irclog2html for #asterisk on 20050326

00:00.03sivanabrc_: cannot create_adm_p /tmp/cvs-serv32470/include/solaris-compat/sys
00:00.09sivananot sure if that's helpful
00:00.09brc_yup
00:00.21ChujiOk, 6:00, just started my make on Asterisk with a P133. Let's see how long this takes
00:00.34ChujiDamin: Don't you use a 133 too?
00:02.33sivanabrc_: did you fix?
00:02.38brc_sivana, I can't fix
00:02.44sivanabrc_: its working now
00:02.46brc_just going to figure out who's it is and tell them
00:02.54brc_I think the same one keeps breaking
00:02.54sivanaunless I hit a mirror
00:03.02brc_use the ip
00:03.06brc_instead of cvs.
00:03.08brc_to make sure
00:03.20sivanaprobably bkws.. hehe
00:03.24*** join/#asterisk eXoR` (~exor@xdsl-213-196-200-5.netcologne.de)
00:03.26brc_yes
00:04.23*** join/#asterisk Sedorox (~Sed@Neptune-W.client.wlgrv.pa.sed6.net)
00:05.21jessteranyone know if SetCallerPres works on sip ? i don't see it working for me. Im doing a call to another Sip provider
00:05.34Chujisheesh, g729 licensing is a bitch
00:05.38brc_yup
00:05.52ChujiEvery time I monitor, or send to vm, or anything
00:05.57ChujiI need another damn license
00:06.01ChujiThat's crazy
00:06.06brc_no no no no
00:06.10brc_you need a license per channel
00:06.33Chujiif I'm monitoring a g729 call saving to wav, I need two licenses
00:06.40brc_yup
00:06.51bparkerdoes that translate to a license per conconcurrent call
00:06.52brc_I don't really see a better way to do it then per channel
00:06.55brc_bparker, no
00:06.59ChujiOr if a g729 call goes to voicemail, I need two
00:07.03*** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com)
00:07.09brc_Chuji, you sure about that?
00:07.26brc_if you've got a g729 phone, that calls a number then goes to leave a message I believe that only takes one
00:07.28brc_one channel...
00:07.43ChujiFrom Digium support:
00:07.44ChujiVoicemail going to G.729 will use a license.
00:07.44ChujiThe sames goes for flashing over on a call using G.729.  This will us an
00:07.44Chujiadditional license.
00:07.44ChujiIf you are using the monitor application to record the call to something
00:07.44Chujiother than G.729 this will use a license as well.
00:08.06brc_isn't that what I just said?
00:08.21brc_one license per channel up to vm from a g729 device
00:08.38ChujiSo if they leave a message, only one license?
00:09.01bparkera channel = a device?
00:09.05*** join/#asterisk lesouvage (~erik@cc341200-a.assen1.dr.home.nl)
00:09.15brc_no, not at all
00:09.16*** join/#asterisk cripito (~saul@c-65-34-156-173.hsd1.fl.comcast.net)
00:09.16Chujiyeah sip/
00:09.23cripitohi
00:09.45brc_bparker, a channel can be thought of as a pipe to somewhere
00:09.53brc_that could be another device, or voicemail, or whatever
00:09.54bparkerplease explain what a channel is.  I am about to have to license g729 so I need to know this
00:10.08brc_I just did
00:10.08bparkeroh ok
00:10.17bparkeryeah I just saw that
00:10.48ChujiGuess I'm not going to freely use g729 like I thought
00:10.57Chujiespecially since I was going to monitor each call
00:11.08ChujiHave to buy 2 per user
00:11.17brc_if you are monitoring a call between two g729 devices, the monitor app gets stuck in the middle, with a pipe from g729 device a to monitor, then from monitor to g729 device b
00:11.20NuggetI have never encountered anything in asterisk usage that has made me wish I had g729 available to me.  What am I missing?
00:11.22bparkerso when voicemail is concerned, voicemail = 1 channel no matter how many calls are going into it at a given time
00:11.29brc_Nugget, certian devices
00:11.31drumkillajust don't use 729  :)
00:12.01ChujiNugget : I've got some dsl clients that have terrible upstream
00:12.02Chujilots of jitter
00:12.08Nugget*nod*
00:12.10brc_is that what I just said? no, it is not. I said a channel is a pipe from somewhere to somewhere, so if you have two devices called in to voicemail you will obviously need two pipes
00:12.14Chujigsm is OK, but 729 clears it up
00:12.26drumkillaI guess I'm just cheap
00:12.28drumkillaso I just use gsm
00:12.40bparkerok I think I follow now
00:12.41brc_MOST devices do not support gsm
00:12.42ChujiI may have to go to gsm on some of them
00:13.08Chujibrc_ : Good call, I'll have to look at that
00:13.15brc_sides, speex > gsm :)
00:13.20drumkillamost of my voip stuff is just trunking to other boxes
00:13.23lesouvageI have installed sipp (a sip stresstest tool) on a linux box. It runs but is not connecting to my asterisk box. What do I have to do to really stress my *box?
00:13.24Chujibrc_ : I've been using sipura 2000 for this testing
00:13.38drumkillayeah, it's cool how you can tweak speex
00:13.41brc_yup
00:14.10file[laptop]how are you two doing?
00:14.20brc_I'm doing TERRIBLE
00:14.23drumkillaworking as usual
00:14.25Chujibrc_ : So you would use speex over gsm?
00:14.27drumkillaI'm doing GRRREAT
00:14.51brc_hard to say...probably if it was available since you can tune it to your needs
00:15.03brc_(asterisk to asterisk...)
00:16.29jayeolagoing for my 1st compile of asterisk. anyone tested this with sflphone?
00:16.31Nuggethttp://lnk.nu/slacker.com/na.cgi  <-- heh, you can see my conference call this afternoon
00:17.05*** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net)
00:17.23lesouvagescreen 4 of sipp says  "no action found on any messages".
00:19.42*** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net)
00:20.28*** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu)
00:24.39FuriousGeorgehey guys.  i recently managed to get asterisk somewhat configured and more or less starting up with no problems.  now im trying what i think is the next logical step:  connecting a client to make outbound calls
00:24.48FuriousGeorgefor some reason i cant authenticate
00:25.02FuriousGeorgeand i know its not b/c i have the username or pw i set in sip.conf wrong
00:25.28Nuggetwhat does the asterisk console say when it fails?
00:26.00elriahAnyone here having problems with agi STREAM FILE?  as in no errors on agi debug, but just doesn't playback the file?  It seems to work if I install asterisk 1.0, but 1.0.5 and 1.0.7 break.
00:26.10elriahNo errors at all.  Verbose set to 6.  agi debug on
00:26.24*** join/#asterisk Thus0 (~Thus0@dyn-83-154-121-98.ppp.tiscali.fr)
00:26.49Nuggetthen I have to wonder if the device is reaching the server at all
00:27.00FuriousGeorge*CLI> Mar 25 19:26:36 NOTICE[824]: chan_sip.c:7278 handle_request: Failed to authenticate user Brian <sip:Brian@10.0.0.2>;tag=4035506318
00:27.01FuriousGeorgeMar 25 19:27:04 NOTICE[824]: chan_sip.c:7681 handle_request: Registration from 'Brian <sip:Brian@10.0.0.2>' failed for '10.0.0.100'
00:27.13Nuggetoh, I conflated the two questions, sorry
00:27.20bkw_FuriousGeorge, show me your sip.conf entry
00:27.21bkw_please
00:27.25bkw_put it on pastebin.ca
00:27.44*** join/#asterisk JerJer[mobile] (~jj@65.173.197.174)
00:29.43FuriousGeorgebkw_:  thanks a ton, its there
00:29.56drumkillaFuriousGeorge: you have to copy the address back to him :)
00:30.15Nuggethttp://pastebin.ca/8254
00:30.16Nuggetmost recent post
00:30.28bkw_WRONG
00:30.30bkw_hold up
00:30.49FuriousGeorgedrumkilla:  what you mean?
00:30.56FuriousGeorgebkw_:  be gentle, im new
00:31.12drumkillaFuriousGeorge: nevermind, Nugget got it
00:31.26FuriousGeorgeohhhh, the confimation, i got you
00:31.30drumkilla:)
00:31.33FuriousGeorgeill facilitate, next time
00:31.38drumkillano problem
00:31.44bkw_http://pastebin.ca/8255
00:32.04harryvvWhats a common cause of me hearing the other party though a voip carrier but the other party not hearing me? I do have DMZ pointed to my server for the iaxport. Did that after it failed the first time and still not make a difference.
00:32.37bkw_if you wish to use md5secret you need to do echo -n user:realm:pass | md5sum
00:32.46bkw_and put it in as md5secret
00:32.49bkw_and not use secret
00:33.05bkw_brb
00:33.15FuriousGeorgei get it, thanks again
00:33.38harryvvanyway will work on this more latter.
00:36.24bkw_drumkilla, was I nice enuf?
00:37.02drumkillabkw_: that was great  :D
00:37.36bkw_I swear i'm tired
00:37.38bkw_was up at 4am
00:37.40bkw_dog was sick
00:37.45bkw_took her to vet at 8:30
00:37.45drumkilla:(  I'm sowwy
00:37.49bkw_she's all better now
00:37.55bkw_she goes to the dr. more than I do
00:38.39dogz-breed?
00:38.55dogz-btw hi, heh
00:39.03bkw_she's a mutt
00:39.09bkw_little terrier in her
00:39.12bkw_smart as a whip
00:39.16bkw_took one day to house train her
00:39.21bkw_she had exactly one accident in the house
00:39.38bkw_and she won't chew on stuff that isn't hers
00:40.34*** join/#asterisk IQ (~iq@70-59-164-47.omah.qwest.net)
00:40.54bkw_g726 will now work with broadvoice btw
00:40.55dogz-i wish the VP's dog would be like that (refering to chewing on stuff), she thought it be a great boost for company moral if she brought the ~8month old coon hound puppy in
00:41.11dogz-couple hours later i get a call asking what the blue wire is to, and can it be spliced
00:41.21bkw_drumkilla, that rtp fix was needed for that to work.. and it is proper.
00:41.25dogz-little bugger ate through a cat5
00:41.30bkw_OUCH
00:41.33bkw_ZAPPPPP
00:41.36dogz-rebooted my 2950
00:41.48FuriousGeorgebkw_:  for whatever reason im still getting a failed to authenticate
00:41.49cripitobkw_ about the deadlock
00:42.23cripitoi don't have 1.0.7 but have CVS head..
00:42.28*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc)
00:42.37bkw_cripito, no clue if it locks up on cvs-head
00:42.42bkw_I know 1.0.7 and cdr_mysql = FUCKED
00:42.47cripitono... let me explain
00:42.51cripitoyes is what i mean
00:42.59cripitobefore i was running 2 cdr
00:43.00cripito3
00:43.06drumkillabkw_: i added the fix
00:43.13bkw_drumkilla, for rtp?
00:43.13cripitocdr_csv, cdr_Mysql and rate_engine
00:43.15bkw_I seen it
00:43.24drumkillabkw_: I don't think I have ever changed cdr_mysql ...
00:43.38cripitoas soon i take out the cdr_mysql everything works again
00:43.49cripitobut rate_engine is based in mysql too
00:43.57bkw_ya i'm not saying is a mysql issue
00:44.03bkw_its an issue in the code of cdr_mysql
00:44.29cripitocould be in the implementation of queue?
00:44.34cripitolet me see the code wait a sec
00:44.43bkw_no
00:44.51bkw_its a deadlock.. or a block
00:46.44AgiNamumd5secret is transparent to clients right?
00:46.51bkw_yes
00:47.14bkw_you can't have both
00:47.15AgiNamuso it'd be possible to implement a SHA512secret if we wanted to.
00:47.35AgiNamubut the client still sends the user/pass however they normally do it.
00:47.45AgiNamuit's just an internal Asterisk featuer to keep it as a hash
00:48.11FuriousGeorgebtw_:  any idea what else could be causeing the failure to authenticate now that we have removed the md5 auth.
00:50.53tzangerheh
00:50.55tzangerFinally, and most importantly, the act of unclasping the belt lets you know you mean business. You might just be going to the toilet or getting ready for sleep, but you mean business. The trousers aren't even unbuttoned and you mean business.
00:53.16*** join/#asterisk jayeola (~jayeola@dsl-80-43-34-188.access.as9105.com)
00:53.49bkw_md5secret is just a way to keep the plain text out of the conf file
00:54.02booyeah23it wont stop replay attacks
00:54.16bkw_actually it does
00:54.20jayeolahi guys. compiling asterisk, got this err msg "make: *** [asterisk] Error 1; /usr/bin/ld: cannot find -lssl"
00:54.21bkw_you need to go read the sip spec
00:54.29bkw_install openssl-devel
00:54.31jayeolawhat is lssl?
00:54.34FuriousGeorgei get that, i have a small understanding of checksums.  im wondering why my softphone cant authenticate on my server
00:54.37tzangerout of the conf file?  I thought it was to keep the cleartext off the wire
00:54.43bkw_nope
00:54.47*** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net)
00:54.50bkw_the user/pass isn't cleartext on the wire right now
00:54.54bkw_DUH
00:55.02tzangerI wasn't aware of that
00:55.15tzangerI've studied IAX2 a little but never the authentication
00:55.24bkw_we are talking sip
00:55.27tzangeroh
00:55.28BuckRogershey i just got a handy tone 486 and it seems like it is not getting a IP i can console in but the wan is not funktioning
00:55.29bkw_auth=md5 works on iax
00:55.32tzangernevermind then :-)
00:55.48BuckRogersi can not get it to register
00:56.16BuckRogersis thier some odd setting for these ata's
00:56.20booyeah23i dont think a hashing mechanism with nonces will stop replay attacks
00:56.25BuckRogersi went through the user manual
00:57.51bkw_booyeah23, actually I think it will
00:58.07bkw_does it matter
00:58.17bkw_most people are dumb enuf to put context=default in general in sip.conf
00:58.20bkw_thus ANYONE can make calls
00:59.04*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc)
01:02.44Nuggetonly if default also includes trunkld or trunklocal or equivalent.
01:03.56bkw_well some people dont know better
01:04.21Nuggetthat cannot be denied.
01:04.41bkw_that big SECURITY document doesn't help them
01:04.57bkw_like include the outbound context into the inbound menu
01:05.07bkw_NOT A WISE THING
01:06.37bkw_ok its sad.. but who here is tired of hear about Schiavo
01:06.49Nuggetdon't forget to also put voicemailmain(${CALLERID}) in the default context.
01:07.01Nuggetso the whole world can listen to your voicemail
01:07.01bkw_hahahahhahahaha
01:07.11bkw_ya if you fake the callerid
01:07.14Nuggetright
01:07.53bkw_I think a mailboxes settings should live in a .box file or something inside the mailbox
01:07.58bkw_such as playback prefs
01:08.05bkw_to force asking for a password or not
01:08.06*** join/#asterisk Rob- (~robbie@haylott.plus.com)
01:08.33cripitothe stable version could work with realtime?
01:08.41bkw_no
01:08.44bkw_stable can't do realtime
01:08.52bkw_not without ALOT of patches
01:08.55cripito:( so is still sip_friends for stable..
01:09.00bkw_go use head
01:09.06bkw_strap on a pair and use CVS-HEAD
01:09.10bkw_I run it
01:09.13cripitoyeap.. i am using head
01:11.19Nuggetbkw_ loves head
01:11.37file[laptop]giving it too
01:11.47bkw_haha you guys are not right
01:11.50bkw_file[laptop], how would you know?
01:12.37Mavvieanybody with RTP stream insight here?
01:13.07Mavviesometimes my peer sends me packets on one port higher:
01:13.18Mavvie<PROTECTED>
01:13.18Mavvie<PROTECTED>
01:13.33MavvieAnd I have no idea why that is.
01:13.41bkw_depends
01:13.44bkw_are you using cvs-head?
01:13.45bkw_or stable?
01:13.59MavvieMy side is 1.0.7, the other is a non-asterisk box
01:14.08bkw_ok
01:14.15bkw_is asterisk doing register?
01:14.20bkw_or is something registering to atserisk?
01:14.29Mavvieno registration, it's just pushing through calls.
01:14.40bkw_nothing doing a register?
01:14.41bkw_notify?
01:14.43bkw_or anything
01:14.46bkw_ok
01:14.48bkw_here is what can happen
01:14.54bkw_in cvs-stable if its still there
01:15.04bkw_even if something comes in we were allocating rtp for it
01:15.07bkw_till it died off
01:15.15bkw_I think cvs-head fixed it already
01:15.18bkw_but I do recall this
01:15.26bkw_someone was running out of rtp ports or something
01:15.35cripitowe need bk often here :)
01:15.46cripitomore often i mean
01:16.18cripitoone question....
01:16.22cripitoin cvs head...
01:16.40cripitoi have a client using firefly 3th party
01:17.16*** join/#asterisk verge (~jfargen@rrcs-67-78-209-206.se.biz.rr.com)
01:17.50cripitoin the last distro when the guy try to loguin asterisk blows up... after try 2 register the firefly probably a 5000 times
01:17.53Mavvieaha, but the thing is it's coming in from the other side (non-asterisk).
01:18.10bkw_Mavvie, so
01:18.19bkw_it could be any sip packet that could trigger it
01:18.23bkw_notify
01:18.27bkw_ANYTHING
01:18.57Mavvielet me check the traffic again then.
01:19.04bkw_cripito, get a backtrace
01:19.07bkw_and we can see why
01:19.56*** join/#asterisk seong (~seong@219.94.59.19)
01:20.20BuckRogersany one experince with grandstream 486 and asterisk
01:20.32cripitoyup
01:20.37BuckRogerswe have a sipura and a softphone already working
01:20.49cripitoas soon the guy comes in again you got it.
01:20.49BuckRogersbut the grand stream config just doesnt want to work
01:21.08Mavviefunny. I couldn't get the sipura to work, but the grandstream was easy :-)
01:21.25BuckRogersreally ive set all the addresses
01:21.27cripitomm i think i can simulate it
01:21.31BuckRogersit doenst even try to registar
01:21.57*** join/#asterisk IQ (~iq@70-59-164-47.omah.qwest.net)
01:23.05BuckRogersany tips Mavvie
01:30.47*** join/#asterisk macTijn (martijn@linda.net.insecure.nl)
01:35.12*** join/#asterisk lapdr (~inband@ca-fulrtn-cuda2-c2a-243-a.anhmca.adelphia.net)
01:35.36lapdrhi
01:36.22lapdranyone awake?
01:36.53*** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net)
01:37.16*** join/#asterisk znoG (gs@200.115.216.109)
01:37.40voiperhi guys. i am getting a 404 not found while I try to connect to CISCO 5350. when I do sip debug i see this message does this mean anything ? Answering with non-codec capability 0x1 (telephone-event)
01:37.50newllapdr: Nobody here but us chickens.
01:37.57cripitobk still around?
01:38.05lapdr;P
01:38.27cripitobkw_ still around?
01:39.11lapdrI'm wanting to get ISDN NT mode support on NI1 to be able to connect digital isdn sets to Asterisk... but there doesn't seam to be much support/interest... so I thought I might see if anyone here has any leads
01:39.21*** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc)
01:39.25cripitohttp://www.pastebin.com/262814 even with debug and trace on
01:42.29bkw_ok you know you're lame when you have william shatner do a local cable comercial for your law office
01:42.53newlDenny Crane
01:43.12lapdrlol
01:43.33Mavviedenny crane rules :-)
01:43.40Mavvietoo bad he was there for only one episode.
01:44.55newlDenny is usually always in it.  Danny OTOH gets guest appearances.
01:45.31dogz-Does Shatner do anything besides commercials now?
01:45.46Mavvieone season, not one episode
01:49.30*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
01:49.42PTG123anyone here have a problem undefined symbol ast_pthread_create when compiling asterisk?
01:49.55bkw_include utils.h
01:50.01cripitosounds familiar.. yeap ...
