.58revengervnor wma
21:04.11*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
21:04.42Strom_MRitzerisk: did you set the appropriate settings in zaptel.conf?
21:05.18Ritzeriskso i would have to go there first to set it up as like 24 channel t1
21:05.27Strom_Mwell, yeah
21:05.38Strom_Mit doesnt just magically autoconfigure itself
21:05.55Ritzeriskohh haha geesh
21:06.48jameswf-homeour channel banks automagicly configure tham selves :)
21:06.52Ritzeriskit automatically did it for me in the elastix version so i was kinda huh
21:07.19Ritzeriskbut theres no asterisk gui on this so hopefully i can install a overlapping gui on top of the vicidial gui too
21:09.39*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
21:10.00TrapaI need some helps with queues .. anyone out there a queue guru?
21:10.18jjshoeTrapa I would recommend asking a real question
21:10.26*** join/#asterisk [hC] (n=hardcore@vpn.voxter.com)
21:10.28TrapaSo ... My queue has several seconds of silence before it says anything to the customer .. Is there some way of changing that?
21:10.46*** join/#asterisk l2cache (n=l2cache@117.178.101.97.cfl.res.rr.com)
21:11.39datachomperAGI Rx << EXEC Playtones dial
21:11.40Ritzeriskam i able to just do a yum -y update asterisk its currently at 1.2
21:11.44datachomper<PROTECTED>
21:11.59datachomperAny reason why it would execute playtones, but no actually generate any sound?
21:14.08*** join/#asterisk jmardonesk (n=jmardone@236-152-110.adsl.din.tie.cl)
21:15.58Yourname`Is there a good Canadian online store where we can buy VoIP products that anyone knows about? Preferably in Toronto?
21:16.52*** join/#asterisk Nasra (n=maxshipp@190.166.71.163)
21:17.01revengervnDoes anyone know PYTHON AGI framework except pyst?
21:17.52jmardoneskhi, all.. in a IVR system when I have a backgroung sound waiting for an extension number, the priority of this jump to the extension is always the lower as possible?
21:18.14*** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com)
21:24.36TrapaDoes anyone know what Spawn extension (gssphones, 604, 1) exited non-zero on 'Local/604@gssphones-7902,2' means
21:27.41*** join/#asterisk pikachu2000 (n=pikachu2@196-209-94-228-tpr-esr-2.dynamic.isadsl.co.za)
21:30.09*** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net)
21:30.23watchywhats the best way to pipe in FM radio to * for on hold music?
21:31.48*** join/#asterisk angom_w (n=angom@201.170.65.143)
21:31.52maqrthis whole "follow me" thing is very clever
21:32.04Strom_Mwatchy: bad idea unless you've got all the licensing and such already set up
21:33.54maqrStrom_M: i doubt anyone's ever got sued for that, i'd imagine you'd get c&d first anyway
21:34.18maqrwatchy: i'd pirate a higher quality stream though, if you're going that route... don't want commercials playing on your hold music
21:37.25*** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl)
21:38.42*** join/#asterisk fish-bulb (n=cstewart@216.207.245.1)
21:39.57maqrfish-bulb: i lol'd, good nick
21:39.57*** join/#asterisk jembo_ (n=mannje@217.114.52.2)
21:42.52*** join/#asterisk xipi (n=oliver@91-65-57-226-dynip.superkabel.de)
21:42.55xipihi
21:43.33*** part/#asterisk beek (n=klinebl@65.211.106.242)
21:44.55*** join/#asterisk mercutioviz (n=chatzill@66-17-33-47.biz.visl.arrival.net)
21:44.57xipii am looking for a way to allow users to call from a different account. example: user1 wants to use the line of user5. how can this be done?
21:45.55xipiif i am not mistaken, the setting would need to be made in the extensions.conf file
21:47.03*** join/#asterisk l2cache (n=l2cache@97.101.178.117)
21:47.45xipiis there any place, where i can look it up? like some tutorial?
21:48.17*** part/#asterisk Hydrant (n=aj@CPE0011950c737b-CM0012c90d1420.cpe.net.cable.rogers.com)
21:48.50*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
21:50.24watchymaqr: they want the local radio station
21:50.49watchymaqr: its a super small radio station + small company, aint no one gonna care
21:51.28watchyi just wanna know the best way to pipe in a FM station
21:51.34maqrwatchy: well, i'm not 100% sure how to do it, but any sound card should work as a source...