01:50.02bkw_or is it util.h
01:50.06tainted-what does it mean 'unable to find a path from ulaw to 729'
01:50.14cripitog729 maybe?
01:50.19brc__out of licenses
01:50.36tainted-i have 729 license
01:50.42PTG123hmm someone should fix the cvs :)
01:50.49bkw_or is it lock.h
01:50.52brc__hahah
01:50.56bkw_ya ya thats it
01:50.57PTG123but including a .h isn't gonna fix it.. its a loading problem, when loading the module
01:51.12tainted-how do i make sure the g729 module is working?
01:51.19PTG123chan_modem.so fails to load
01:51.19bkw_PTG123, what modele?
01:51.21bkw_er module?
01:51.25brc__try g729 show <tab>
01:51.25PTG123chan_modem.so
01:51.36bkw_loaded fine here
01:51.42PTG123ast_pthread_create is defined in the core, so that should def. be loaded before modules right?
01:51.57bkw_not if you don't include the headerfile.. but mine works fine
01:52.12PTG123just pulled it fresh off of cvs
01:52.14PTG123and type gmake
01:52.16PTG123and ran it
01:52.23cripitobkw definitelly cvs head have some problems with the seed and the mapping b/c is also unable 2 read the accountcode from the db
01:52.39bkw_you're using freebsd?
01:52.45cripitome?
01:52.47cripitono FC3
01:52.50bkw_not yet
01:52.56bkw_PTG123, because he says gmake
01:53.06brc__eh?
01:53.12bkw_cripito, whats broken about it?
01:53.13bkw_do tell
01:53.18brc__yes
01:53.30PTG123yes i am :)
01:53.33bkw_do you have an accountcode column in the database
01:53.37cripitoyes
01:53.41bkw_PTG123, use supported OS or fix it and submit a patch
01:53.41PTG123bkw_: why would i use anything else? :)
01:53.47bkw_I don't like linux
01:53.51bkw_but i'm not gonna pull my hair out
01:53.59cripito:D
01:54.00bkw_with a half ass supported os for asterisk
01:54.02PTG123bkw_: every version has worked fine on bsd up to now
01:54.05tainted-i'm trying to get my grandstream 486 to talk to asterisk using g729
01:54.07PTG123bkw_ supported it great
01:54.18tainted-but it keeps saying NOTICE[13195]: channel.c:1724 ast_set_read_format: Unable to find a path from g729 to ulaw
01:54.30brc__tainted-, did you try the command I suggested?
01:54.32brc__what did it say?
01:54.39tainted-it's not a command
01:54.43bkw_lots show translation
01:54.46tainted-g729 show <tab>
01:54.47bkw_s/lots/
01:54.50brc__leme check
01:55.27tainted-g729     -     -     -     - ...
01:55.34bkw_load the codec boi
01:55.37tainted-i'm assuming it's not doing translation correctly
01:55.44bkw_you don't have a codec
01:55.45bkw_duh
01:55.47tainted-bkw_ in modules.conf?
01:55.54tainted-i do
01:55.55brc__you don't have g729 loaded correctly
01:55.57file[laptop]the codec fails to be loaded
01:56.04cripitotry load codec_g729a.so
01:56.05bkw_really?
01:56.06MikeJ[Jayden]it's fat albert....
01:56.08MikeJ[Jayden]hey hey hey
01:56.26PTG123what port is ident on>
01:56.40bkw_113
01:56.47brc__ident is lame
01:56.47bkw_aka auth
01:56.59bkw_brc_ is lame
01:57.05*** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
01:57.07brc_asterisk is lame
01:57.11tainted-ok that worked
01:57.16tainted-i'm an dumb ass
01:57.26file[laptop]brc_ is gay!
01:57.34tainted-i moved the codec.so and forgot about it
01:57.41MikeJ[Jayden]file[laptop]... hehe
01:57.44file[laptop]ooh baby touch me down there
01:57.44tainted-but i still get Mar 25 17:56:59 WARNING[13195]: dsp.c:1469 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833
01:57.52tainted-when i have dtmfmode = rfc2833
01:58.00tainted-both in sip.conf and on the ATA
01:58.15bkw_must not
01:58.23cripito:) check your peer definition
01:58.29cripitob/c definitelly looks like not
01:58.50bkw_your conf file is tainted
01:58.50brc_whooop whooooooop, warning, configuration error detected
01:58.51bkw_:P
01:58.51cripitook bkw_ coming back to the point i have 2 versions of cvs head
01:59.07bkw_k
01:59.07tainted-problem is.. peer is ulaw
01:59.09bkw_and?
01:59.09cripito1 works fine
01:59.23bkw_diff them
01:59.31file[laptop]bkw_: we need more Mac hardware
01:59.36bkw_file yes I know
01:59.38tainted-and peer<---ulaw--->*<---g729--->gs486ATA
01:59.40bkw_I want MORE MORE MORE
01:59.51file[laptop]bkw_: wanna go half on an Xserve and we can colo it? :p
01:59.54brc_file[laptop], you are utterly corrupted
01:59.57bkw_file tempted
02:00.01brc_I'll go 1/3
02:00.08bkw_I have colo space
02:00.17file[laptop]eep Peacekeeper Wars
02:00.18file[laptop]bbl
02:00.23Nuggetyay xserve
02:00.28bkw_once ya go mac you never go back
02:00.35cripito:D i have a spare server
02:00.36Nuggetyup
02:00.45bkw_I went mac.. never gonna go back
02:00.45cripitoAsterisk CVS-HEAD-02/16/05-13:50:59 built by root@sls-ce2p4 on a i686 running Linux
02:00.45cripito<PROTECTED>
02:01.21cripito#define BUILD_VERSION "CVS-HEAD-03/25/05-20:22:08" this one
02:01.21bkw_cripito, step forward day by day till it breaks from 2/15
02:01.26bkw_er 2/16
02:01.30cripitowow!!
02:01.31cripitook
02:01.31bkw_or jump 4 days at a time
02:01.38bkw_then zero in on when it broke
02:01.46cripitothat's sounds better
02:01.47bkw_thats usually what I do when I can't find it
02:02.03MikeJ[Jayden]once you get to day, get to the specific file version that broke it if there are multiple that day
02:02.05bkw_its amazing when something totally unrelated chances.. then breaks something else
02:02.09*** join/#asterisk jayeola (~jayeola@dsl-80-43-34-188.access.as9105.com)
02:02.17cripitoyeap i known :)
02:02.32cripitook bk i will
02:03.28tainted-is rfc2833 supposed to be ass?
02:03.40bkw_no
02:03.41bkw_it works great
02:03.48MikeJ[Jayden]sip is ass
02:03.49tainted-keeps missing digits
02:03.53bkw_no sip isn't ass
02:03.56bkw_just the way asterisk does it is
02:03.57MikeJ[Jayden]:)
02:04.18MikeJ[Jayden]ok, how about sip is very complicated for just voice... is that better :)
02:04.38bkw_what dual 2.5ghz with 1.25 FSB not good enuf?
02:04.39jayeolai keep getting this error "/usr/bin/ld: cannot find -lssl " even though i have libidn libidn-devel installled
02:04.49bkw_dis I say libidn
02:04.49cripitotaitend 1 question
02:04.51bkw_you dummy
02:04.54bkw_I said OpenSSL
02:04.57bkw_OpenSSL-Devel
02:04.57NuggetI'd kick myself if I bought one and a month later the new ones came out.
02:05.00cripitoyour provider is?
02:05.02Nuggetand new ones are overdue
02:05.14bkw_-lssl = OpenSSL
02:05.29jayeolaaha!
02:05.36bkw_Nugget, honestly this 1.6ghz G5 is whippin the piss out of my P4
02:06.08bkw_1.2ghz G4 in my ibook
02:06.08NuggetI believe it
02:06.14*** join/#asterisk infra (~infra@216-251-177-106.ips.cpinternet.com)
02:06.15bkw_1.6ghz G5 in my imac
02:06.21bkw_768 in my ibook
02:06.25NuggetI'm on a 1ghz powerbook right now.  coming up on three years old and it's time to move on.
02:06.26bkw_1 gig ram in my imac
02:06.57infrahi, anyone with Quicknet PhoneJack experience?
02:07.08cripito:)
02:07.09bkw_RUN FOR THE HILLZ
02:07.13cripitoasterisk in mac?
02:07.18cripitogood 2 known
02:07.28bkw_I run asterisk on my ibook
02:07.30MikeJ[Jayden]asterisk runs everywhere man.....
02:07.30bkw_works great
02:07.35NuggetI run asterisk on my powerbook
02:07.40bkw_its fun
02:07.41tainted-what is DTMF Payload Type and what should it be?
02:07.46bkw_101
02:07.48bkw_AVT
02:08.16cripitomike :) now yes... was funny find the win32 version for moco$
02:08.33bkw_and did he put up patches yet
02:08.34MikeJ[Jayden]I have had it on windows for a few months now
02:08.53MikeJ[Jayden]bkw_, I e-mailed with him, he said monday
02:09.19cripitoi prefer the linux version...
02:09.22MikeJ[Jayden]but his runs a lot smoother than my hack job... so I really want to see his patches
02:09.28jayeolaopen-ssl ain't in my repositary, where can i get it from? sourceforge?
02:09.33bkw_AsteriskWin32 Source Code 0.51
02:09.33bkw_You need asterisk 1.0.5 source code at ftp.asterisk.org
02:09.33bkw_and AsteriskWin32 cygwin patches, avaible soon, come back later !
02:09.40bkw_Mike I doubt it
02:09.47bkw_I so totally doubt it
02:09.58bkw_ya the "working" linux is the best
02:10.02MikeJ[Jayden]I know, I they are not out there next week....
02:10.06MikeJ[Jayden]but we will see
02:10.11cripitojayeola what SO are you running?
02:10.48cripitook any way to bring the daily patch for asterisk apart of cvs?
02:11.03MikeJ[Jayden]cripito, huh?
02:11.09hmodeswhich do you seek?
02:11.21bkw_OMG ITS HMODES
02:11.23cripitoall from 2/16
02:11.24bkw_hey honey how are you?
02:11.33cripitotill today
02:11.33hmodeshowdy bkw :)
02:11.39bkw_cvs co -d "2005/02/16" asterisk
02:11.40hmodesi'm ok i guess, you?
02:11.41bkw_or some shit like that
02:11.43bkw_man cvs
02:11.43cripitoyea yeap
02:11.56bkw_oh don't forget the -r
02:11.57cripitoi have 2 pull a months!!! don't be bad guy ;)
02:12.01bkw_for v1.0
02:12.02bkw_or what ever
02:12.16bkw_;)
02:12.18jayeolacripito: i'm a bit s-l-o-w. all i had to do was look for openssl and not "open-ssl"
02:12.34cripitoyes jayeola.. all that u need is openssl
02:12.39cripitodevel
02:12.44jayeolauh-huh
02:12.50cripitoif u have yum
02:12.52jayeola<-- apt-getting
02:12.57cripitobetter
02:13.06infraSeeking Quicknet experience; have rxgain problem.
02:13.14cripitou need openssl and openssl-devel
02:13.35cripitook bkw i am bringing the patches i will tell u what day broke
02:19.07*** join/#asterisk abcdefgh (~infra@216-251-177-106.ips.cpinternet.com)
02:22.32infraAnyone have experience with Quicknet Internet PhoneJack cards?
02:23.07newlWhen the A party calls B party and their number is not presenting, is there a common string that's given such as "private number" or does Asterisk mung that for common output?
02:23.10infraalso...anyone recommend best current apps reference (old *docs book is from 2003)?
02:25.20bkw_app_skel.c
02:26.12DaminWhat about app_moospenis?
02:27.34bkw_ya ya
02:27.37bkw_i'm in da conf
02:27.41bkw_but nobody is there with me
02:29.00booyeah23is it possible for a context to be null?
02:29.19bkw_depends.. in some cases it falls back to default
02:29.21booyeah23what does it mean context ' '
02:29.30bkw_paste the message in here
02:29.34booyeah23i think the problem im having with this bridging thing is because of it
02:30.05booyeah23<PROTECTED>
02:30.15booyeah23ignore that
02:30.16booyeah23Unable to find extension '12' in context ''
02:30.23booyeah23basically im bridging to calls together
02:30.49booyeah23user1 calls pbx, forwards to user2
02:30.52booyeah23with tT
02:30.58booyeah23T doesnt work
02:31.15booyeah23t works, user1 can transfer
02:31.45booyeah23user2 cannont transfer
02:31.57booyeah23*cannot
02:32.09booyeah23i fixed the problem with the native bridging
02:32.50booyeah23notransfer=yes was not working correctly, so i added a check to just not native bridge if that was set
02:34.00booyeah23in chan_iax2.c
02:34.14PTG123app_moosepenis is bkw_s favorite app
02:34.38booyeah23http://pastebin.ca/8257
02:35.18booyeah23i guess i am going to run a debugger on it when it gets the #
02:35.21booyeah23and see what the structures are
02:35.34booyeah23im guessing the context for the person being called is null
02:36.31jayeolahurrah! asterisk compiled
02:36.58jayeolanow to get it working with gnomemeeting or sflphone
02:37.01*** part/#asterisk JerJer[mobile] (~jj@65.173.197.174)
02:42.52bparkeris there a trick to getting moh working
02:44.40infraIs this channel's traffic archived anywhere?
02:48.10tzafrir_laptopgnomemeeting requires h323, right?
02:49.37jayeolainfra: if you can log this convo in irssi or xchat quite easily
02:50.05jayeoladepends on what irc client you are running
02:50.14FuriousGeorgecan someone give me a hand getting x-lite aquthenticated with my very preliminary configuration of asterisk
02:50.15*** join/#asterisk Othello (Othello@nusnet-219-211.dynip.nus.edu.sg)
02:50.27FuriousGeorgecuz im at a wall over here
02:50.30*** join/#asterisk sedwards50 (~chatzilla@adsl-66-120-116-250.dsl.sndg02.pacbell.net)
02:50.46infraI mean a searchable archive of old traffic...
02:51.05infraPerhaps my problems have been discussed...
02:51.06sedwards50anybody have experience with info-digits/ani2?
02:51.24cripitoguys anyone can recommend a phone for linux?
02:51.34FuriousGeorgesoftphone?
02:51.47Syncrosiaxcomm
02:51.49Shido6FuriousGeorge - user and peer
02:51.54Shido6brb - Pistons Game
02:52.11cripitoyes softphone
02:52.47sedwards50iaxcomm 1.0rc2 seems a bit buggy
02:52.52FuriousGeorgeShido6:  ill try that, but the exaples i see list it as friend always
02:53.29FuriousGeorgecripto:  let me preface this by saying ive never used any.  there's also gnuphone, and if u contact xten u can get a beta of xten for linux
02:53.52sedwards50Does anybody know how to read the info digits (aka ani2) from a pri t1?
02:53.56jayeolaanyone tried/used this; http://sflphone.org ?
02:53.56cripitommmmm that one sounds nices
02:54.08bparkerdoes anyone know what codec moh uses by default
02:54.50cripitosip... mmm lets trying.. thanks guys
02:55.23infraIn the absence of any experience on Quicknet, anyone recommend a hardware line amp?
02:57.07*** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net)
02:57.50FuriousGeorgeshido6:  i tried that.  x-lite says "404 Not Found" and asterisk says: Mar 25 21:57:32 NOTICE[4901]: chan_sip.c:7681 handle_request: Registration from 'brian <sip:brian@10.0.0.2>' failed for '10.0.0.100'
02:57.55infraI guess I'll try during business hours :-)
02:58.46newlFuriousGeorge: I've got mine configured as friend and it works fine.
02:59.14FuriousGeorgenewl:  yeah i tried that too
02:59.29FuriousGeorgekeep in mind im very new to this.
03:00.00FuriousGeorgebesides the obvious parts where i define my sip clients, give them pw's etc, what else is required for x-lite to log into *
03:00.08jayeolaany uk users here? or from euroland? i'm looking for a good sip provider
03:01.37*** join/#asterisk clh (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net)
03:02.20*** join/#asterisk sedwards50 (~chatzilla@adsl-66-120-116-250.dsl.sndg02.pacbell.net)
03:02.32*** part/#asterisk clh (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net)
03:03.03*** join/#asterisk doughecka (~Doug@doughecka.user)
03:03.40newlFuriousGeorge: Try here and see how you go. http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20xten%20xlite
03:04.27jayeolathanks
03:05.05jayeola<-- just compiled asterisk successfully. using google and this guide; http://www.automated.it/guidetoasterisk.htm#_Toc49248757
03:08.44FuriousGeorgenewl:  thanks, that was the guide i found and used earlier today, and i had forgotten about it.  i still get the same error but maybe after i apply the registry changes it recommends...
03:09.49newlFuriousGeorge: maybe.  I've not tried it from the Win32 client.  The Linux version works as expected though.
03:10.39Othellohi guys
03:11.05OthelloI got a little problem ...
03:11.34Othelloit seems that the console driver for asterisk will NOT work on a ES1371 sound card on a real machine
03:11.54OthelloI have tried it on a VMWare machine and it woerks
03:11.58Othello:/
03:12.23niZonThe console driver doesn't seem to work in vmware for me
03:12.26niZonunless i missed something
03:12.37jayeolanewl: did you use some kinda phone to get it working or regular headphones and a mic?
03:12.45OthelloniZon: what kernel are you running?
03:12.58Othelloand niZon: which version of VMWare?
03:13.05niZon2.4.27
03:13.05newljaiger: headphone/mic
03:13.12niZonvmware umm
03:13.20niZon4.5.2
03:14.07Othelloah ok
03:14.13Othelloyou got the right audio drivers working?
03:14.22niZonprobablly not :P
03:14.44Othelloit's supposed to emulate an es1371
03:14.50Othellogo configure that in your kernel
03:14.57niZonmk
03:15.02niZondoes ztdummy work for you?
03:15.10Othelloor try 'modprobe es1371'
03:15.37OthelloniZon ... sorry for asking but what does ztdummy do?
03:15.50niZonfake zaptel device for timing
03:15.56niZonused by meetme
03:16.43OthelloI see...
03:16.49Othello:S ...
03:16.59OthelloI still have a lot to learn
03:18.28niZonyeah me too
03:18.44niZonwhat does your dialplan look like for dialing console?
03:18.59*** join/#asterisk klasstek (~nunyobiz@c-24-9-148-246.client.comcast.net)
03:19.06newlphone $HOME
03:19.53bkw_so guys
03:20.26file[laptop]yes bkw?
03:21.46*** join/#asterisk jayeola (~jayeola@dsl-80-43-34-188.access.as9105.com)
03:21.51Qwellhmm
03:21.53Shido6you can use it for iax2 trunking
03:21.54Shido6too
03:22.34OthelloniZon ...
03:22.34QwellShido6: Do I have to call to get that "other" DID removed from my account?
03:22.34OthelloI don't have a dialplan
03:22.34Shido6did removed?
03:22.34OthelloI used the "make samples" thingy
03:22.36Shido6is it a MI number?
03:22.40Othelloand just type 'dial' on the console
03:22.43Qwellno, tollfree.  the one that was mapped...somewhere
03:22.44Shido6dont make samples
03:22.57Shido6tollfree? hrmm
03:23.00Shido6dont worry about it
03:23.08niZonOthello: um, oh :P
03:23.10Shido6ur not getting billed if no one uses it
03:23.16Qwellok
03:26.53FuriousGeorgegrrr...  still nothing.  wtf
03:27.18Othellointerestingly enough ... mpg123  works and I do hear a "beep" when I type 'dial' but nothing more
03:27.38Othelloit's almost as if something made the playback thread get stuck
03:28.29OthelloShido6 ... what do I do then?
03:28.30FuriousGeorgewhen i put in the wrong pw it says the same thing as when i put in the right one.  what gives.
03:28.55*** join/#asterisk NewSole (~david@i216-58-44-245.avalonworks.net)
03:29.05IQHi, anyone using Festival with * ?
03:29.49*** join/#asterisk sergey (~none@195.151.15.13)
03:30.01*** join/#asterisk jskcr (~jskcr@jskcr.user)
03:32.01jayeolado i need a -statis- ip? at the moment i have a dynamic one provide, (i think), by my adsl ISP
03:32.55IQFestival TTS wit *. Anyone using it ?
03:33.28newlFuriousGeorge: try setting nat=yes for that entry and see if it still happens.
03:35.16FuriousGeorgestill happens
03:35.34ariel_jayeola, you can use services like dyndns.org to setup your ip address and I use my asterisk system with it.
03:37.12FuriousGeorgewhat does it mean "handel_request"  does it have a problem with my suername or soemthing?
03:37.38ariel_FuriousGeorge, your xlite in on the same network as the asterisk is it not?
03:38.20FuriousGeorgeyes
03:38.37Shido6.
03:39.03ariel_FuriousGeorge, your settings for sip.conf your account is brian? [brain]
03:39.34jayeolathanks ariel_ !
03:40.19*** join/#asterisk yaboo (~jsirucka@220.245.131.131)
03:40.25FuriousGeorgeyeah, in parenthesis above the settings for the client
03:41.26ariel_that is your login name.  what are the settings you have in the xlite?
03:43.48FuriousGeorgei thought username= sets the username and the thing in parenthesis was just an arbitrary name for the settingfs in that part
03:44.06anachronwhen you checkout the latest zaptel via svn .. where does it put the files?
03:44.11FuriousGeorgecuz in the [] its capital and in the username= and xlite its lowercase
03:44.14ariel_In setting the xlite you need username, and authorization user, same brian, your password next and put the asterisk ip address in the domain, and sip proxy only.
03:44.26ariel_FuriousGeorge, change it to lower case
03:44.44FuriousGeorgein process
03:46.46FuriousGeorgebrb
03:48.27*** join/#asterisk Newbie___ (some@60.48.163.164)
03:49.08Newbie___hi all, i am about to purchase a Zhone Channel Banks 16FXS & 8FXO, any advice ?
03:50.23ariel_Newbie___, good luck. I have had nothing good about the zhones I used. I have 8 of them sitting getting dust in there boxes. I don't think there even good enough to sell on ebay.
03:51.05ManxPowerI assume everyone has seen this? http://www.packet8.net/about/virtual_office.asp
03:51.08*** join/#asterisk Negrobird (~Rober@resnet240001.utdallas.edu)
03:51.17ariel_no I have not
03:52.04jayeolabrb
03:53.03*** part/#asterisk sergey (~none@195.151.15.13)
03:53.46Newbie___ariel_: is a piece of junk ?
03:54.32ariel_Newbie___, the ones I have are. But that just my view of them. I have 8 of them as paper holders.
03:54.46IQNewbie___: No, ariel_ is not a piece of junk
03:55.14ariel_IQ you want to buy mine 8 c/b
03:55.49Newbie___voip-info.org says is good if use as FXS
03:55.50ManxPowerAs long as you FULLY UNDERSTAND the limitations of them.
03:55.52Newbie___hmmm
03:55.54*** part/#asterisk trig_hm (~jb@home.monkeypr0n.org)
03:56.06IQariel_: I am on your side dude. I'm just telling him that ariel_ is not piece of junk :)
03:56.12NewSolewhat are they
03:56.17NewSoleZhone Channel Banks
03:56.42*** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net)
03:56.58dmccollumHello Everyone!
03:57.04ManxPower~docs
03:57.05jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
03:57.05Newbie___NewSole: the one i am looking at has 16 FXS and 8 FXO
03:57.07ManxPower~mailinglist
03:57.08jboti guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
03:58.38ManxPowerI think the Packet8 Virtual PBX a tad expensive.
03:58.48dmccollumI saw a post in the mailinglist archives about changing the default voice that festival uses and I can't seem to figure out the post from bkw on how it actually works. Anyone care to help me if I post the short post?
03:59.10ManxPowerdmccollum: there are several docs out there, check google
03:59.17ManxPower~google festival mbrola
03:59.47IQWhen Digium will go public?
04:00.02ariel_Newbie___, hope they work for you.  What is the unit going to cost you?
04:01.19dogz-im currently having problems with * answering incoming calls, outgoing calls work fine.. Im concentrating my efforts in extensions.conf
04:01.26dogz-is this correct? or should i be looking somewhere else
04:01.37ManxPowerdogz-: Either that or zapata.conf
04:01.44ManxPower/etc/asterisk/zapata.conf
04:01.55dogz-thanks
04:02.17ManxPowerdogz-: Do you see ANYTHING in the CLI when a call comes in?
04:02.24dogz-no
04:02.33ariel_argh all they have on the news is this thing with Terry Schiavo
04:02.36Newbie___ariel_: 99.00
04:02.59dogz-ManxPower: im leaning towards my "incoming" statement, but i may be wrong
04:03.00ariel_Newbie___, ebay?
04:03.09Newbie___ariel_: yes, but undecided
04:03.13dogz-i will play with it a bit, see if i cant get this bugger worken
04:03.31dmccollumWould you rather listen to news about Michael Jackson?
04:03.45Newbie___Packet8 offers international plan at 49.90, but cell calls not included
04:03.46ariel_dmccollum, no
04:04.10Newbie___why dont they include cell calls
04:04.14ariel_Newbie___, if you can't return it. think if what your time is worth.
04:04.37ManxPowerdogz-: dunno
04:05.12ManxPowerNewbie___: because in places other than the USA/Canada calling a cell phone costs more than calling landlines.
04:05.15Newbie___ariel_: maybe i would go for something more reputable like ADT or ADC i forgot the name
04:05.19ariel_Newbie___, international cell calls are more per minute then what they are like in the US and Canada.
04:05.24QwellManxPower: I think I saw a place where it was less
04:05.31ariel_Adtran,,,,adtran
04:06.24Newbie___there is this web site www.telextreme.com, they include selected countires cell. when i tried to call them, i think the lady know nothing
04:06.32*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc)
04:06.42Newbie___is a MLM company
04:07.07dmccollumHmm, Swedish Female MBrola voices. I may have to put the weather extension on my speed dial.
04:07.55FuriousGeorgeariel:  im not sure what i did, but i had some degree of success.  now asterisk says -- Registered SIP 'brian' at 10.0.0.100 port 5060 expires 1800
04:07.55FuriousGeorge<PROTECTED>
04:08.14*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc)
04:08.29FuriousGeorgebut when i dial out x-lite makes the busy sound and displays 404:  not found
04:09.33ariel_FuriousGeorge, then your context is incorrect for your calls
04:09.38*** join/#asterisk jayeola (~jayeola@dsl-80-43-16-212.access.as9105.com)
04:09.38IQFuriousGeorge: Once I was getting 404, kept looking on my configuration. but didn't find anything - then I found out that the number I'm dial is busy
04:10.31IQFuriousGeorge: Then everything started working - I'm not sure if this could be the case - or maybe I was facing some network issue
04:10.46*** part/#asterisk jayeola (~jayeola@dsl-80-43-16-212.access.as9105.com)
04:11.22FuriousGeorgeariel:  you mean at the top of my sip.conf where i have context=default (for now)?
04:11.36FuriousGeorgeIQ:  the number im trying to  dial is the info # for the SIP provider
04:11.58QwellFuriousGeorge: Do you have a context specified for your xlite user?
04:12.02Newbie___life in communication really sucks
04:12.07IQFuriousGeorge: I see... well, just wanted to tell you what I experienced
04:12.13FuriousGeorgemaybe it is busy, ill try another.
04:12.22FuriousGeorgei do appreciate
04:12.43Kattyhi lads
04:12.54IQFuriousGeorge: by any chance are you using brain net ?
04:13.13ariel_evening Katty
04:13.20FuriousGeorgei will put context=default with the clients settings,
04:13.29Kattyariel_: how's teh family?
04:13.43FuriousGeorgebut to be honest, i just picked this up last week so my understanding is weak to say the least
04:13.50*** join/#asterisk jayeola (~jayeola@dsl-80-43-16-212.access.as9105.com)
04:13.54ariel_FuriousGeorge, you should not use default for anything other then a trap.
04:14.11FuriousGeorgetrap?
04:14.30ariel_Katty, fine thanks.  Getting ready for my little girls bday. She will be 2 years old.
04:14.31Qwelldefault should "do" anything
04:14.50Kattyariel_: yay!
04:14.58Kattyariel_: remember napkins!
04:15.06ariel_default should only have exten => X.,1,Congestion
04:15.24Kattyariel_: i've been working the arms out. they're getting dreamy
04:15.32Kattyariel_: wanna see? (=
04:15.42ariel_sure
04:15.50*** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net)
04:15.52Kattyhttp://www.brick.net/~izaah/tehflex.jpg
04:16.04yaboowhats a fwd test number I can call
04:16.14ariel_k
04:16.14Qwellyaboo: 612 I believe
04:16.41yaboothanks
04:16.51yabooQwell still trying to sort out my fwd dialing
04:17.08ariel_65342
04:17.40CoolAcidMy FWD number: 510016
04:17.41Kattyi think i'm going to try for the croft look
04:18.03Kattymaybe that way i could lift a monitor without help ;)
04:18.22ariel_Katty, you look fine the way you are.
04:18.39Kattyariel_: i'm not toned though. i like toned.
04:18.58FuriousGeorgex is acting up.  brb
04:19.00CoolAcidAnyone got IAXTel working ok?
04:19.09QwellCoolAcid: yep
04:19.31CoolAcidQWell: Herm.. Mine doesn't dial well, and when it does, it's garbled. But FWD is fine.
04:20.38CoolAcidHave good ping times to iaxtel.com too
04:20.55ariel_CoolAcid, yes
04:21.12*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
04:21.44Newbie___ariel_: if i want to connect my cell to *, i would need a GSM destop phone and a FXO , right ?
04:21.57yaboodamm this fwd is going to tear me around
04:22.14yaboofollowed the voip-info.org page to no success
04:22.15ariel_Newbie___, I have no idea. I have never looked into connecting a cell to asterisk
04:22.42*** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net)
04:22.47KattyNewbie___: you sure have a long tail.
04:23.01Newbie___it seems logical beacuse GSM desktop phone has a dial tone , but than i can be wrong
04:23.16Newbie___Katty: lol
04:23.39yabooanyone able to walk me with why fwd under dosen't work
04:23.45CoolAcidyahoo msg me
04:25.31*** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca)
04:26.31*** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net)
04:30.19*** join/#asterisk nix000 (~nix000@66.11.165.71)
04:30.21ariel_it's late and I need to get up early.  See you all have fun.
04:31.28nix000stevek: ping
04:32.20dmccollumSo do I need to download the mbrola binaries also, or just the additional voice files?
04:34.25*** join/#asterisk wildcard` (~blah@d209-121-36-100.bchsia.telus.net)
04:34.29wildcard`hi every one
04:34.44wildcard`im new to asterisk im about to be setting up some units inside a few businesses
04:34.53wildcard`and i was wondering if theirs any access to jamica
04:34.59wildcard`jamacia
04:35.04implicityes
04:35.15wildcard`awesome ! how many nodes ?
04:35.19wildcard`or can u tell
04:35.45wildcard`im gonna be purchasing 2 of these Grandstream Budgetone 100-101 (black)
04:35.51wildcard`from pulver innovations
04:35.56implicitwhat kind of 'access' are you talking about
04:36.10implicityou want jamaican routes?
04:36.12wildcard`can i setup a asterisk server here in british columbia canada
04:36.12*** join/#asterisk JT19 (~JT@71.98.88.202)
04:36.19wildcard`and connect to a land line in jamaica
04:36.25implicitsure
04:36.33wildcard`ya jamaican routes
04:36.40wildcard`is there a list available
04:36.41implicitwhat kind of capacity are you looking for
04:36.42wildcard`of the routes
04:36.47wildcard`just 1 line
04:36.50implicitE1, 10xE1, DS3
04:36.50implicitoh
04:37.00implicityou want DIDs or just outbound?
04:37.26wildcard`i wana have a customer be able to place a call outbound from british columbia over the asterisk network to jamacia
04:37.35wildcard`if thats how u spell it
04:37.38implicitasterisk is not a network
04:37.45wildcard`oh ya thats the PBX
04:37.47implicitbut it is possible to get jamaican routes
04:37.49wildcard`software rite
04:37.49Shido6heh
04:37.56wildcard`whats that online network
04:37.57dogz-someone mind taking a quick look at my configs http://pastebin.ca/8262 Im having issues accepting incoming calls... Im thinking its my extension.conf
04:37.59wildcard`u can join with
04:38.13wildcard`where you provide your phone
04:38.21wildcard`in exchange for access to other ppls phones
04:39.04Shido6accepting incoming calls from here, dogz and to hat
04:40.01dogz-?
04:41.06IQHI, how do I call console from an extension?
04:41.17JunK-YIQ: huH?
04:41.27JunK-Ya system command ya mean?
04:41.34JunK-YSystem(blah)
04:42.20IQlike in dial plan I can set up an extension like 999?
04:42.31CoolAcidDial(Console/dsp)
04:42.44IQ<PROTECTED>
04:42.55CoolAcidno , /
04:42.56dogz-Shido6: sorry didnt understand what u said
04:43.14CoolAcidexten => 999,1,Dial(Console/dsp)
04:43.33IQCoolAcid: thanks :)
04:43.59*** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net)
04:45.29*** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net)
04:45.40dogz-Shido6: If you were asking where accepting incoming calls, i mean from my POTS connection (channel 4)
04:45.56*** join/#asterisk coppice (~chatzilla@209.203.17.210.dyn.pacific.net.hk)
04:46.27CoolAcidDamn.. ipv6 is working.. really slow to pastebin.ca :)
04:46.49dogz-=]
04:47.47CoolAcidI have no work with FXO's but shouldn't signalling=fxs_ks read something about FXO not fxs?
04:48.10CoolAciderm.. no support with zaptels at all :)
04:48.47dogz-I thought with the analog connections you had to inform asterisk what function it was perviding
04:48.57dogz-providing*
04:48.59bkw_moving on
04:49.04bkw_you signal FXO with FXS
04:49.07bkw_and FXS with FXO
04:49.08bkw_NEXT!!!
04:49.28dogz-woohoo did something right :)
04:49.30dogz-i dink
04:49.35bkw_dink?
04:49.36CoolAcidNo one was answering.. Sorry to help.
04:49.40CoolAcidthink.
04:50.08dogz-dink = think, sorry drinking atm
04:50.36bkw_YAY
04:50.39bkw_i'm crankin my ipod up
04:50.43bkw_trying to brain storm
04:50.48dogz-bkw_, u mind taking a look at my pastebin?
04:50.55dogz-http://pastebin.ca/8262
04:50.56bkw_I can try
04:51.05dogz-thank you
04:51.11bkw_LET ME GUESS
04:51.16dogz-im having issues accepting calls
04:51.17bkw_you're using 2.6.11? or .10
04:51.19dogz-over the fxo
04:51.20bkw_something higher than 2.6.9
04:51.24bkw_kernel
04:51.30dogz-freebsd actually
04:51.35bkw_move along....
04:51.45cypromislol
04:51.52bkw_can't help or even begin to help on a platform thats not offically supported...
04:52.05bkw_I like hardware that somewhat works
04:52.17bkw_but running in on FREEBSD = hardware that might never work right
04:52.18cypromisyou o ?
04:52.22cypromiswhere do you find it ?
04:52.23dogz-=p aight, thanks anyways
04:52.31bkw_I see you lookin at me like I got what ya need
04:52.51bkw_I wish I could help
04:52.59bkw_but the freebsd stuff differs a bit from linux
04:53.00Kattybkw_: you're just in time to say g'night!
04:53.11bkw_Katty, I LOVE YOU!!!!!!!!
04:53.15Kattyoh
04:53.19Kattyuh
04:53.21Kattyk
04:53.24bkw_you can be my sexless lover
04:53.32dogz-no problem :) im just waiting for FC3 isos to finish downloading
04:53.33Kattythat sounds fun
04:53.36Kattydoes it include muffins?
04:53.40bkw_yes
04:53.44Kattyexcellent, brt
04:53.46bkw_muffins are good
04:53.58bkw_I have about 10 sexless lovers.. most of them are men
04:54.02bkw_straight men at that
04:54.03IQCoolAcid: can't dial Console from SIP extension :( ... Console to SIP works
04:54.06Katty:<
04:54.13bkw_you're the first female
04:54.16bkw_hehe
04:54.20Kattyyou've obviously insaned
04:54.29KattyTHIS CALLS FOR MUFFINS
04:54.30CoolAcidwhat happeneds in cli?
04:54.31*** join/#asterisk NormAst (~NormAst@toronto-HSE-ppp3972900.sympatico.ca)
04:54.37bkw_muffins mmmmm.. yummy
04:54.39bkw_oh file dear
04:54.40bkw_wake up
04:54.43bkw_someone is talking muffins
04:54.48Kattyi mean
04:54.49lilneonhey has anyone got SMS working with asterisk?
04:54.50Kattybedtime
04:54.51Kattyyes
04:54.54bkw_hehe
04:55.00Kattystop distracting me!
04:55.06Kattycharlie is waiting!
04:55.08bkw_lilneon, the SMS in asterisk is euro only
04:55.16lilneon:( bummer
04:55.17bkw_if you're in the US you're SOL
04:55.18Kattyand heaven forbid i keep the kitty waiting
04:55.27bkw_see in euro they send callerid before the first ring
04:55.34bkw_so you can get away with sending message to phones
04:56.12lilneonbkw_:ouch.. so the rest of us gotta just sit on our thumbs and lok for other sms solutions?
04:56.17JT19newbie here with a question -- I'm looking to replace an old Nortel system in a small office that is used to having 4 line appearances, and everyone placing / picking up calls from on hold... Is there a way to implement this with asterisk and sip? I've got asterisk doing a TON of other cool, stuff, but not sure how to do this....
04:56.17IQCoolAcid: Mar 25 16:53:02 WARNING[6820]: channel.c:2033 ast_request: No channel type registered for 'System'
04:56.34bkw_lilneon, their isn't a "true" sms solution
04:56.49bkw_IQ don't dial(System/
04:56.49bkw_ninny
04:56.55bkw_system is an app
04:56.57bkw_not a channel driver
04:57.01bkw_:)
04:57.06cypromishey babe dial my system
04:57.10lilneonbkw_:so.. there are only ppl talking to providers smsc's? and charging for it?
04:57.21bkw_smsc's are too fucking expensive
04:57.28bkw_5 cents per message my FUCKING ASS
04:57.36lilneonbkw_:yes they are..
04:57.38bkw_some guy posted an asterisk solution
04:57.38cypromisdepends how many smses you send
04:57.43bkw_but its not really an asterisk solution
04:57.54bkw_cypromis, ya but I want a fucking connection to the SMSC
04:57.57IQbkw_: so I can not dial Console from an extension ?
04:57.57bkw_how can I do that?
04:58.01bkw_without paying out the ass
04:58.02lilneonbkw_:ok.. worth a look.. where???
04:58.05coppicebkw_: free SMSCs are not expensive, but connecting them to others can be
04:58.07CoolAcidyes.. as I posted before.
04:58.11cypromisbkw: run your own smsc
04:58.19bkw_coppice, where can I get connected to an smsc?
04:58.24CoolAcidIQ: exten => 999,1,Dial(Console/dsp)
04:58.26bkw_got any links?
04:58.26Newbie___anyone try out 2VA VoIP Asterisk yet ?
04:58.38IQCoolAcid: No need for 999,2,Hangup ?
04:58.45coppicebkw_: you must pay and pay and pay....
04:58.51bkw_haha
04:58.55CoolAcidIQ: You would probably want that.