21:51.39maqryou'd have to hook up a radio, of course
21:51.42watchyyea
21:51.44maqrit might look a little silly
21:51.44maqrheh
21:51.49watchybut i'm wondering how to get * to see it
21:53.16watchycan * play streaming mp3s?
21:53.38riddleboxwatchy, yes
21:54.19spokraanyone have any wisdom on getting festival working
21:54.24watchyi guess put a soundcard in it, have some software stream from the soundcard to a mp3
21:54.35spokrathe howto on voip-info doesn;t work
21:54.47riddleboxwatchy, you could do that
21:54.52riddleboxspokra, which distro?
21:55.28spokra<PROTECTED>
21:55.28spokralenny/sid
21:55.42watchyriddle: is that the most sane/simplest way?
21:55.53riddleboxspokra, you could apt-get install festival
21:55.58spokrai did
21:56.15spokraand tried the change to scn file
21:56.26watchyanyone wanna write a XM radio internet streaming addon for *?
21:56.29watchyi'll pay you $25
21:56.29riddleboxwatchy, well the other was is to have a sound card with an input, then plug a cord from there to the headphone jack of the radio and use some app to play it
21:57.05riddleboxwatchy, there are xm and sirius apps for mythtv, I am sure you can find one if you search for it
21:57.24Qwellwatchy: can't just use madplay or whatever?
21:57.29riddleboxspokra, I use it on ubuntu and just apt-get install it and it works fine, whats the problem you have?
21:57.34watchydunno what madplay is.
21:57.39watchybut ill goog it
21:57.40jbeez.
21:57.40Qwellsurely it's just a "standard" stream.  does it require a proprietary player?
21:57.58watchyhmm, it streams from the web in their own player thing
21:58.09Qwellso then it's likely a standard stream
21:58.11watchyno idea what type of steam it is. i know it requires a l/p
21:58.20Qwellcheck the source
21:58.20spokrapbx*CLI>
21:58.20spokrax7f;    -- Attempting call on sip/14252810448@sip.broadvoice.com for 5555@local:2 (Retry 1)
21:58.20spokrax7f;  == Using SIP RTP CoS mark 5
21:58.20spokrax7f;       > Channel SIP/sip.broadvoice.com-0829e990 was answered.
21:58.20spokrax7f;    -- Executing [5555@local:2] Festival("SIP/sip.broadvoice.com-0829e990", "mary had a little lamb") in new stack
21:58.22spokrax7f;  == Parsing '/etc/asterisk/festival.conf': x7f;  == Found
21:58.24spokrax7f;  == Spawn extension (local, 5555, 2) exited non-zero on 'SIP/sip.broadvoice.com-0829e990'
21:58.32jbeezhi watchy
21:58.38watchysup mr jbeez
21:58.39riddleboxspokra, please pastebin it
21:58.46spokrasorry for the cut and paste
21:58.51Ritzeriskis there like a config tool for a zapata i cant seem to get the te122 T1 to work....
21:59.59*** join/#asterisk murdock_ut (n=chatzill@70.102.148.44)
22:00.21riddleboxRitzerisk, there is zttest, zttool
22:00.33maqrcould anyone recommend an SMS/MMS provider for the US?
22:01.07spokrahttp://pastebin.com/d15e4a39b
22:01.33Ritzeriskhmm
22:02.08*** join/#asterisk l2cache (n=l2cache@117.178.101.97.cfl.res.rr.com)
22:02.36watchyhmm i guess * don't do mp3s anymore.
22:02.41riddleboxspokra, so you dial 5555 and want it to say mary had a little lamb?
22:02.53riddleboxwatchy, I am using mp3 as moh
22:03.05watchywell it says 1.4 went to wav
22:03.19maqr~sms
22:03.20jbotextra, extra, read all about it, sms is Stop Making Sense, the greatest concert film ever, starring Talking Heads and directed by Jonathan Demme.  Send message to mobile phones via the internet for free.
22:03.21maqrtries the bot
22:03.35maqrnope, not useful
22:04.07Corydon76-digmaqr: T-Mobile ?
22:04.08spokrayep..
22:04.28Corydon76-digI'd go with them and a GSM modem
22:04.41spokrait tries to   be festival doesn't like the config.. for asterisk
22:04.54maqrCorydon76-dig: err, yeah, i actually am on t-mobile.... but i mean, isn't there an SMS gateway or something i could just use?
22:05.01maqrCorydon76-dig: it'd be mighty silly to wire a phone up to the pbx
22:05.11Corydon76-digmaqr: for what direction?