04:59.21bkw_http://www.bayhamsystems.com/asterisk.html
04:59.27coppicebkw_: I have connected to SMSCs for high volume, but it only works out economic for pretty high volume
05:00.18lilneonbkw_: yeah i checked that one already..still damn expensive.. ROI not worth it.. not doing too much bulk..or not consistent with bulk number of msg's
05:00.36bkw_0.05356 per message
05:00.46bkw_wonder if you can receive messages
05:00.56bkw_I wanna get hooked up to receive messages from the smsc
05:01.00coppiceyes, you can
05:01.04IQCoolAcid: exten => 999,1,Dial(System/dsp)
05:01.04IQexten => 999,2,Hangup
05:01.14CoolAcidsure
05:01.16bkw_System/ is not valid
05:01.32CoolAcidArg.. missed that.
05:01.45IQbkw_: so its not possible to dial Console from an extension ?
05:01.45bkw_hehe
05:01.49anachronin asterisk on freebsd where are sounds stored?
05:01.52bkw_ya if you do Console/
05:01.55bkw_and not System/
05:02.01bkw_look at that for a moment
05:02.02IQbkw_: Okay, thanks
05:02.03bkw_let it sink in
05:02.06bkw_;)
05:02.25bkw_coppice, where is a cheap way to get messages from the sms network?
05:02.42bkw_anyone have the at command set to send SMS's with a motarola?
05:02.58bkw_hehe
05:03.12coppicebkw_: I can give you software for free, but they will charge to send you the messages
05:03.23bkw_coppice, thats fine with me
05:03.25bkw_I don't mind paying
05:03.36bkw_if its not over 4-5 cents a message
05:03.47coppicebkw_: all phones use the same commands - its a part of the ETSI specs.
05:04.03bkw_ok I can find it then I suspect
05:04.06IQbkw_ , CoolAcid its working now with  999,1,Dial(Console/dsp)    - thaks
05:04.20*** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net)
05:04.25bkw_newl, t-mobile?
05:04.44QwellDo people in the US actually use SMS?
05:04.50bkw_I do like crazy
05:04.54newmemberdoes the FXO module on the TDM400P handle distinctive ring?
05:04.55newlbkw_: Nope.  I work for the largest telco in the southern hemisphere. :)
05:04.57QwellYou're special though
05:05.14coppicebkw_: we used some unlimited SMS accounts, but found after about 20,000 per week they were not really free :-)
05:05.50*** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net)
05:06.07lilneonhey guys, know anywhere i can get a tdm400p card for less than 305US? FXS only..
05:06.14IQI've SprintPCS unlimited SMS... but they dont go outside US/Canada :(
05:06.22lilneonor am i better of with a channel bank and a X100p?
05:06.39PatrickDKchannelbankand x100p?
05:06.45jayeolaon my phone network, called "3", i've just discovered that sms to nigeria are not free!
05:06.49mikegrbjiop'
05:06.52mikegrbiho
05:06.56mikegrbI'm nobdrunk
05:07.02PatrickDKyour better off with a channelbank but you need a t1 or e1 for that
05:07.11bkw_coppice, hehe
05:07.17mikegrbchannel bnakl is bestg
05:07.18bkw_coppice, so how can I receive these SMS's?
05:07.42lilneonPatrickDK: what if i just wanna connect some analog phones to my asterisk box.. and perhaps 4 or so POTs from the local telco
05:08.01mikegrblilneon: just get channnnel bnak
05:08.12coppicebkw_ if you are american surely you believe SMS is just a figment of european's imaginations :-)
05:08.13mikegrbuoy owant expand  n fthei futrer
05:08.34bkw_coppice, na
05:08.39lilneonmikegrb:.. english please?..?
05:08.43mikegrbbkw_: I want lvoe you
05:08.45lilneonyeah i want to expand in the future
05:08.49PatrickDKif you are going at more than probably 6 lines,I would go channel bank
05:08.50newllysdexic. :)
05:08.53mikegrblilneon: chanlke bank
05:08.57PatrickDKthen you just plug fxo or fxs into the channel bank
05:08.58mikegrbmmm beart
05:09.31lilneonbut wont i still need a digium card or something to connect that channel bank to my asterisk box/boxes?
05:09.44PatrickDKya, t1 or e1, card
05:09.56newlOr a BRI card.
05:09.59PatrickDKunless you use a different device to connect the channelbank to anetwork
05:10.09PatrickDKhmm, channelbank over bri, how evil
05:10.15newl8)
05:10.22lilneonPatrickDK:that would depend ojn the channel bank i get..
05:10.28CoolAcidJT19: just shoot the question.
05:10.37PatrickDKthat, or really what country your in
05:10.39JT19I'm looking to replace an old Nortel system in a small office that is used to having 4 line appearances, and everyone placing / picking up calls from on hold... Is there a way to implement this with asterisk and sip? I've got asterisk doing a TON of other cool, stuff, but not sure how to do this....
05:10.49PatrickDKulaw works better in usa than alaw does
05:11.00PatrickDKcause of the different properties of them
05:11.02lilneonPatrickDK: in caribbean..
05:11.13PatrickDKyou use ulaw or alaw down there?
05:11.13CoolAcidJT19 -- look at call parking
05:12.05JT19Is there anyway to show the status of the lines on the actual phones?
05:12.10lilneonPatrickDK: honestly i don't know..
05:12.24PatrickDKlook it up online
05:12.27JT19I would like the functionality and features of asterisk, but I need it to still be like a "key" system...
05:12.30PatrickDKulaw=t1 alaw=e1
05:12.56lilneonPatrickDK: well t1..  :D.. haven't heard anything about e1's down here..
05:13.15lilneonmost companies get T1's, t2's or t3's..
05:13.18*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
05:13.57*** join/#asterisk Sedorox (~Sed@Neptune-W.client.wlgrv.pa.sed6.net)
05:14.23*** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com)
05:14.23CoolAcidJT19 - that i'm not sure.
05:15.06JT19ok, thank you for the info
05:15.42IQHow can I send Caller's Info when dialing from Console?
05:16.13JT19I was hoping i would find someone who was trying to do what I am. I've found a few phones that say they have "line appearances", but I wasn't sure if that is exactly what I needed.
05:16.31CoolAcidIQ: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID
05:16.58IQCoolAcid: this will work when dialing from console too ?
05:17.10CoolAcidIQ: Should.. it sets the CallerID of the call..
05:18.31lilneonok.. so back to my question.. where can i get a T1card at a good price?
05:18.53*** join/#asterisk blueskiesokie (~blueskies@65.242.87.151)
05:18.59IQCoolAcid: I've used it for extensions - but never for console, not sure how it will work with console... hmmm
05:19.00newllilo: Checked ebay?
05:19.15newlmeh..nick completion
05:19.18CoolAcidIQ: So try? if it doesn't work, try something else. ;)
05:19.31IQCoolAcid: :) yeah
05:21.13bkw_OMG someone send me about 40k
05:21.40brc_no
05:22.52tainted-bkw_ for what?
05:23.46brc_JT19, what you want are snom phones
05:23.59brc_then you will need to look into the 'hint priority'
05:24.12brc_but honestly, if you want a key system you probably won't be happy with asterisk
05:24.44coppiceif you want a key system, you probably won't be happy with a phone system :-)
05:24.47brc_unless people are willing to change and learn how to use a pbx
05:24.48IQCoolAcid: confused... dont know exactly where to call this method for Console :P
05:25.01CoolAcidin your dialplan
05:25.06brc_transfer calls instead of telling somebody to pickup line 2
05:25.08brc_etc
05:25.09CoolAcidbefore the dial()
05:25.14MocJT19, Polycom IP phone does Line apperance too
05:25.23brc_does it actually work?
05:25.24lilneonhey guys, would u recommend the X100P card? since digum has disconinued them..
05:25.44dmccollumI just installed one in my system and it works fine.
05:25.46brc_they are okay for playing around with
05:26.07Mocmy big issue with analog card are Sidetone !!!
05:26.26IQCoolAcid: "NO CALLER ID" - thats what my phone says :P
05:27.09lilneonbrc_: how bout planning for actual deployment?..
05:27.14CoolAcidIQ: Did you try setting the other CallerID functions?
05:27.22IQCoolAcid: actually, I'm confused about where in dialplan this will go.
05:27.22brc_lilneon, I wouldn't use em
05:27.33CoolAcidBefore the dial()
05:27.37IQCoolAcid: Yeah, for my other extensions and it work fine
05:27.40brc_but then again I don't have any deployments with less then a T1
05:30.43IQCoolAcid: exten => 999,1,SetCallerID(1234567))
05:30.43IQexten => 999,2,Dial(Console/dsp)
05:30.43IQexten => 999,3,Hangup
05:30.46lilneonbrc_: ok here's what i need.. a channel bank(6-8fxa / 4fxo) and T1 card for asterisk box..  ebay seems kinda hot.. got anywhere with good prices other than ebay?
05:31.03brc_for what
05:31.04brc_the cb?
05:31.32CoolAcidIQ: how about setcidname
05:31.57IQCoolAcid: tried that too... also setcidnum
05:32.10lilneonbrc_:um yeah.. and well i plan to move to t1 straight after a while
05:32.29brc_dunno...ebay... maybe atacomm.com
05:33.01brc_and if I may make a suggestion...buy a 4 port t1 card
05:33.13brc_you will wish you had eventually
05:34.03lilneonbrc_: yeah but those kinda HOT!!!...
05:34.14brc_hot?
05:34.15IQtried all three methods, but still can't send CID from console :(
05:34.21lilneonbrc_:expensive
05:34.27brc_BWHHAHA HAHAH AHAHHAHAHAHAH
05:34.35brc_no.
05:35.03lilneonbrc_: yeah on ebay those are like not less than $1200
05:35.20brc_it's a decent price for a 4 port t1 card...go compare with a 4 port t1 card for any other system
05:35.29brc_and buy from digium.com
05:37.19lilneonsigh.. now all i need is a good broadband provider and i will be up and running in a month.. any ideas guys?
05:39.04anachrondoes someone have a few minutes to help me get zaptel working on my FreeBSD box?
05:40.28*** join/#asterisk Moc_ (~Moc@modemcable012.47-80-70.mc.videotron.ca)
05:40.36*** join/#asterisk jeffik (~jeffik@69.158.33.127)
05:46.37*** join/#asterisk cactus1 (~fasdfsdfa@179.118.33.65.cfl.res.rr.com)
05:46.53cactus1anyone want to give me a quick hand?
05:47.53IQin sip.conf, if we set fromuser=xyz, it'll be sent to the called device as CID ?
05:48.09newlWill a SIP client calling out to PSTN stay within its context until the call terminates?  i.e. /* context stuff */ Dial(foo) /* do some more stuff here before ending context */
05:48.10cactus1im not sure
05:48.43*** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com)
05:49.21cactus1i installed asterisk from an iso i burned, and when i try to log into the web portal it asks me for a password, i tried root and all that the basic stuff and it wont log me in
05:49.34cactus1im not sure what to do
05:50.56newlerm..Asterisk doesn't come with a web portal.  Are you using something like Asterisk@home?  If so, checking the docs, faq, etc on their site would be my first stop.
05:51.25cactus1yea im using @ home
05:51.39cactus1and i printed the guidebook and it says nothing about the default web portal password
05:51.51Cherebrumhttp://www.wingtunes.com/public/songs.aspx
05:54.37cactus1oh well
05:56.28cactus1i thought this channel would be of some type of help
05:56.36lilneoncatcus1:  (To access AMP use user:maint, pass:password)
05:56.50lilneontry that one b4 u leave
05:56.58cactus1alright
05:57.18lilneonor root:root and password:password
05:57.19cactus1how did you aquire that knowledge?
05:57.26cactus1the maint worked
05:57.33cactus1i tried root already
05:57.34lilneonon asterisk wiki
05:57.43lilneonhold on i bookmarked the page
05:57.54lilneonhttp://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home
05:58.06lilneonseems the default passwords have been hacked or something like that
05:59.21cactus1hmm
06:00.10cactus1lilneon can i use a voicemodem as a device on my asterisk box?
06:01.13cactus1or anyone that wants to answer
06:01.14lilneoncactus1:not too sure.. i think so have never tried it though
06:02.07cactus1what type of device would you recommend getting so i can hook this thing up to a pots line
06:03.48lilneonum.. a X100p from digium
06:03.58lilneonor some card like that..
06:04.18cactus1hmm
06:04.37cactus1right now im going to try and figure out if i can just get away without buying anything ;)
06:04.44lilneonyes u can
06:04.57*** join/#asterisk Daishi (~daishi@pool-162-83-251-165.ny5030.east.verizon.net)
06:04.59cactus1how so
06:05.03*** part/#asterisk Daishi (~daishi@pool-162-83-251-165.ny5030.east.verizon.net)
06:05.04lilneonu can use voip  providers
06:05.07lilneonsoftphones etc
06:05.09cactus1packet8 sucks and isnt sip compatible
06:05.30lilneonasterisk uses many protocols u know..
06:05.44lilneoniax,h323 etc
06:06.11*** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com)
06:06.21cactus1this stuff is complicated
06:07.11Shido6no it isnt
06:07.19Shido6what do you want to do ultimately
06:07.48cactus1i want to be able to call my landline from my cell
06:07.54cactus1and call people on FWD
06:07.58cactus1or call out on my packet8 line
06:08.01cactus1and have voicemail
06:08.11cactus1im using asterik @ home
06:08.15cactus1asterisk*
06:08.46cactus1i want my asterisk computer to allow me to call FWD people
06:08.57cactus1but from any phone, like i call in
06:09.06cactus1ya know?
06:09.24lilneonhmm sounds cool enough..
06:09.47lilneonshould be able to do that once u get a card..
06:10.19lilneonso.. landline->card (on asterisk box)-->internet(FWD,packet8 etc)..
06:10.33cactus1they are cheep on ebay
06:10.39cactus1yea basically
06:10.41lilneonrelatively
06:11.40cactus1its getting bought tommorow
06:11.49cactus1it looks just like a modem though
06:13.14cactus1hey lilneon technically i could plug my packet8 line into the back of that x100p and pretend like its a POTS line
06:17.20bparkeranyone in here have any experience with Netlogic's IAX service
06:20.30lilneoncactus1:was away.. um.. technically.. i guess.. but ask the guys on here tomorrow.. wen most of em are awake.. i have never done it
06:20.52*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
06:21.00lilneoni connect my local telco pstn to my asterisk box and got voip provider to call long distance
06:22.42cactus1a voip provider?
06:22.49cactus1why not just use the voip phone itself?
06:23.11*** join/#asterisk pulu (~chatzilla@64.200.224.158)
06:24.31lilneoncactus1: i need the voip provider to terminate my calls oversees..
06:25.03cactus1well dont you have like an adapter that plugs into your router
06:25.07cactus1and then plugs into a phone
06:25.34cactus1vonage, voicepulse, and packet8 all give you that stuff
06:26.22lilneoncactus1:yeah basically.. u use them to terminate your calls over seas.. i have slightly more needs than just one analog phone connected to my asterisk box
06:27.47cactus1alright
06:27.50cactus1well im going to go to bed now
06:27.55cactus1night
06:29.34IQHi, How can I override user's entered "User Name" ? Like, I dont want SIP clients to be able to change their names, is it possible?
06:32.19*** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net)
06:32.23sudhir492hi there
06:33.20IQhi
06:36.23*** join/#asterisk rvhi (~rv@66.175.65.89)
06:37.02rvhiin dial by directory, can i have to try both first and last name?
06:37.09rvhicurrently seems to be either or
06:38.33*** part/#asterisk lilneon (~tj_r3@cuscon12874.tstt.net.tt)
06:41.36*** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net)
06:44.38IQHow to override SIP client's "User Name"? I dont want them to change the name I assign - help ?
06:45.23NewSolein sip profile
06:45.26NewSoleon server
06:45.35NewSoleuse
06:45.46IQon server side
06:45.55NewSolecallerid = Some User <6665551212>
06:46.11IQin sip.conf ?
06:46.28NewSoleyup
06:47.39IQNewSole: It works... appreciate your help!
06:47.46NewSolenp
06:48.10IQNewSole: would you have a link to all available sip.conf commands? Thanks again for the help :)
06:48.54NewSolehttp://www.voip-info.org/wiki-Asterisk+config+sip.conf
06:49.12IQThanks
06:50.09NewSolenp... wish I was as lucky to get help with what I need...
06:50.43IQI wish I can help... ask for my life :)
06:52.00NewSolewell my problem is not with asterisk its self.... I am making a Module that does call routing and accounting for prepaid and post paid systems
06:52.02Shido6heh
06:52.10Shido6yeah thats consulting
06:52.15Shido6and ppl will charge you
06:52.33Qwellcha-ching :p
06:53.04IQNewSole: C++ ?
06:53.35NewSoleShido6... even that would not help... I am sure the $$ is there the backers like what I am doing and want it done fast...
06:53.50NewSoleits just working out the bugs
06:54.54*** join/#asterisk clh (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net)
06:56.57NewSoleShido6... what are you up to tonite
06:57.03IQNewSole: what kind of routing?
06:57.13Shido6chillin wiff the fam
06:57.17clhas far as the topic is concerned, shouldn't the person that broke it fix it?
06:57.35Shido6then returnin phone calls in the morning
06:57.42Shido6knock out a few sessions
06:58.04Shido6and write up a marketing plan for hosted services and new products
06:58.08NewSolewell.. I created a database with 4 different providers.. and the module finds the cheapest provider and routes the call
06:58.12Shido6then execute in the new week
06:58.47*** join/#asterisk laternight (laternight@cpe-68-201-251-41.houston.res.rr.com)
06:58.55clhlaternight : hi there
06:58.59NewSoleI spent the last 6 months doing that Shido6... finding investor
06:59.01laternighthi there
06:59.02IQNewSole: can't we just create a dial-plan to do all this ?
06:59.22*** join/#asterisk Martz (Martz_test@62.3.201.11)
07:00.33*** join/#asterisk Los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net)
07:00.33*** join/#asterisk Los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net)
07:00.34NewSoleya but problem is it has to manage the 4 providers and the rates change on a daily basis
07:01.10NewSolethe database is updatd evey nite at 4am
07:01.27IQNewSole: can't we update the dial-plan at 4:15am - automatically :P
07:02.17NewSoleya... you could... but in my case I would have to update 5 systems every nite...
07:02.36IQNewSole: 5 systems with same dial-plan, right ?
07:02.37Los415write a script
07:02.51Qwellhello, IAX trunking?
07:02.58QwellThats what its for, right?
07:03.08sudhir492anyone of you using h323?
07:03.21sudhir492what is preferred: h323 or oh323?
07:03.30NewSoleno not same scripts... 3 servers have PRI's and 2 dont
07:04.28IQso u need 2 scripts - slightly different
07:04.30sudhir492I have been using oh323 since Apr last year and it has been fairly stable. But do not know the status now - i.e. which one is more robust on newer version of Asterisk
07:07.00NewSoleI did not ask for help....
07:07.34QwellThen I have no regrets putting you on ignore.
07:09.19NewSoleo well some peoples kids
07:10.03PTG123hey qwell found a fix for the sip problem
07:10.07Qwelloh yeah?
07:10.46PTG123yah
07:10.55PTG123you would think i woul dhave been smart enough to save the url though :)
07:10.56PTG123one second
07:10.59Qwellheh
07:12.14PTG123http://loans.way2fast.com/patch.txt
07:12.18PTG123apply that patch
07:12.23QwellI find it funny when you dumb down an explanation for somebody...
07:12.34Qwelland they turn around and start using technical terms about the subject matter
07:12.50QwellI called a coworker today, to ask what my CID was showing up as...dumbed it down a bit
07:13.00Qwelland he turns around and explains that his internal DID was showing up
07:13.11PTG123heh
07:13.20PTG123hey what happened to our calling plan today? :)
07:13.33Qwellfamily has been here all day
07:13.42QwellSunday?
07:14.10PTG123easter? :)
07:14.15Qwelldamn
07:14.22QwellMonday it is then
07:14.25PTG123i am sure that would work :) not :)
07:14.41QwellI don't even know when holidays are coming up...