22:05.12*** join/#asterisk thepacmanfan (n=thepacma@12-218-140-147.client.mchsi.com)
22:05.21Corydon76-digSending?
22:05.31spokraare you in the us?
22:05.42Corydon76-dig2342342345@tmomail.net
22:06.00QwellI assume he means cheaper than $0.15c/min
22:06.09maqrCorydon76-dig: sending and receiving, ideally
22:06.13Qwellerm, -c
22:06.22Corydon76-digYou're going to need a GSM modem for receiving
22:06.23maqrCorydon76-dig: you'd think if someone sends an SMS to my DID, i could get it... somehow
22:06.33Corydon76-digNot in the US you can't
22:06.34spokrathere are cellular cards for PC.. that look like a modem.. and you can send sms thru them
22:06.36Qwellactually, chan_mobile supports SMS
22:06.43*** join/#asterisk fish-bulb (n=cstewart@216.207.245.1)
22:06.46Corydon76-digOh, right, that...
22:07.04maqrseriously? you need a physical gsm connection to do ti?
22:07.05maqr*it
22:07.18Qwellthere are some sms services I've seen people talk about on the asterisk-biz list
22:07.25Corydon76-digmaqr: you know what GSM modems are, don't you?
22:07.28Qwelldoing it yourself is non-trivial though
22:07.50Corydon76-digLittle box, hooks up via serial port
22:08.11maqrCorydon76-dig: well, that implies you get cell service in the data center, right?
22:08.12Corydon76-digTakes the same SIM card as a cell phone
22:08.29maqrCorydon76-dig: i'm on a VPS for my hosting anyway, i can't send them a modem to hook up to a serial port :p
22:08.44Corydon76-digmaqr: I've actually run one of those little things in an office, off an old Pentium 200MMX
22:08.53spokrawhy not send it via email to your cell phone?
22:09.02maqrhmm
22:09.10Corydon76-digmaqr: and then I ran openvpn between the two machines
22:09.18maqrCorydon76-dig: that's pretty clever
22:09.34maqrCorydon76-dig: how about if i wanted one of those short codes, like 6 digits or something? any idea how you get those?
22:09.43Corydon76-digand then I passed messages via HTTP and a CGI script
22:09.57*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
22:10.01Corydon76-digmaqr: Are you willing to pony up the big bucks?
22:10.02spokraI worked for ATT in the SMS group  good luck getting a short code
22:10.14Qwellspokra: oh?  do tell
22:10.20spokralet alone a 6 digit
22:10.23Corydon76-digmaqr: I don't think they let them go for under $100k/mo
22:10.24maqrCorydon76-dig: no, i'd much prefer to pony up the very tiny bucks
22:10.26maqrwtf?
22:10.28spokrawhat do you want to know
22:10.34Qwellspokra: why?
22:10.41maqrCorydon76-dig: that's insane
22:10.46Qwellwhy good luck?  why not 6-digit?
22:10.48Yourname`<PROTECTED>
22:11.01Corydon76-digmaqr: that's the amount of traffic that you need to generate in order to make it worth your while
22:11.02Qwellpolycom
22:11.23Yourname`Qwell: One with backlist display and no PoE?
22:11.23spokrathey don;t sell short codes   but it's a pain in the A$$ to get cerified to connect to ther enetwork..
22:11.25Corydon76-digmaqr: otherwise, you get yourself a phone number
22:11.31maqrCorydon76-dig: phone number it is!
22:11.48maqrCorydon76-dig: those scam services must make a lot of money selling jokes and ringtones though, to be able to afford that plus the tv commercials
22:11.52*** join/#asterisk macros73 (n=cs@c-24-131-77-140.hsd1.pa.comcast.net)
22:12.12Corydon76-digmaqr: that is precisely why it is so expensive
22:12.42Qwellbesides, there are only 1 million 6 digit codes (maximum)
22:12.47maqrCorydon76-dig: oh
22:13.17spokrathe only way into the cell companies is to be a good o boy..
22:14.33spokrathe smpp protocal is available on the internet.. good luck writing it!!