07:14.47Qwellwe get like...6
07:14.50PTG123you have kids now
07:14.55PTG123you gotta know that stuff
07:14.56PTG123:)
07:15.04Qwellyeah, so in about...5 years, when he starts going to school... :p
07:15.24PTG123no man the easter bunny is coming tommorow night
07:15.34Qwellhe comes at night?
07:15.39PTG123yes
07:15.44Qwelllike santa?
07:15.49PTG123and leaves easter baskets and eggs hidden for your son
07:15.51QwellThat explains the gifts then...
07:15.56PTG123you don't know that? :)
07:16.20Sedoroxanyone use lingo?
07:16.26Qwellhave you ever heard of "easter gifts"?
07:16.30PTG123yes
07:16.32Qwellcoworker told me about those yesterday
07:16.35PTG123heh
07:16.41PTG123were your parents athesists?
07:16.43QwellI was like, "wtf is an easter gift?"
07:16.49PTG123non holiday loving ones at that?
07:16.54Qwellthe easter bunny was the devil :P
07:16.55SedoroxPTG123: do you have any tips on setting it up? or a howto?
07:17.04Qwellhe killed jesus...or something
07:17.09PTG123Sedorox, why lingo?
07:17.10QwellI never really paid much attention to my parents
07:17.11Sedoroxor you answering Qwell
07:17.20SedoroxI'm helping someone set it up...
07:17.30PTG123Sedorox, will it even work with asterisk?
07:17.39PTG123Sedorox, i would use an asterisk friendly provider
07:17.41Sedoroxdunno
07:17.48Sedoroxits a little late :-p
07:18.04PTG123cancel :)
07:18.14PTG123Qwell, its your job to spoil that kid
07:18.17Sedoroxaha
07:18.57Qwellbrb, gonna patch this up
07:19.38*** part/#asterisk clh (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net)
07:19.44*** part/#asterisk laternight (laternight@cpe-68-201-251-41.houston.res.rr.com)
07:23.31*** part/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net)
07:25.57QwellPTG123: Wanna go to aim?
07:26.10PTG123sure
07:26.15PTG123oh shit :)
07:26.30PTG123nah use irc
07:26.39PTG123aim is open underneath countrstrike on my other pc
07:26.40PTG123:)
07:29.10Sedoroxhow does teliax work for most pople... good?
07:29.30PTG123yah its not too bad, what kind of account are you looking for?
07:29.47Sedoroxresidential
07:30.00PTG123which state?
07:30.04PTG123like the unlimited plan or what?
07:30.17SedoroxMA. and I would assume so.. this is actually for someone else
07:30.17Sedorox:-p
07:30.31Sedoroxjust seeing if people have had luck with it
07:32.22PTG123sure try em out :)
07:32.56Sedoroxkk
07:34.03IQIf we get "Pay as you go" package from Teliax - they charge 2 cent/incoming-call? (A two cents connection charge applies unless call is from another teliax user)
07:34.42Sedoroxhmm
07:34.56PTG123ya
07:35.58IQnot bad
07:47.53*** join/#asterisk pulu (~chatzilla@64.200.224.158)
08:11.12sudhir492Anyone isomg chan_h323 here?
08:11.22sudhir492using
08:31.53*** join/#asterisk AlexW (~AlexW@adsl-46-159.swiftdsl.com.au)
08:40.49*** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com)
08:46.05*** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net)
08:46.07SexyKenHey gulys.
08:46.09SexyKen*guys.
08:46.14SexyKenSorry, a little intoxicated.
08:46.15SexyKenAnywho.
08:46.18SexyKenAnyone want a job?
08:47.15bonez39I need a job..heheh
08:48.44*** join/#asterisk mbranca (~matteo@host-84-222-14-188.cust-adsl.tiscali.it)
08:54.32*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
08:55.35SexyKenbonez, what can you do for me?
08:56.17Sedoroxdoes it pay?
08:56.31SexyKenYes.
08:56.42Sedoroxhmm
08:56.47Sedoroxinvolves?
08:57.05SexyKenSetting up an entire asterisk system
08:57.19Sedoroxfor how many people/extentions and hardware? :-p
08:58.31SexyKenIt's a remote job.
08:58.34SexyKenWork from home.
08:58.38Sedoroxnice
08:58.45*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
08:58.46Sedoroxany T* hardware involved? lol
08:58.49SexyKenI need many extensions and a GUI to mange everything.
08:58.50SexyKenNo.
08:58.52SexyKenOIP only.
08:58.56SexyKenVIPO.
08:58.58SexyKenVOIP
08:59.00SexyKenDamn alcohol.
08:59.01Sedoroxnice
08:59.07Sedoroxgui = amp
08:59.08Sedorox:-p
08:59.18SexyKenI need 3 companies to be allowed on asteriskl.
08:59.29Sedoroxhell.. I'd be interested in it
08:59.37SedoroxI run three asterisk boxes right now
08:59.51SexyKenI need a single asterisk box running 3 companies.
09:00.02Sedoroxpushing a lot...
09:00.11Sedoroxsounds easy enough tho...
09:00.29SexyKenFor instance, companies a, b, c would be able to have separate extenions ( each could have an extension 200, just going to different phones)
09:00.38Sedoroxyea
09:00.40SexyKen7k/mintues a week max. (total)
09:00.42newlEverything sounds easy until you hear the details. :)
09:00.50Sedoroxnewl: true
09:01.14SexyKenNeed user/extension management.
09:01.28SexyKenNeed to be able to view complete cdr shit.
09:01.29SexyKenetc.
09:01.30SedoroxI'm interested in it...
09:01.38Sedoroxbut just don't know the cdr stuff... yet...
09:01.46Sedoroxand the management stuff...
09:01.48SexyKenWell it's not a 3 month project.
09:01.51Sedoroxonly stuff I know of is amp
09:01.59BoRiSof course
09:02.00SedoroxI know.. probably more like 2 weeks
09:02.00Sedorox:-p
09:02.01SexyKenIt's something I needed done 3 months ago.
09:02.12BoRiSBut again...depends on how much :)
09:02.19SexyKenHow much what?
09:02.20Sedoroxthatgs true too
09:02.21Sedorox:-p
09:02.25Sedoroxto do this
09:02.29SexyKen$2k.
09:02.33Sedorox0_o
09:02.35BoRiSkeep going
09:02.44Sedoroxlol
09:03.01BoRiS:)
09:03.08SedoroxI need money.. so...
09:03.08Sedorox:-p
09:03.13newlDo four deals like that a month and you'd be a happy camper. 8)
09:03.36OthelloI'm getting pissed off that the console driver is not working for *...
09:03.45SexyKenI need it done, yesterday. GUI with Comopany Management, Extension Management (add/edit/delete etc) should be able to add IVR's through the stupid gui thing. Should be able to view entire call details based on queues, agents, etc.
09:03.46Sedoroxyea.. really...
09:04.02*** join/#asterisk jskcr (~kvirc@jskcr.user)
09:04.10Sedorox:-p
09:04.28SexyKenAMP doesn't work for multiple copmanies ;-)
09:04.37Sedorox:-p
09:04.59BoRiSAmg....Ugh
09:07.05SedoroxBoRiS: you do the gui.. I'll do the rest.. and we split
09:07.06Sedorox:-p
09:08.12BoRiShehe
09:08.21BoRiSI prefer the rest "other" then the gui
09:08.35Sedoroxlol
09:16.05Sedoroxallwell...
09:39.08*** join/#asterisk _Omer (~eegg@202.38.63.84)
09:40.10_OmerMy question is not related to Asterisk but I want to know. What things are required to develop FREEWORLDDIALUP.COM (FWD) kind of thing??
09:48.36*** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net)
09:55.40CleanerXnew linphone is out
09:55.43CleanerX1.0.x
11:18.45*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
11:22.03*** join/#asterisk loxza (~loxza@118.Red-81-33-60.pooles.rima-tde.net)
11:29.02*** join/#asterisk pluto70 (~me@80.70.179.76)
11:31.55*** join/#asterisk cjk (~cjk@80.92.75.145)
11:32.35cjkhi, can sip an iax RTP streams bridge? i mean, if i have canreinvite set to yes, will the rtp stream go through * ?
11:51.46*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
11:51.46*** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || cdr_addon_mysql.c with 1.0.7 == DEADLOCK someone that cares needs to fix it.. because I don't (bkw_)
11:57.22*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
11:57.49*** join/#asterisk ckruetze (~nospam@i3ED63E95.versanet.de)
11:58.07*** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net)
12:02.55*** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net)
12:03.42cjkmbranca, never
12:03.58cjkmbranca, forget about portforwdings, i never got one phone working because of portforwarding
12:04.03cjkthats for sure not the solution
12:04.17cjkmbranca, there are several nats where it just does not work.
12:04.24cjkbut its working on most nat's
12:04.39*** part/#asterisk loxza (~loxza@118.Red-81-33-60.pooles.rima-tde.net)
12:07.40*** join/#asterisk alphaque (~Alphaque@218.208.239.39)
12:07.48*** join/#asterisk DannyF (~wizardone@h163n9c1o848.bredband.skanova.com)
12:08.48mbrancacjk: letting rtp flowing through asterisk made sip devices working behind any kind of nat (at least in my experience)
12:08.59mbrancaof course there's a bw problem
12:09.09mbrancabut depends on what you need to do
12:09.18mbrancaand without any sort of port forwarding
12:09.38cjkmbranca, thats true for several nats
12:09.52cjkbut its not necessary for many nats
12:10.06mbrancatill now I never experienced any problems
12:10.22mbrancaand we have a rather large customers base :)
12:13.03Mw3hi folks, have you seen any ata which supports vpn ?
12:29.00Mavviehttp://www.voip-info.org/wiki-Asterisk+cmd+SrxEchoCan <- /me needs it for SIP channels :-/
12:36.35*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
12:37.26*** join/#asterisk WilliamK (~wkeller@c-24-0-130-177.client.comcast.net)
12:56.44*** join/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter)
12:59.31*** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net)
13:01.18*** join/#asterisk jeffik (~jeffik@69.158.33.127)
13:05.48cjkmaybe someone know here, if i do an sip-2-iax call or iax-2-sip (both set to canreinvite=yes) will the voice traffic pass my * or not?
13:06.17cjkmaybe someone knows here, if i do a sip-2-iax or iax-2-sip phone call (both set to canreinvite=yes) will the voice traffic pass my * or not?
13:06.18Zeeekit would seem logical
13:06.27Zeeekthat it would since the two devices can't talk
13:07.04ckruetzecjk it will, since you sip phone doesn't speak iax and your iax phone doesn't speak sip
13:07.50*** join/#asterisk mmlj4 (~looseduk@ip68-14-39-201.no.no.cox.net)
13:07.59cjkyeah but i thought maybe the voicestream would be the same or something like that
13:08.12Zeeekthe ports used are totally different
13:10.07Zeeekso why do you wonder?
13:10.15Mavviewell, reload doesn't work on it.
13:10.33Zeeektry a restart and you'll have the definitive answer
13:10.40Mavvieoh! reload only does re-read the configuration files.
13:10.47Mavvieit doesn't reload the module.
13:11.39Zeeekmakes sense. Thank you for sharing that info with us.
13:12.24*** join/#asterisk queuetue (~Scott@h69-21-252-54.69-21.unk.tds.net)
13:12.30*** part/#asterisk sysdef (~sysdef@sysdef.admin.debiancenter)
13:15.14*** join/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net)
13:15.18Mother_multiple greetings
13:17.07*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
13:18.47cjkdoes sip-2-iax and iax-2-sip translation use alot of cpu?
13:20.41queuetueI'm trying to set up iax between two servers, and on the north server, I see "Peer 'south peer' is not dynamic (from x.x.x.x)" ... What does this mean?
13:21.28queuetueThe south server reports "registration refused" ...
13:21.37Mother_check the host setting in iax.conf
13:21.43Zeeekcjk my logic may be faulty but if I needed to worry about this, do it all in IAX. Obviously, we can't dictate what protocols users have, though
13:21.52Mother_I have mine set to the remote IP
13:21.58Mother_hi Zeeek
13:22.01Zeeeklo ma
13:22.12queuetueNow, north to south, the connection seems fine, and the two iax.confs are copies of each other with hosts swapped ...
13:22.28queuetueMother_, The settings on which server?
13:22.49queuetueWhat does s"not dynamic" mean, in this context?
13:22.54Mother_on both, I have a similar setup, two servers each behind their respective NATed DSL lines
13:23.39Mother_well I'm not sure, I've never seen that message, however I can tell you that setting host=x.x.x.x instead of dynamic doesn't cause me any problems
13:23.44Mother_I have fixed IPs however
13:24.17queuetueMother_, As do I - neither of these servers is dynamic.
13:24.38Mother_and have you got the IP address in the host parameter in iax.conf?
13:24.43queuetueand I have host=x.y.y.z on both servers.
13:24.48queuetueyes.
13:25.04Mother_and your ports are mapped OK if you have routers/NAT/etc?
13:25.24queuetueMother_, Yes, but I will test again.
13:25.36queuetueThe wierd thing is that one connection (north->south) works ... south->north does not...
13:25.40Mother_you can use iax2 debug to see what is happening
13:25.50queuetueMother_, How do I do that?
13:25.51Mother_in the CLI
13:25.55queuetueAh.
13:25.58Mother_CLI> iax2 debug
13:28.28Mother_gotta go eat some lunch
13:28.30Mother_bbiab
13:33.24cjkif i have a sip and an iax client, both doing ilbc, will * have a lot of work to translate?
13:39.28DaminIt is way to early in the morning..
13:41.57tzangerthis is fucking terrible
13:42.05mmlj4does the cheap intel modem that works as an FXO also do the timing that's needed for IAX links?
13:42.12tzangerall the legal music download sites want you ot register before you can SEARCH their archives
13:44.31Zeeektzanger why is that so terrible?
13:44.46tzangerwhy should I give anyone my credit card information in order to see if they have the music I want?
13:45.02Zeeekregister means giving cc number?
13:45.05tzangerif I want to purchase from them, sure.  But to search?  fuck that.
13:45.07tzangeryup
13:45.12Zeeekscew that
13:45.16Zeeekscrew
13:45.20tzanger"You can cancel if you don't want to buy anything from us"
13:45.34ZeeekI never give a number unless I know I'm gonna buy and that the place is cool
13:45.40tzangerexactly
13:45.41Zeeeksounds BOGUS to me
13:45.46*** join/#asterisk das1 (~dh@cau138.neoplus.adsl.tpnet.pl)
13:45.55Zeeekno one would do that with *their own* cc number :)
13:46.05tzangermindawn.com looks cool but the selection blows
13:46.10tzangerogg/flac formats
13:46.16das1does someone have an idea on this: 2 asterisk calling each other, one is immediately hanging up after haveing received ack and first ring.  1.0.7 debian SID the not working one
13:46.31tzangerand itunes wants me to download software to search
13:46.37tzangerwhere's bkw or file when you need them
13:46.48tzangerI bet both those buggers have itunes accounts, they can search that for me
13:46.57mmlj4das1: is either behind a NAT connection?
13:47.06das1yes
13:47.18mmlj4using SIP, or AIX?
13:47.27das1but if you start from scratch it's working few minutes
13:47.31das1iax
13:47.40mmlj4hmm... no clue, then
13:48.06tzangerallofmp3.com lets me search without registering, but they don't have the artist either
13:48.08das1mmlj4: thanks
13:48.12mmlj4SIP + NAT = problems, as we know
13:48.20tzangerSIP + Internte = problems
13:48.23Zeeekwhat are you looking for, Perry Como ?
13:48.25tzangerer internet
13:48.33tzangerZeeek: hahaha  no, Kevin Laliberte
13:48.39Zeeekcanadian?
13:48.42tzangeryes
13:48.50Zeeektry this: virgin.fr
13:48.57Zeeekpretty good search engine
13:49.14Zeeekbut wait that's not free
13:49.21tzangerit says I need Win98 to even get on to the site
13:49.26Zeeeknah!
13:49.28dantlol
13:49.28tzangerat least fromw hat I can tell from my broken French
13:49.35Zeeekno way!
13:49.43jeffiktry jazzmusique
13:49.48tzangeryup change konqueror's id and it goes
13:50.37Zeeekthat don't have Kevin Laliberte
13:50.41Zeeekanyway
13:50.45tzangeruh virgin.fr's search is completely busted
13:50.57Zeeekworks great for me with Firebird
13:51.00tzangerI put in "Laliberte" in the search and I get 16 pages of hits, none of which have "Laliberte" in them
13:51.12tzangerthey do have Mel Torme though..  hahaha
13:51.18tzangerjazzmusique?
13:51.36Zeeektzanger ya want free or to pay?
13:51.48tzangerI don't mind paying for the album at all, I just can't seem to find it online
13:52.18tzangerjazzmusique has streams, not music to buy
13:52.26jeffikyes streaming although only 3 formats but is free
13:52.47Zeeeksony connect, msn music club, tiscali music, jam label, uploud, mp3tunes.com
13:53.00*** part/#asterisk das1 (~dh@cau138.neoplus.adsl.tpnet.pl)
13:53.07Zeeekyou can easily record streams...; on WIndows :)
13:53.22jeffikok then subscribe to ripcast/shoutcast it's cheap, 100s o genras of streaming and you can captue it using ripcast
13:53.28tzangerI can easily do it with LInux and mplayer, but I don't want to record streams, I want to download the album
13:53.43tzangerjeffik: I want to download an album, not a random stream
13:53.45Zeeeknobody has heard of it around here!
13:53.56tzangerZeeek: he's relatively indie
13:54.10tzangerhttp://www.kevinlaliberte.com/
13:54.20Zeeekhttp://newmusic.clearchannel.com/artist/kevinlaliberte/song/14500
13:54.51Zeeekhaha "You must be a New Music Network member to perform that function."
13:54.51tzangerheh
13:54.53Zeeekthe net ain't what it once was bioz
13:55.01Zeeekboyz
13:55.14jeffikboiz
13:55.25Zeeek<PROTECTED>
13:55.41jeffikyes all the free services either went to pay or went away
13:55.44tzangerPeter Smith Quartet?
13:55.53Zeeekhe's on that
13:55.55tzangerjeffik: like I said, I'm looking to BUY the album
13:56.12Zeeekhttp://www.guitarnoise.com/news/2003/20030914.html
13:56.30jeffikZeeek: just trying to help you find some music
13:56.44Zeeeknot me - Is the album elation?
13:56.55tzangeryes
13:58.25tzangerI can order the physical CD online but that doesn't get it here for tomorrow :-)
13:58.47Zeeekyou're right, it isn't easy to download that CD
13:58.59Zeeeklooks good, too
13:59.04tzangerI'm going to go to Kitchener today to see if I can't find it on a shelf but it may prove difficult
13:59.16tzangeryeah I discovered that I like it quite by accident
14:01.34Zeeekapparently you can download two songs free here: http://newmusic.clearchannel.com/
14:01.39Zeeekmembership is free
14:01.41tzangeryeah
14:01.48Zeeeklooks like a decent site actually
14:03.31tzangerI hate "enter your email address....  now enter it again"
14:03.34tzangerI just copy and paste
14:04.01Zeeekme too - some idiot thought that was a good idea
14:04.10tzangerwell for people who can't type, sure
14:04.25Zeeekespecially since my address on those sites is always something-mykey@sneakemail.com
14:04.33tzangerhehe that is EXACTLY what I do
14:04.35tzangerEXACTLY
14:04.39Zeeeknew address for each account
14:04.41tzangeryup
14:04.53Zeeekthe only trouble is remembering it when trying to log in :)
14:04.58tzanger:-)
14:05.12Zeeeksneakemail with auto address keys is the greatest thing since asterisk
14:06.02tzangerheh
14:06.10Zeeek"Dear FuckSpam-UpYours@sneakemail.com. It has come to our attention..."
14:06.31tzangerI have my own domain though
14:06.39Zeeek"Our client, Mr. Fuckspam has passed away leaving you..."
14:06.41tzangergraylisting is the best
14:07.05ZeeekI have my domains too, but I never give that out
14:07.34tzangerhmm that elation tune was the one I really wanted but I still want the rest of them too
14:07.35Zeeeknot since the domain registrars spammers CD captures
14:07.53ZeeekSo sing in to the network thing, they have elation on download
14:07.56Zeeeksign
14:08.00tzangerit was funny I just got finished practising guitar for about an hour when I heard that song and I said "damn... I don't sound anything like that"
14:08.09tzangeryes I know I already downloaded it
14:08.12tzangerand the other too
14:08.20Zeeekcool!
14:08.27Zeeekyou play guitar?
14:09.33Zeeekcheck this: http://resmo.com/rare/mp3/logjam.mp3
14:12.28*** join/#asterisk Darwin[laptop] (~darwin-la@c-24-3-226-147.client.comcast.net)
14:12.42*** join/#asterisk Blackvel (~blackvel@dsl-082-082-059-204.arcor-ip.net)
14:13.26tzangerZeeek: yeah but I'm nowhere near as good as him
14:14.18Zeeekas Kevin?
14:14.41tzangeryeah
14:14.43ZeeekKevin looks like he's world class - played with a lot of great people
14:14.45tzangerI have never tried playing latin guitar either
14:14.49Mother_as for the giving out a CC to sites, I have one of these "virtual" CCs that's only a number
14:14.58Mother_the credit is exactly zero euros
14:14.59tzangerMother_: really?  that's a good idea
14:15.16*** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net)
14:15.30tzangerZeeek: yeah I am taking this lady I'm seeing down to toronto to see him next month
14:15.33Mother_yep, and since I can get as many of these virtual CCs as I want, I keep one for these one-off sites and payments
14:16.02tzangerI knew this song sounded familliar (then and now)
14:16.04ZeeekMother_ that's a little like the email I was talking about. How can we get these?
14:16.09Mother_when I have to purchase, I recharge the exact amount, this can be done on ATMs and bank branches
14:16.24tzangerwhy's logjam seem familliar
14:16.40Zeeekyou listened to logjam?
14:16.40Mother_well my bank is http://www.lacaixa.es the only problem you may have is the recharge
14:17.23Zeeekwe have that bank in France too
14:17.32Zeeekthere's one down the street in fact
14:17.43Mother_http://portal1.lacaixa.es/Channel/Ch_Redirect_Tx?dest=1-30-10-00000
14:17.57Mother_that's the actual card, "Cybertargeta"
14:18.17Zeeekvery good idea
14:18.28tzangerit doesn't bother me
14:18.41tzangerat least in canada and the us you are only liable for $50 on your CC if hte number's been stolen
14:18.48Mother_only problem is the initial recharge has to be done at an office or ATM of this bank
14:18.48tzangeryou don't need to pay extra insurance or anything
14:18.57tzangerif someone racks up $20000 on your card you pay $50 and that's it
14:19.14Mother_that's not bad, here you either buy extra insurance for the card, or it's up to the time when you report it stolen
14:19.26tzanger*nods*
14:19.42Mother_I have a friend who lost some $4000 and was able to get some back only after suing the bank
14:20.07robl^tzanger: only for *now*  part of the new Bankruptcy re-form the current administration is tryin got pass includes removal of some of that protection.. and prevents you also from removing creditcard debt if you file bankruptcy
14:20.58Zeeekthere must be some advantage to living in Canada now that there's no draft in the US anymore :)
14:21.07Mother_lol
14:22.00ZeeekI suppose one might say that Canada combines the best parts of UK and American culture... which is little the opposite in Oz :)
14:22.30Zeeekjust kidding guys... no offen[cs]e
14:23.01ZeeekMother_ you're in Spain, eh? I never noticed
14:23.10Mother_yep :)
14:23.58Blackvelwhat is the easy way to send "ANSWER" inside AGI?
14:24.07robl^Zeeek: true!  at least Canada doesn't have Cricket.  that game is evil. each game takes an average of 7.23 years to complete..  just wathing guys running back and forth with little sticks shoved in the ground
14:24.22Mother_slight doubt: what would be a good tool to batch convert PCM recordings to mp3 that I could stick in cron?
14:25.08tzangerrobl^: yeah i know
14:25.26Mother_robl^: agreed, I fell asleep the two times I tried to watch a cricket match
14:25.34tzangerrobl^: the US is heading to hell in a handbasket and awfully damn fast too...  it's like the city's on fire and nobody's able to see the flames
14:26.24*** join/#asterisk Chuji (Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net)
14:26.36robl^tzanger: I think plenty of people see it, but are put into a state of "terra" and are too scared to do anything about it
14:26.39tzangerI dunno...  so much is on credit these days it can only go on so long it's just like the dot-com bubble
14:28.01*** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com)
14:28.11tzangerI mean I'm not TOOOOOO bad off, I have a line of credit and a mortgage... that's it.  my credit card debt is miniscule (maybe $500).  If things were to go sour I'd lose the house and that might be nasty, especially if the real estate market pops, but I don't think it'd be toooo terribly bad.  I have *land*, not just a house, and property has intrinsic value.
14:28.49tzangerI don't have much land, but it is enough to build 6-8 houses on by today's standards, so that's a saving grace
14:29.51newlSubdivide and conquer!
14:30.12tzangerhehe  I'm 4km from town
14:30.18robl^tzanger: well.. what if you loose yer job and a family member gets very sick and has to be in the hospital for an extended stay?  a lot of credit debt in the US at this point is due to lack of healthcare insurance for so many people.
14:30.21tzangeryou could do it but yeah
14:30.39Zeeekplant a server farm!
14:30.43Mother_here they stats are that only 35% of people can save up something at the end of the month
14:30.50Mother_the rest just manage to get by
14:31.02tzangerrobl^: lose my job, yes that's a problem.  I'm trying to get the LoC paid down so I can indeed survive at my current rate of spending for four months.  family member sick -- I have insurance for that
14:31.10Mother_and the trend is visible when you see TV ads for cars where you can pay them in no less than 9 years!!!
14:31.20tzangerMother_: yes that is spooky
14:31.51Mother_I mean a car will be worth 1/2 a few months after you buy it....that's just wrong
14:31.59tzangeryup
14:32.11tzangerI've never purchased new and likely never will due to the extreme depreciation
14:32.21robl^tzanger: if you loose your job, you likely loose healthcare benefits too..  that's the problem.
14:32.24Mother_I already know cases of people who have had the car stolen or wrecked and will have to repay it during a few more years, while having to buy another car
14:33.14tzangerrobl^: I have health insurance through the company that is true, but my mortgage is insured and my life insurance covers *my* health problems to a certain degree.  i am not saying I'm safe by any means but I do have some coverage
14:33.46Zeeekfwiw, our coverage goes on for at least a year during unemployment
14:34.00Zeeekit's state operated, not private
14:34.01robl^tzanger: that's good to h ear.  unfortuantely you are much better off than a large number of americans
14:34.21tzanger*nods*  I realize that and I do try to work hard to maintain it.  :-)
14:34.27tzangerthere are many, many who are worse off than me
14:34.37blitzragewhat is a keytree, and how does it differ from a key family?
14:34.53Zeeekthat's where the keys grow, blitz!
14:35.31Chujiblitzrage : Does this look like an Asterisk help channel? It's debt/motivation channel... sheesh
14:35.41Zeeekyeah!
14:35.49blitzrageoops.. sorry
14:35.57Blackvel510 Invalid or unknown command
14:36.02Zeeekdo your own research, blitz! There is a site that has some good stuff....
14:36.11Chuji/join #a$terisk
14:36.12Zeeeksome .org thingie
14:36.16Blackvelwhy does this happen? I am sending the command "ANSWER"  of the AGI channel
14:37.02blitzrageZeeek: people who say to go and do their own research say so because they don't know the answer :D
14:37.08blitzrageZeeek: it's coming
14:37.19Zeeeknah, when I say it, I give the answer too - I'm no crtich !
14:38.00ZeeekI owuld like to announce a monument day today, a landmark in asterisk history
14:38.20Chujiheh, the legacy of critch
14:38.29ZeeekI got the answer to a problem I repeatedly submitted to EDigium support and here for weeks...; on the mailing list!
14:38.44blitzragewhat was it?
14:38.59ZeeekJEEZE
14:39.07Zeeekthe question or the answer?
14:39.22Zeeekthe question was, why no callerid on ONE recent phone
14:39.36Zeeekthe answer was, change the ring cadece
14:39.41Zeeekcadence
14:40.01blitzragehrmmmm, crazy
14:40.09blitzragedamn non-north americans
14:40.11blitzrage;)
14:40.13*** join/#asterisk Landim (~daniel@200.157.72.19)
14:40.23Zeeekas to your query blitz, I'd have guessed they were the same thing written by different people.
14:40.36Zeeekfamily here, tree there
14:40.43blitzrageZeeek: possible
14:41.06Zeeekunless there are multiple trees of which the / we see is only one?
14:41.29ZeeekI am not intimate enuf with source to know
14:42.38LandimHi All, I need some help with Zap Quad E1 card configuration. When I try to make a call I receive this msg: Unable to create channel of type 'Zap'
14:42.39Landim<PROTECTED>
14:43.08Zeeekblitz : snprintf(prefix, sizeof(prefix), "/%s/%s", family, prefix);
14:43.32Zeeek<PROTECTED>
14:43.37Mother_maybe everyone *is* busy and congested
14:43.41BlackvelI try to send over AGI ANSWER + NEW_LINE. my java application sends ANSWER\r\n instead of ANSWER\n
14:43.44ZeeekI've been congested recently
14:43.46Blackvelwhy is that a problem?
14:43.50LandimBut all are free.
14:44.02Mother_me too
14:44.12LandimI'm setting up the system right now!
14:44.20Mother_btw I found the notlame encoder, it works off the command line and is very fast
14:44.32Zeeekand has a great name
14:45.02Mother_so now I can compress all the call recordings to mp3 at night when the system is idle :)
14:47.56ChujiMother_ : From what to mp3?
14:50.04*** join/#asterisk fenlander (~irc@82.152.81.57)
14:50.42Mother_Chuji: PCM recordings
14:50.55Mother_from calls
14:51.08*** join/#asterisk logicalonline (~logicalon@border.logicalonline.com)
14:51.17Blackvelhow can I send this over AGI: EXEC DIAL(ZAP/g1/90,30,tr)?
14:52.11Mother_dwmw2: then why don't you setup a cron that compresses and then emails?
14:52.30dwmw2Mother_: I meant voicemail. It's emailed automatically by app_voicemail isn't it?
14:52.35dwmw2I just need to implement format_ogg
14:52.59dwmw2but I'll do that after I have chan_bluetooth working properly
14:53.26Mother_hmmkay
14:53.42Mother_you can still do without the built-in email notifications and cook your own
14:53.48dwmw2true
14:53.50Chujidwmw2 : You working on chan_bluetooth?
14:53.56dwmw2Chuji: yeah
14:54.02dwmw2I have it working with * CVS and full-duplex
14:54.04ChujiCool, how's it coming?
14:54.09dwmw2but there's something wrong with the incoming audio still
14:54.13dwmw2it's very bizarre.
14:54.21Mother_what bluetooth hardware are you using?
14:54.23ChujiGroovy, glad it's getting some attention
14:54.49dwmw2there's distortion I can't quite characterise, and an echo whose delay is proportional to the size of the ring buffer we use between the sco_thread() and blt_read/blt_write
14:55.01dwmw2if I feed frames directly from blt_read to blt_write I get perfect echo
14:55.08dwmw2using a Plantronics M3000
14:55.18Mother_ok
14:55.18dwmw2and an Ericsson T630
14:55.35dwmw2the mailing list seems dead
14:57.11*** join/#asterisk point (1000@213.27.44.55)
14:57.27*** join/#asterisk yaboo (~jsirucka@220.245.131.131)
14:58.11dwmw2I'm utterly confused by the echo
14:58.38dwmw2I've fixed it to memset the contents of the ring buffer to zero after reading them out. We really shouldn't be hearing them again
14:58.46cypromisECHO ECHo ECho Echo echo
14:59.06dwmw2<PROTECTED>
15:02.57Zeeekanyone have a good hardware recommendation for a small quite asterisk box?
15:07.03mmlj4there's that linksys wireless thingie, Zeeek
15:07.11mmlj4small and cheap
15:07.23dwmw2that'd be cute. Is it little-endian?
15:07.32dwmw2asterisk is, erm, 'fun' on big-endian machines ATM.
15:07.35mmlj4no idea
15:07.43dwmw2I'm running it on my powerbook.
15:07.48Zeeekthe WRT54g ?
15:07.48Othello...
15:08.01Othelloit's more "fun" to run it with a console driver on a ES1371
15:08.11ManxPower"Does this software make my end look big?"
15:08.18mmlj4heh
15:08.19Othello>.<;
15:09.13robl^ManxPower: BIG is relative :)
15:09.26*** join/#asterisk zoa (zoa@82.103.76.147)
15:09.27zoayo
15:10.20blitzragezoa: y0!
15:10.40Daminzoa: What up?
15:10.54file[laptop]eek
15:10.54blitzrageha!
15:11.42DaminYou know.. as hacked up and ugly as chan_sip is, it has to be the most compatible SIP stack out there. ;)
15:11.56file[laptop]Damin: ...nah
15:12.18Daminfile[laptop]: According to the guy from SIPFoundry it is..
15:12.54blitzragerobl^: lol
15:13.04file[laptop]if chan_sip is the most compatible SIP stack... something is really really wrong
15:13.27robl^its the most compatible, just nothing else is compatible with chan_sip
15:13.50Daminfile[laptop]: As far as communicating with all the other broken shot out there? How many hacks have gone into chan_sip to make it work with other fucked up stacks?
15:14.08file[laptop]SIP is so, fun
15:14.34zoaAHA!
15:14.57Daminfile[laptop]: I know people that have gateways that won't communicate with each other, so they bridge them through Asterisk.
15:15.06file[laptop]Damin: but yes, I agree with that
15:16.13Daminthink for a moment, what would happen if we ripped out the current chan_sip and slapped some other stack in..
15:16.44file[laptop]some are too strict, some are too lenient... *sigh*
15:16.58DaminWithin a couple of weeks, the new stack would barely resemble what was originally inserted as it would get hacked up the same way that chan_sip is today just to mainain the compatability that Asterisk has today.
15:17.11ManxPower~docs
15:17.12jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org
15:17.13ManxPower~mailinglist
15:17.14jbotit has been said that mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
15:17.42DaminAssuming that is true, aren't we better off sticking with chan_sip and just cleaning it up?
15:18.12file[laptop]cleaning it up... hehe
15:18.14robl^okie. I am off to find some hard drives.  never have enufff
15:19.08Daminrobl^: Look under the couch cushions.. I once found a 2 gig laptop drive there.. :)
15:19.39robl^Damin: hahah!  are you sure you're not my roommate?
15:20.10Mother_I really like when it's said that a stack is strict or lenient, if there's a protocol/document/RFC/whatever, *that* is what should be implemented
15:20.29Mother_then come Cisco, Nortel, etc. etc. and make their own little 'tweaks'
15:20.46file[laptop]oh, it's not that
15:20.48ManxPowerThere IS chan_sip2.c ya know.
15:20.57file[laptop]it's because the RFC contradicts itself and stuff... it's hard to implement strict to the RFC
15:21.22robl^Damin:  I found a Sun Sparchstation IPX, and a mouse for an Atari TT030 under the couch before.
15:21.30file[laptop]phasing out while reading the RFC is recommended so as to not damage your brain
15:21.31Mother_file[laptop]: agreed, but still, there are some things that are blatant
15:21.37Mother_LOL agreed too
15:22.24DaminMother_: Yeah, but blatant to you, is totally different than blatant to a Cisco engineer. ;)
15:22.33Mother_would there be interest in a SIP-to-IAX protocol converter? I'm cooking up something that should be a small box that does protocol translation
15:22.35DaminMother_: Sip is a pile of shit protocol..
15:22.51DaminMother_: Well.. that's called "Asterisk".. ;)
15:23.05Mother_well yeah, but I would do it the size of half a pack of ciggies
15:23.32*** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com)
15:23.34jontowso a little smaller than an IAXy to do the job of an IAXy but instead plugging in a SIP phone?
15:23.40jontowyou're scaring me :/
15:23.41Mother_something you could plug a SIP phone into, and get IAX or IAX2 out
15:23.54DaminMother_: Hehehe... you are funny..
15:23.56Mother_jontow: yes, along those lines
15:24.02DaminMother_: You funny man..
15:24.08robl^Mother_: I already did that.. with a router that runs embedded Linux.  Just added a slimmed down copy of Asterisk and a few config files.. does nothing more than convert IAX2 <-> SIP
15:24.11Mother_Damin: why? I'm not wanting to do anything to the RTP
15:24.18Blackvelhow can I do this with AGI: "EXEC DIAL ZAP/g1/90,30,tr\n"?
15:24.27Mother_robl^: nice
15:25.01Mother_my idea is to simply convert the "signalling" so-to-speak, and leave the RTP format alone, whatever codec etc. you may want to use would be irrelevant
15:25.50Mother_robl^: what router would that be?
15:26.21DaminMother_: WRTG54GS
15:26.31robl^ahh.  mine is basically to help overcome the SIP/NAT nasties  :)  It allows someone to use a SIP device inside of NAT and IAX2 out to another NATed Asterisk server
15:26.45robl^Mother_: Linksys WRT54
15:26.58Mother_robl^, Damin: thanks, will look into it
15:27.12DaminMother_: You obviously don't know Brian Capouche..
15:27.33Mother_who would that be?
15:28.05Mother_I get ZGR so it cannot be that well known :)
15:28.30*** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl)
15:28.55Mother_BTW ZRG == Zero Google Results just FYI
15:29.10MuppetMasterHello.
15:29.13DaminMother_: http://www.voip-info.org/wiki-Asterisk+Linksys+WRT54G
15:29.19MuppetMasterAnyone know how to call an xmlrpc from a dialplan?
15:31.07Mother_Damin: you appended an 'e' that's why I didn't get anything...I remember playing with shells on a WiFi AP some time ago
15:31.09Mother_thanks
15:31.21*** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it)
15:31.29Aze`re
15:38.44*** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net)
15:39.22*** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com)
15:39.37dmccollumAnyone here that understands Festival/mbrola well enough to help point me in the right direction?
15:40.58ManxPowerdmccollum: I have you links to set it up last night.
15:41.04ManxPowerIt's not easy.
15:41.09*** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl)
15:43.29*** join/#asterisk jeffik (~jeffik@69.158.33.127)
15:44.28*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmoh.dialup.mindspring.com)
15:44.54*** part/#asterisk logicalonline (~logicalon@border.logicalonline.com)
15:45.35Blackvelhow can I do this with AGI: "EXEC DIAL ZAP/g1/90,30,tr\n"?
15:47.57dmccollumManxPower: I looked at all the sites you provided. I installed the mbrola files and made the changes in the config files, but festival complains it needs .scm files that are not part of the mbrola files. I've compared them to the kal_diphone directories and they are no where the same.
15:50.07*** join/#asterisk loick (~loick@APuteaux-151-1-22-92.w82-124.abo.wanadoo.fr)
15:51.28*** join/#asterisk Dabba (~d@ipv6.mfnx.ip6net.net)
15:52.14*** join/#asterisk Jovu (~bert@ev6.net)
15:53.22Dabbaif i do SetVar: var2=1234 in a sample.call file can i use it like Dial(IAX2/8444444@voiptelco/$var2)
16:00.08Jovuasterisk appears to shut itself down immediately, one minute it loads chan_iax2.so the next its "Beginning asterisk shutdown" for no aparrent reason.. any ideas?