22:14.34Corydon76-digIt's a simple matter of putting the price up to the point where the demand drops to only want what's available
22:17.04maqrmakes sense to me
22:17.17maqrCorydon76-dig: still, you'd think that tmo or whoever could route sms to any DID
22:17.31Corydon76-digmaqr: they can
22:17.32maqrscammers can't be responsible for that not happening
22:18.20Corydon76-digbut there's a central SMS authority
22:18.34maqrthat should make it even easier then
22:23.26*** join/#asterisk bsaxon (n=bsaxon@71-8-14-108.dhcp.leds.al.charter.com)
22:26.08watchyaparently mplayer will stream xm
22:26.22watchyfrom console
22:29.14Qwellwatchy: if mplayer can, it's likely that madplay can (see musiconhold.conf)
22:30.08watchythanks
22:30.16*** join/#asterisk exothermc (n=miles@74.85.89.146)
22:30.20watchyi gotta log off but i'm gonna check that out
22:30.22watchyim heading home
22:30.39exothermcwhat is a recommended SIP client that does video for linux?
22:32.08Ritzeriskahhhh i did a modprobe and it said fatal error not found
22:32.23jjshoewhen shortcodes first came out they where roughly 10k a month..
22:32.56jjshoeor that was my experience anywho
22:33.01jjshoedunno now, it's been years.
22:33.47*** join/#asterisk bsaxon (n=bsaxon@71-8-14-108.dhcp.leds.al.charter.com)
22:34.07jjshoebut if you're just looking to send sms's just sign up to one of many available pay per message sms gateways, many even accept emails
22:34.29*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
22:35.33jjshoeI'm pretty sure most of them will handle responses as well
22:41.54*** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194)
22:43.34*** join/#asterisk bsdwarrior (n=mfahey@fahey.enter.net)
22:43.44bsdwarriorhow do you re register with your iax server?
22:43.49bsdwarriorI forget the command
22:46.11bsdwarrioriax2 reload ?
22:48.30bsdwarrioranyone ?
22:49.08seanbrightbsdwarrior: when you type 'iax2 reload' into the asterisk CLI and hit enter... what happens?
22:49.31bsdwarriorit loads the template, but I dont see it register
22:49.49seanbrightok, then iax2 reload isn't the command you want
22:49.54seanbrighthope that was helpful
22:49.56seanbright:)
22:50.09bsdwarriorI have the config in iax.conf
22:50.42drmessanoasterisk -rx reload
22:50.43drmessanoDone
22:50.52seanbrightservice asterisk restart
22:50.53seanbrightheh
22:51.18drmessanoservice trixbox restart
22:51.29seanbrighthalt
22:51.54bsdwarriorits not asterisk, its my crappy provider
22:52.23drmessanoso?
22:57.44[TK]D-FenderDarn just a little slow to tell him a trick to get it to re-register..
22:58.09seanbrightshare the wealth
22:59.29seanbright[TK]D-Fender: ?
23:00.07[TK]D-Fenderseanbright, Where do you think * stores its registry info?
23:00.22seanbrightdbm
23:00.27jjshoeQwell bahahaha, what a bozo.
23:00.28[TK]D-FenderAstDB <--
23:00.36*** part/#asterisk exothermc (n=miles@74.85.89.146)
23:00.48QwellI should've had drmessano join..
23:00.55[TK]D-Fenderseanbright, So relatively easy to kill off the key, issue a reload and watch it rereg
23:01.19seanbright[TK]D-Fender: indeed.  good call.
23:05.28[TK]D-FenderKernel upgrade.  Stepping out to recompile stuff.
23:05.30[TK]D-FenderBBIAB
23:07.12*** join/#asterisk Frogzoo (n=Frogzoo@124.184.33.9)
23:08.15jjshoeQwell you've got me thinking you sound like you know wtf is going on
23:08.32Qwelljjshoe: GOOGLE
23:08.34Qwell:)
23:14.00Qwelljjshoe: you didn't think I could pull that off, did you?
23:14.05*** join/#asterisk RoyK (n=roy@ip-26-13-149-91.dialup.ice.no)
23:16.01*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
23:16.50*** join/#asterisk Mavvie (n=edwin@ppp121-44-40-133.lns10.syd7.internode.on.net)
23:18.22jjshoeQwell I didn't realize you were going to be so dedicated into fooling him :D
23:18.37jjshoealthough it's friday, and I've had enough of javascript, so I am looking at houses in SD
23:20.39jjshoeanything on friday before a long weekend but work..
23:21.50*** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com)
23:22.15mackesMackes is in.
23:29.13*** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194)
23:40.12*** join/#asterisk anthm (n=anthm@mbf0736d0.tmodns.net)
23:51.34*** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com)
23:56.31*** join/#asterisk angom (n=angom@201.170.65.143)

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