16:00.58cjkhow cpu intensive is sip to iax and iax to sip translation
16:01.10Mother_grrrrr
16:01.43*** join/#asterisk coppice (~chatzilla@186.168.17.210.dyn.pacific.net.hk)
16:01.45Mother_this may sound silly, but how do I turn off this yellowish highlighting in vim? I pressed some wrong key combination, and it now highlights every # in yellow
16:01.54Mother_google is not being helpful today
16:02.16Mother_and the docs are too damn long for something this simple, I tried :syn off to no avail
16:04.09CoolAcidYahoo?
16:04.56*** part/#asterisk Sebbbb (~sebastian@oven.f0o.de)
16:05.11Jovuis there any way to find out what's triggering the shutdown? theres nothing i can see in the logs..
16:08.24Mother_nm found it
16:08.38Mother_erased the higlighting part out of vimrc
16:13.55*** join/#asterisk Maxxed (Maxxed@pppte03-324.ght.iadfw.net)
16:14.06*** part/#asterisk point (1000@213.27.44.55)
16:14.44Maxxedshould i be worried about a few warnings on asterisks startup ?
16:15.03MaxxedMar 26 04:18:19 WARNING[1533]: chan_mgcp.c:4044 reload_config: Unable to get our IP address, MGCP disabled
16:15.13MaxxedMar 26 04:18:19 WARNING[1533]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled
16:15.20MaxxedMar 26 04:18:19 WARNING[1533]: chan_oss.c:434 soundcard_init: Unable to open /dev/dsp: No such device or address
16:15.26MaxxedMar 26 04:18:19 WARNING[1533]: chan_zap.c:9615 setup_zap: Ignoring switchtype
16:17.09Maxxedthe last one kinda looks important
16:17.21Maxxedoh the machine is just dhcp for testing and what not
16:17.30*** join/#asterisk loud (~ariel@201.139.192.101)
16:17.37Maxxedit has an ip, so those top few, um, *shurgs*
16:17.58file[laptop]MGCP is cute
16:18.12Maxxedyeah i can give a rats about mgcp
16:18.16Maxxedim a skinny guy :p
16:18.27Maxxedbut none the less, whats with those?
16:18.43file[laptop]it couldn't get the IP, so they're disabled... duh
16:18.53Maxxedi dont have a soundcard in the machine, why is it bitchin at me about /dev/dsp
16:18.57file[laptop]if you wanna use them, set the ip in their config file
16:19.06file[laptop]because it tried to load a console driver to use a soundcard, and it couldn't
16:19.09Maxxedfile[laptop]: i figured the disabled part ;)
16:19.22Maxxedfile[laptop]: ah, the config, got cha :)
16:19.43Maxxedfile[laptop]: so the soundcard thing, thats not a must is it?
16:19.55file[laptop]nope...
16:20.07Blackvelhow can i pass EXEC DIAL ZAP/g1/90 inside AGI?
16:20.22Maxxedwarnings n such upset me, red things flashing on my console..
16:20.24BlackvelAGI Script Executing Application: (DIAL) Options: (ZAP/g1/90,30,tr)
16:20.24Blackvel<PROTECTED>
16:20.25file[laptop]it doesn't matter... you don't need the driver
16:20.25Maxxedheh
16:20.32Blackvelthat is probably wrong
16:20.34file[laptop]Blackvel: | instead of ,
16:20.57BlackvelCould not find application (DIAL,ZAP/g1/90,30,tr)
16:21.00Blackveloh
16:21.06Blackvellike EXEC DIAL
16:21.08Blackveland then?
16:21.23BlackvelEXEC DIAL|ZAP?
16:21.29Blackvelor EXEC DIAL ZAP?
16:21.32file[laptop]same as you were doing, except for arguments after the Zap use |
16:21.43Blackveldo I have to pass ZAP/g1/90,30,tr as the AGI option field?
16:21.54file[laptop]pass ZAP/g1/90|30|tr
16:24.25BlackvelEXEC DIAL ZAP/g1/90|30|tr\n
16:24.27Blackvelhas to be
16:24.32Blackvelthnx file :)
16:24.42file[laptop]uh huh
16:24.43Blackvelmy first xlite->zap call initiated from java
16:24.44Blackvelhehe
16:24.52file[laptop]java, that makes me sad
16:26.25*** join/#asterisk zwhitley (~zwhitley@69-162-31-243.stcgpa.adelphia.net)
16:28.09Maxxedhey file[laptop], u coverd the ip issues, sure enuff i overlooked it in the configs, but what about the....Mar 26 04:18:19 WARNING[1533]: chan_zap.c:9615 setup_zap: Ignoring switchtype
16:28.26*** join/#asterisk bah (048830696@AC92FC1B.ipt.aol.com)
16:28.30*** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com)
16:28.56newlWill an exten using '.' alone be executed regardless if it's been included? i.e. exten => .,1,Setvar(moo=cow)
16:30.05*** join/#asterisk LarsAC (~chatzilla@p508A0A6E.dip0.t-ipconnect.de)
16:30.46Maxxedpssh.. me again jacking off in my configs..
16:30.47Maxxedswitchtype=national
16:31.30Maxxedso! im good, im down to one warning
16:31.40Maxxedand thats the soundcard /dev/dsp one...
16:31.49Maxxedin what config should i look to disable that
16:31.57file[laptop]just don't load the module
16:32.02file[laptop]noload => chan_oss.so in modules.conf
16:32.07Maxxedso in the modules.conf
16:32.10Maxxedah thanks :)
16:32.24Maxxedi am a noob, so thanks for bearing with me :p
16:33.06Maxxed;noload => chan_alsa.so
16:33.06Maxxed;noload => chan_oss.so
16:33.12Zawi'm thinking about ditching livevoip.com since their whole local DID fiasco. anyone have a recommended alternative for IAX?
16:33.42LarsACwhat format is the value of the origtime tag in the .txt voicebox files ?
16:35.00Maxxedsuccess!
16:35.06Maxxedmr file[laptop], i ow u a soda ;)
16:35.42file[laptop]yeh yeah
16:36.04Maxxed:)
16:36.27file[laptop]paypal donations accepted :p
16:36.53file[laptop]haha
16:37.11Maxxed:)
16:37.31file[laptop]joshnet@nbnet.nb.ca everyone - donate a buck - get a truck - run amuck - oh just die :p
16:37.45Maxxedheh
16:38.15file[laptop]if you happened to have a TV tuner card that is Linux compatible though, I'd accept that too
16:39.26MaxxedYou have sent cash! An email has been sent to the recipient.
16:39.29newlany bt8x8 card should do the trick. :)
16:39.37file[laptop]pure silly
16:39.44Mother_notlame rocks
16:39.49file[laptop]yes yes
16:39.55file[laptop]but if anyone has one they'd give me, I'd like that :p
16:40.02Maxxedum, i gota few ati's somewere er unuther, but i dobt there linux compat, hell there bearly windows compatable
16:40.08Maxxedheh im silly like putty
16:40.17file[laptop]all in wonders?
16:40.28Maxxednutty like squirrel turds
16:40.33*** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com)
16:40.46Maxxedum, there like, value tv or some bs
16:40.52file[laptop]creepy
16:40.55Maxxedur best bet is ebay :p
16:41.02file[laptop]yeah yeah
16:41.17file[laptop]Maxxed: thx regardless
16:41.21Maxxedheh
16:41.26Maxxednp :)
16:42.12*** join/#asterisk E|nyPRI_ (les@S0106000bdb97681e.wp.shawcable.net)
16:42.12*** join/#asterisk gopherspidey (~spidey@12-216-167-110.client.mchsi.com)
16:42.20E|nyPRI_Hi
16:42.30gopherspideyHi
16:42.45E|nyPRI_I have a problem with my *.   When I receive a call to it, it runs the context twice.
16:43.09E|nyPRI_http://pastebin.ca/8279
16:43.37E|nyPRI_so i head "Invalid extension" twice.
16:44.19*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
16:44.25gopherspideyIt plays invalid extension twice then hangsup
16:44.26Maxxedum, mabe comment out the ball sack in there ;p
16:44.27Maxxedhar har har..
16:44.44file[laptop]don't... don't... use _.
16:44.55Maxxeduse s
16:45.01file[laptop]or _X. if you HAVE to
16:45.22E|nyPRI_gopher; yes.
16:45.59gopherspideyE|nyPRI_: use the "s" extension
16:46.05E|nyPRI_whoah.  ok, I used _X.
16:46.07E|nyPRI_and it worked
16:46.18E|nyPRI_dunno why that did not occure to me
16:46.31MaxxedE|nyPRI_, file does take paypal donations :p
16:46.40file[laptop]yes, I do
16:46.47E|nyPRI_:)
16:46.50file[laptop]crazy coconuts
16:46.57Maxxed:p
16:47.07*** part/#asterisk gopherspidey (~spidey@12-216-167-110.client.mchsi.com)
16:47.13file[laptop]I can actually get two sodas with that at the corner store
16:47.16*** join/#asterisk gopherspidey (~spidey@12-216-167-110.client.mchsi.com)
16:47.34Maxxedoh well, i have a lil assistance credit :p
16:48.48Mochi E|nyPRI_
16:49.43gopherspideyDoes anyone know if someone is working on the new Asterisk Manager interface?
16:49.47gopherspideyhttp://www.voip-forum.com/news.php?p=168&c=1
16:51.43*** join/#asterisk IQ (~IQ@70-59-165-40.omah.qwest.net)
16:53.33*** join/#asterisk Sedorox (~Sed@Neptune-W.client.wlgrv.pa.sed6.net)
17:02.22*** join/#asterisk lilneon (~tj_r3@cuscon12874.tstt.net.tt)
17:02.30lilneonhey good mornin guys
17:03.14Maxxed"Groundstart signalling is sometimes used by PBX's. If you don't know what it is, don't worry, you won't need it."
17:03.16Maxxedlol
17:04.06*** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net)
17:04.26*** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk)
17:04.26*** mode/#asterisk [+o bkw_] by ChanServ
17:05.29Maxxedwell im off for now
17:05.38Maxxedprob be back in a bit, ima trash my * configs :p
17:05.45Maxxedharass yous guys lata ;)
17:05.49*** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com)
17:05.51Maxxedoh and thanks again file[laptop] :D
17:06.21Jovumy asterisk server begins starting up, and then appears to shut itself down for no aparrent reason.. the logs aren't very helpfull either.. anyone care to take a look?
17:08.43E|nyPRI_moc; hi
17:09.27E|nyPRI_start it from the command line using asterisk -vvvvvvvvvvvvvvvvvvvvvvcg
17:09.34E|nyPRI_see where it bails
17:09.38*** join/#asterisk lilneon_ (~tj_r3@cuscon12874.tstt.net.tt)
17:09.46lilneon_hey everyone mornin again..
17:09.55*** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net)
17:09.56lilneon_anyone here?
17:10.03IQmorning
17:10.10cactus1_awaygood morning
17:10.12*** join/#asterisk CaptChris (~Chris@c-67-181-99-1.client.comcast.net)
17:10.35lilneon_hey guys, if i wanted to connect to my local telco's gsm network.. what would be the best method?
17:10.45lilneon_that is connect my asterisk box to the gsm network
17:10.49Mocby gsm you mean cellular É
17:11.02lilneon_Moc: yeah
17:11.07JovuE|nyPRI_, i tried that... doesnt seem to be very helpfull, can i paste the last few lines in /query ?
17:11.14E|nyPRI_run a DS1 to their access tandem
17:11.40E|nyPRI_jovu; pastebin.ca
17:11.42lilneon_and if they aren't willling?
17:11.46DyOShey does anyone have any recommendations as to what is the best cheapest voip provider to use for international calling unlimited
17:11.52Mocyes, just put a DS1 to them... or if your cheap, you can get those cellphone docking station, but those are crap hehe
17:11.56E|nyPRI_You in canada or usa ?
17:12.25cactus1whats a DS1?
17:12.27Jovuthe output seems to be different every time
17:12.27E|nyPRI_t1
17:12.46lilneon_cactus1:googling it now... gooogle is your friend :D
17:12.48E|nyPRI_cuz if your in Canada/USA they cannoy refuse you equal access.
17:12.57E|nyPRI_dunno how it works otherplaces
17:12.58gopherspidey1.5M or 24 ds0
17:13.01cactus1indeed lilneon
17:13.12MocE|nyPRI_, they can't, but they can charge you alot I guess ;)
17:13.19lilneon_E|nyPRI: well i am not in the US.. live in the caribbean
17:13.28E|nyPRI_Oh.  dunno how it is regulated there
17:13.41lilneon_country has like one or two telco's who abusing their monopoloy big time!!!
17:13.42E|nyPRI_moc; all those rates are tariff'd. you get charged the same as every other carrier.  its not dependant on volume.
17:13.49cactus1carribean thats friggin sweet :)
17:14.00JovuE|nyPRI_, http://pastebin.ca/8281
17:14.12MocE|nyPRI_, I dont even know what exactly connecting to the 'gsm' network do ..
17:14.15CaptChrishello all. i'm looking for some help in setup/config.
17:14.24Mocfree outgoing and incoming call to cell phone for you
17:14.25Moc?
17:14.43CaptChrisi'm using a Mac, and i think i have everything setup proper....
17:14.58E|nyPRI_unknown.  cellular networks are a bit of a mystery to me.
17:15.16E|nyPRI_you'd probably be able to terminate calls cheaper.
17:15.18CaptChriswhile using Jon's Phone Tool trying to connect to vm extension (or any extension) ...
17:15.39MocI wish I found a cheap PRI provider in mtl..
17:15.43E|nyPRI_jovu; hm dunno!
17:15.45CaptChrisI keep getting error message of"channel.c:1817 __ast_request_and_dial: Unable to request channel SIP"
17:15.50E|nyPRI_moc; whats the price for pri there?
17:16.01IQE|nyPRI_,  Moc: Free SMS ?
17:16.03E|nyPRI_jovu; did you just compile it? or is that a binary version
17:16.14Mocwell cheapest I found is 650$ for ye year contract... but I can't get that price rightnow
17:16.22Mocye = 1
17:16.30Mocit about 750 to 1000$
17:17.08JovuE|nyPRI_, its compiled from cvs, i was using 1.0.6 and 1.0.7 before with the same issues.. i've no idea where its failing either, the output is often different depending how far each thread got before it got the shutdown signal
17:17.14CaptChriscan anyone help?  I've tried google for the past week, but no luck
17:18.38E|nyPRI_moc; $750 is reasonable
17:18.47bparkerwho is the best voip provider (with IAX access)?
17:18.50SedoroxCaptChris: have you restarted asterisk?
17:19.02Mocthat still 32$ per channel
17:19.03CaptChrisyes... and reload several times
17:19.19E|nyPRI_moc; yep.   PRI isnt much cheaper than pots
17:19.29cactus1bparker: voicepulse is good if your in the USA
17:19.39E|nyPRI_unless your colocated with the telco.  ie: Colo with GT here in Winnipeg, PRI's are $600 or $490 for incoming only
17:19.41Mother_question: on a large Siemens PBX I worked with some time ago there was a function that allowed you to get a callback when someone that you called was busy became available, by pressing a key combination, is there an equivalent in *?
17:20.00MocE|nyPRI_, incoming only doesnt give you callerid + name ?
17:20.01E|nyPRI_moc; go with a clec tho.  because they can negotiate price.  versus ilec bell who cannot
17:20.09E|nyPRI_moc; sure it does.
17:20.19*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
17:20.22Mocbtw how do you manage with no NI3 support in * ?
17:20.30E|nyPRI_NI3 ?
17:20.43gopherspideyDo any one know if someone is working at "http://www.voip-forum.com/news.php?p=168&c=1"?
17:20.49E|nyPRI_by incoming only, I mean its a normal PRI, but you cant make outbound calls
17:20.56Mocwell from a GT doc, they support Caller Name only with NI3 support (maybe)
17:20.57E|nyPRI_ISP PRI is what they're called sometimes
17:21.06MocE|nyPRI_, the PDF Ive read doesnt have it, hold
17:21.11E|nyPRI_Oh. gt yeah, that propogandah is a bit off.
17:21.11newlNI3, kinda like NA3, only for ISDN. :D
17:21.48E|nyPRI_it is unlikely you will find a telco PRI provider who cannot send you callerid/name in Canada.
17:21.49MocE|nyPRI_: http://www.gt.ca/pdfs/products/Digital_GT_EN.pdf
17:22.04SedoroxCaptChris: I think something is wrong in either youe extentions.conf or sip.conf
17:22.09E|nyPRI_yeah, i've read that exact same pdf.
17:22.14Mocpri isp has no did
17:22.30E|nyPRI_well, it can.
17:22.41Mocdamn GT!!
17:22.42*** join/#asterisk PTG123 (~PTG@ip68-106-24-139.ph.ph.cox.net)
17:22.42E|nyPRI_just have to ask for it
17:22.49PTG123howdy ho
17:23.05*** join/#asterisk lilneon (~tj_r3@cuscon14967.tstt.net.tt)
17:23.14lilneonhi again..
17:23.22E|nyPRI_I've asked them to turn off CallerID screening :)  (ie: 100% spoofable outbound CallerID)
17:23.26lilneondumb internet keeps disconnecting here..
17:23.27Mocok, anyway, I gota find a buisness who need a PRI first..
17:23.35MocE|nyPRI_ good
17:23.46CaptChrisSedorox: I would guess so as well.  i'm very new to Asterisk and have only SIP software currently.  Also haven't been able to establish any connections... but I can tell that I the SIP client is talking to Asterisk (via logs)
17:24.09Sedoroxyou using a soft phone?
17:24.10MocE|nyPRI_, Broadvox doesn't have their caller id blocked
17:24.24E|nyPRI_is broadvox canadian?
17:24.26CaptChrisyes "Jon's Phone Tool" mostly
17:24.33CaptChrissometimes X-Lite
17:24.35Sedoroxnever heard of it... anyway
17:24.38MocE|nyPRI_ they have canadian server with GT
17:24.44MocI had a few cnd DID with them
17:24.59Sedoroxpastebin me your extentions.conf.... the part that your trying to dial
17:25.01Mocbut broadvox su..k
17:25.02E|nyPRI_yeah.  by default callerid screening is on.  you have to issue a NSR to have it removed, and then deal with their legal department.
17:25.19Mocyea
17:25.48Mocwell if Im really hired by this other buisnes, I'll do * deployment in mtl fulltime and developpement also ..
17:26.01CoolAcidAnyone written some interesting dialplans as of yet? Todo something usless or anything?
17:26.08SedoroxMother_: You could do that.. I don't think there is anything in Asterisk now for it... but I would think it would combine macro's.. and maybe agi's
17:26.10CoolAcids/usless/useless
17:26.18CaptChrisPastebin? .. is that a DCC Send?
17:26.24Sedorox~pastebin
17:26.25jbotmethinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
17:26.28Mocso I'll probably have budget for a PRI!!! Can't wait
17:26.38E|nyPRI_:-D
17:26.47E|nyPRI_just wait till you find out that the 800# called is not delivered :P
17:27.03Moc?
17:27.07E|nyPRI_so you'll need a normal DID for every 800
17:27.11*** join/#asterisk lilneon (~tj_r3@cuscon14609.tstt.net.tt)
17:27.14lilneonk back
17:27.21implicitE|nyPRI_, by 100% do you mean any #? i wasn't aware that they allow for intl cid's to be set through them even after you ask them to remove
17:27.22lilneonanyone got my last question?
17:27.45SedoroxCaptChris: I can't do dcc's
17:28.01Sedoroxjust copy and paste the content into pastebin.ca
17:28.12E|nyPRI_implicit; as far as I know yes.   I've talked with a bunch of their switch techs & isdn guru's.   basically, I said I want to be able to set my caller ID to anything.   they said it can be done, just need a non-standard request, and then legal dept.
17:28.15lilneonguys
17:28.18Sedoroxlilneon: doesn't look like it
17:28.46lilneonok.. what are my options if a DS1 not a possibility?
17:28.55lilneonneed to connect to a cellular network..
17:28.58MocE|nyPRI_, you can set the caller ID & Name with GT right ?
17:29.04IQHi, any h.323 softphone?
17:29.09Sedoroxdunno here...
17:29.14implicitE|nyPRI_, usually they might think you are talking about +1 CID
17:29.15E|nyPRI_moc; yes.
17:29.22MocI love canada for that ..
17:29.34implicitMoc, yep
17:29.45CaptChrisSedorox:excuse my ignorance.  It's also been a long time since I've used irc.  I'm not familiar with pastebin. can you clarify?
17:29.47E|nyPRI_my caller name has not worked in a while thouhg, not sure where the problem exists.  It worked in 1.0.5 but 'appears' to of stopped working in 1.0.6+ and CVS head
17:29.52MocE|nyPRI_, you connect * with GT and everything is great ?
17:29.57Sedoroxgoto http://www.pastebin.ca
17:29.58CoolAcidCaptChris: Goto www.patebin.ca
17:30.03implicitnot patebin.ca
17:30.07implicitit is a porn site
17:30.10implicitjust pastebin.ca
17:30.13CaptChrisok
17:30.15CoolAcidarg.. mist the s.. my bad..
17:30.17Sedoroxyea.. ignore CoolAcid :-p
17:30.18E|nyPRI_well, i use GT.  It works, its been reliable.  But I've had a few support issues which resulted in my crying to the CRTC
17:30.30Sedoroxand you can paste large blocks of text
17:30.37Mocgt is bell now so ... :(
17:30.38Sedoroxthen when you submit... give me the URL
17:30.49E|nyPRI_yeah, but only in ownership, the network has not been integrated with them.
17:30.56E|nyPRI_so for all intents and purposes, gt <> bell.
17:31.03E|nyPRI_for now.
17:31.16Mocyea, well I'll continue to dig up a PRI somewhere
17:31.23E|nyPRI_ok, gotta go to the chiro. and let him kick my ass.  ttyl.
17:31.23file[laptop]dig up a PRI... haha
17:31.28file[laptop]I could imagine you literally doing that
17:31.34Mochehe
17:31.37file[laptop]going outside, finding someone's PRI, and moving it to your place
17:31.45IQH.323 SoftPhone - anyone ? help ?
17:31.55MocIQ, good luck is all I could say
17:32.10IQMoc: thanks :P
17:32.17Sedoroxwhats so good about H323?
17:32.26Mocnothing is good about h323
17:32.27file[laptop]absolutely nothin'
17:32.29Sedoroxlol
17:32.29IQGot Avaya at work :(
17:32.35Sedoroxthen why do so many people use it?
17:32.40Sedoroxthats what they use?
17:33.09MocSedorox, is sadly was the first out, in today world, only video conference use h323, and outdated voip software
17:33.12CoolAcidIQ: http://www.openh323.org/ Openphone and OhPhone
17:33.16CaptChrisSedorox: OK. I think I have it http://pastebin.ca/8283
17:33.17IQgot to do some debugging - and dont want to use Avaya's IP SoftPhone
17:33.22SedoroxMoc: gotcha
17:33.32*** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net)
17:33.36Mw3anyone knows vpn capable device with 1or2 fxs port?
17:33.38file[laptop]yeah my school uses H323
17:33.42SedoroxCaptChris: ok.. now.. what extention you trying to dial?
17:33.54CaptChrisSedorox: OK. I think I have it http://pastebin.ca/8283
17:34.03SedoroxCaptChris: yes.. I have it loaded
17:34.06Syncroslinux box + pci fxs
17:34.09Sedoroxwhat extention are you trying to dial?
17:34.09CaptChris8500
17:34.17Sedoroxok
17:34.23IQCoolAcid: was looking more like x-lite :) . openh323.org is a good site tough
17:34.27Sedoroxso trying to dial voicemail.. from a softphone....
17:34.52CoolAcidIQ: Sorry dude. You asked for softphones, it's the only place I know that does H323 softphones :)
17:34.59CaptChrisyes
17:35.12SedoroxCaptChris: ok.... now pastbin the contents of your sip.conf that pertain to the phone in question.. and X out the password(s)
17:35.13IQCoolAcid: yeah, u r right - thanks :)
17:35.37CoolAcidNo worries.. Now if only I could figure out whats wrong with my IAXtel :)
17:35.44CaptChrisok. 2 files will be in pastebin for the complete config
17:36.07Sedoroxummm ok
17:38.09lilneonhey, who told me about the docking stations for cellular phones?... what problem did u have with them?
17:38.26*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com)
17:38.27*** mode/#asterisk [+o anthm] by ChanServ
17:38.50IQlilneon: I think you'll need an ATA with cell phone's dock station
17:39.11SedoroxIQ: actually.. a FXO
17:39.13CaptChrisSedorox: got it.  3 files in sequence in the pastbin  http://pastebin.ca/8284
17:39.29SedoroxCaptChris: is that
17:39.30Sedorox301 #include /etc/asterisk/SIP/SIP-incoming.conf
17:39.31Sedorox302 #include /etc/asterisk/SIP/SIP-100.conf
17:39.31Sedorox303 #include /etc/asterisk/SIP/SIP-200.conf
17:39.32Sedoroxfiles?
17:39.40IQSedorox: yeah - or ATA that has FXO :)
17:39.40CaptChrisyes
17:39.45SedoroxIQ: true :-p
17:39.55lilneonIQ:true.. know anywhere i could find one? apart frm ebay ofcourse
17:39.55blitzragedoes anyone know what iaxcompat actaully does?
17:40.00SedoroxCaptChris: ok.... 1. what phone you dialing from?
17:40.24IQlilneon: they are everywhere... I got SPA-3000, its a great ATA
17:40.33IQlilneon: with 1 FXO and 1 FXS
17:40.45Sedoroxand it isn't that expensive either
17:40.46CaptChris100
17:40.52lilneonk cool
17:41.06SedoroxCaptChris: do you have anything included in sip.conf?
17:42.03CaptChrisonly the 3 files listed at the end
17:42.14Sedoroxin sip.conf? not extentions.conf...
17:42.57CaptChrisyes.  it's the same 3 listed in both
17:43.08Sedoroxwell I can tell you right no
17:43.13SedoroxI've never seen a config like that before
17:43.18Sedoroxeither something's really messed up
17:43.26Sedoroxor I didn't know you could intermix sip and extentions
17:43.52Sedoroxdo yourself a favor
17:43.56CaptChrisah... perhaps thats the trouble.
17:44.04Sedoroxget rid of the includes till your more farmiliar with Asterisk
17:44.05Sedoroxthen...
17:44.13Sedoroxmove the extentions stuff into extentions.conf
17:44.17Sedoroxand the sip stuff into sip.conf
17:44.35CaptChrisi can't remember where i had seen this type of config set
17:44.37Sedoroxand you don't need type=friend in the context either :-p
17:44.40CaptChrisok. i will do that
17:44.47Sedoroxso try that
17:44.54SedoroxI'll be back on later this afternoon...
17:45.03SedoroxI g2g... need to fix a computer.. then lunch/dinner
17:45.35Sedoroxso good luck.. but just seperate that stuff out.. make sure sip stuff is only in sip.conf.. and extentions stuff is only in extentions.conf...
17:45.36Sedoroxenjoy
17:46.08IQCaptChris: Did you try "make samples" after installnig asterisk?
17:47.20CaptChristhanx. i have changed the extensions and sip config files. will reload and try again
17:48.05CaptChrissame trouble
17:48.10lilneonguys, if i am buying a tdm400P card frm digium, going to connect it to a channel bank, or perhaps a straight t1 what config should i take? 4fxs,4fxo's or a mixture..
17:48.39lilneoncouldn't i take 4fxs and have the channel banks contain fxo's and fxs channels?
17:49.09IQCaptChris: its a new installation?
17:49.21Qwelllilneon: gonna get a tdm for a T1?
17:49.23CaptChrisIQ: yes
17:49.31DyOSi have a question about voicepulse...say i get the $45/month unlimited....If two sip devices make calls at the same time using the IAX connection what happens?  Does one of them get a busy tone or do they both go through?
17:49.37QwellYou'd need like...6 of them
17:49.46IQCaptChris: Did you try "make samples" ?
17:49.48ZeeekCaptChris can you repeat in two sentences what the trouble is? I missed that part
17:50.03Zeeekyou want to call 8500 from a soft=phone and... what?
17:50.04QwellDyOS: I believe they both go through, but it charges more for the second call.
17:50.10QwellDyOS: call and ask to make sure
17:50.12IQlilneon: why can't they make it configurable - technical limitation?
17:50.14newlI need to execute an extension block without an extension number. [test] include => moo ; other includes [moo] exten => s,1,Playback(is), the [moo] context never gets ran.  Specifying _. runs the extension twice.  Ideas?
17:50.18CaptChrisIQ:I tried to compile Asterisk myself, but I can't remember if I had too much trouble and simply installed a package from someone else
17:50.21*** join/#asterisk mbranca (~matteo@host-84-222-14-188.cust-adsl.tiscali.it)
17:50.49CaptChrisIQ: but I'm certain I didn't "make samples"
17:50.58IQCaptChris: sorry, I didn't read your old messages.
17:51.34IQCaptChris: doing "make samples" give you some default .conf files - which is a good start even if you are an expert (unless you want to copy your.conf files from another machine)
17:51.56ZeeekCaptChris what happenes on the CLI when you dial?
17:52.06lilneonIQ,Qwell: yeah they can make it configurable..  i just wanted to know if i should take a quad t1 card (T410P) or a TDM400p with (fxs/fxo).. this still kinda confuses me bout fxs and fxo..  :S
17:52.30Qwelllilneon: If you're getting a full T1, the T1 card is best
17:52.33CaptChristhere were alot of .conf files after the install... i'm they did appear to be samples
17:52.41*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
17:53.03CaptChrismessage of "__ast_request_and_dial: Unable to request channel SIP/100
17:53.14ManxPowerlilneon: FXO provides dialtone and ring voltage.  FXS expects to receive dialtone and receive ring voltage
17:53.24lilneonQwell: this config is simply going to be analogphones & POTs->channel bank->asterisk->voip... for now.. but i need to think bout the future.. adn expandability
17:53.39CaptChrisand now getting "No such Host: 100"
17:53.43lilneonthnx manxpower
17:53.52Qwelllilneon: well, 6 TDM cards isn't exactly what I would call "expandable"
17:54.14ZeeekCaptChris here's a suggestion: read the fiirst article below:
17:54.15ZeeekStarter tutorial:
17:54.15Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
17:54.15Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
17:54.15Zeeekhttp://www.automated.it/guidetoasterisk.htm
17:54.15ZeeekTHE reference of the moment:
17:54.17Zeeekhttp://www.asteriskdocs.org
17:54.27Zeeekthen you will see what the lines in those files mean
17:54.37Zeeekit's only a few pages long
17:54.38lilneonqwell:k cool.. so that settles it... quad t1 then.. thnx guys
17:55.29CaptChrisok. I will do that. then if I still have troubles...I'll be back... but probably not until late tonight (PST) or tomorrow
17:55.47ZeeekCaptChris if you read that one article, you'll be miles ahead believe me
17:55.54CaptChrisI think my biggest problem is in not fully understanding SIP and extensions.
17:55.59CaptChristhanks for the help
17:56.07blitzrageAsterisk requires actual reading unfortunately... you can't fake it :)
17:56.19Zeeekexactly: the SIP stuff defines the clients. The extensions.conf defines the dialplan
17:56.24Shido6...
17:56.30Zeeekblitzrage - I can fake it
17:56.31CaptChrisyes!  I seem to have been studying for past 2 weeks now
17:56.41blitzrageZeeek: I fake things plenty :)
17:57.06ZeeekI never got the 70's show
17:57.16CaptChrisnow i need to run.  thanks again all for the help
17:57.16ZeeekToo busy reading about asterisk I guess
17:57.28Zeeekok CaptChris good luck
17:57.44*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
17:57.45Zeeekso... family and tree ?
17:58.07Zeeek<PROTECTED>
17:59.49DyOSI'm looking at the voiceblue product which provides ip to gsm network connectivity...can someone explain to me why someone would want to do this and pay cellular fees?  Why not just use a pstn connection to dial gsm #'s....?
18:00.29IQDyOS: link please
18:00.50Zeeekin this country, it's cheaper to call gsm from gsm than landline
18:00.55Zeeekmaybe that's why?
18:01.19Zeeekhowever, there are voip providers that offer cheaper rates
18:01.19DyOShttp://www.2n.cz/products/gsm_gateways/voip/voiceblue.html
18:02.10DyOSwait you're saying it's cheaper to call gsm from gsm....but i'm looking at this voiceblue device and it looks like it takes sim cards...I'm imaging you could take the sim card outof yoru gsm capable cell phone and throw it in that box
18:02.28DyOSonly problem is...you're going ot incur cellular charges as if you were using your cell phone...i guess i just don't understand
18:02.37DyOSwhy would you want ot do this and not just use viop or pstn
18:02.42DyOSto connect to the gsm network
18:02.49QwellDyOS: Don't forget about "IN" or "sprint to sprint"
18:02.56Shido6dood
18:03.09QwellI just woke up though, and I didn't RTFA
18:03.14IQDyOS: I would do that to send/receive SMS all over the world
18:03.30Shido6setup your own cell net
18:03.48QwellShido6: lets do it
18:03.55Shido6Ive done it in .au
18:04.03IQonly very few carriers in USA can do international SMS
18:04.19QwellShido6: fairly easy?
18:04.21DyOSshido my man my spandsp still isn't working
18:04.23newlShido6: we use the crows for communication here. :)
18:04.26Qwellexpensive though, I'm sure
18:04.50Shido6no
18:04.52Shido6wow
18:05.02Shido6I cant beleive they FINALLY have instructions on their page
18:05.14Shido6that wasnt there when I went through it
18:05.31QwellWhat has instructions?
18:05.33Shido6voice and data on one 1 and out another port to the damn thing, struggled for like 3 days
18:05.43Shido6http://www.2n.cz/news/press_releases/voiceblue_with_asterisk.html?
18:05.46Qwelloh
18:07.56DyOSso i'm not sure i understand yet can someone explain to me the benefit of routing gsm calls when you coudl just use voip or pstn to call cellular?
18:10.11DyOSyou still have to buy the gsm service to get the sim card to stick into the voiceblue device correct?
18:10.15jakepdevI'm trying to make libpri and I get an error on this line "include .depend"
18:11.27Shido6and if you dont have cellular in your area?
18:11.51Shido6if you dont want to use any of the super expensive cellular networks that are in the area?
18:12.03Shido6$1.05 to call anywhere
18:12.04*** join/#asterisk montoya (montoya@200.195.87.156)
18:12.12Shido6per minute
18:12.40jakepdevI think I found out how to make this Avaya connection work
18:12.46jakepdev2b xfer
18:12.52jakepdevAvaya supports it
18:13.03DyOSso the point of voiceblue would be to start your own gsm cellular network and charge people to use it then?
18:13.04gopherspideySome GSM networks in parts of the world are inexpensive
18:13.15jakepdevjust gotta make the newest build of *
18:13.38ZeeekDyOS here's one answer: http://www.2n.cz/news/press_releases/case_study_small_office.html
18:13.52lilneonshido6 : cool idea.. but how does it hook up to the cellular netwrks? u add in sim cards or something .. sorry haven't reaqd thru anything on the site just yet
18:16.33gopherspideyYep, just the sim card that the provider gives you
18:16.56cripito:} there is lot of ppl using this also with stolen sim cards :D
18:17.02lilneonsweet
18:17.08DyOSbut you're still going ot incur normal cellular costs of whatever sim card you stick in that box correct?
18:17.21cripitoyes.
18:17.38lilneonDyos: for me it would wrk great... three cellular companies here.. interconnection between them real high
18:17.50lilneoni could use this to bypass the interconnection feees
18:17.55cripitobrb
18:17.59DyOSso then the only ppl that would benefit from this are people where there are no telephone lines high speed internet etc?
18:18.36lilneonDyos: well no.. i think u are looking at this the wrong way
18:18.42DyOSexplain to me
18:19.11Zeeekactaully the benefits are given on the site - maybe read it?
18:19.20lilneonDyos: well.. i only now checked this out.. but frm what i can see.. i could use it to connect to cellular networks without using a ds1
18:20.11lilneondyos: usually more expensive to connect frm gsm to landlines and vici versa..  getting them to connect directly would be of some use to me.. and i could also allow them to use VOIP for long distance..
18:20.20*** join/#asterisk marks_ (~Marks_Dis@cpe-70-112-81-84.austin.res.rr.com)
18:20.22marks_Hello?
18:20.43marks_Will someone pm me and give me a howto install ASTERISK on GENTOO mini INSTALL.. I WILL PAY!! needs to work with FWD!!
18:21.39IQI think mark got too many PMs
18:21.40*** join/#asterisk marks_ (~Marks_Dis@cpe-70-112-81-84.austin.res.rr.com)
18:21.59IQmarks_: too many PMs ?
18:22.06marks_nope
18:22.12marks_i closed the connection
18:22.12lilneondyos: ideas flowing yet?
18:22.15marks_i didnt gget any!!
18:22.18IQmarks_: I thought ur ISP couldn't handle all PMs u r getting :P
18:23.09DyOSholy shit i think i just put 2 and 2 together and got 5....so say you have 2 companies in diff locations instead of using the internet to route your calls you just get 2 accounts with a normal gsm phone company that allows internetwork calls for free...you basically route all your calls for free over gsm for a very low monthly price..???
18:23.16DyOSam i on track or still confused?
18:24.30marks_nope
18:24.32lilneonDyos: no u getting it.. nice idea though..  so instead of iax between asterisk servers... use this to route your calls between them for free.. and ease up your broadband badnwidth as well for other things.. man u are brilliant!!
18:24.33marks_it can
18:25.36IQlilneon, DyOS, one active call per service provider at a time ?
18:25.45DyOSi would assume
18:26.49IQeven then can same some. Its good for called person too - at times they get free in network calls
18:26.55G0shenDyOS: 3000 minutes limit on most mobile to mobile plans
18:27.03G0shenor somewhere around there
18:27.10DyOSat&t was offering unlimited for awhile
18:27.15IQG0shen: most in USA are offering unlimited
18:27.18G0shenyea "unlimited"
18:27.28DyOSbut you could add unlimited nights and weekends and route calls during a certain time to that provider
18:27.29G0shensame as packet8's "unlimited"
18:27.50G0shenread the fine print, my AT&T contract has a limit on "unlmited" minutes
18:28.01IQyeah - for non gsm as well :)
18:28.20IQG0shen: talk to Cingular - they own AT&T now... Or switch to sprint pcs
18:28.40DyOSso how would it be of benefit to interface with gsm when i could get unlimited plan through voicepulse and call everywhere for free anyways...i don't see why you would want to route cellular
18:28.45file[laptop]we couldn't override their override of our override!
18:28.55marks_CAN ANYONE HELP ME WITH ASTERISK INSTALL AND FWD ON GENTOO? PM ME, I WILL PAY A LITTLE!!!
18:28.59marks_â€
18:29.09G0shenno way, I get 7pm starting nights with my AT&T plan :)
18:29.10file[laptop]marks_: don't do that.
18:29.15DyOShey marks you have a dedicated computer for *?
18:29.17file[laptop]marks_: there's guides out there, Google,
18:29.20file[laptop]just don't do that..
18:29.24marks_ive triied, ive tried
18:29.35_Vileheh
18:29.42_Vileit'd take me 20 seconds to find a url for you
18:29.44_Vileyou haven't
18:30.01marks_i need a step by step
18:30.15IQmarks_: Asterisk@Home
18:30.16blitzrageyou don't get step by steps with Asterisk
18:30.39IQmarks_: just insert the CD and it'll do everything for u
18:30.43marks_yea
18:30.49marks_this is on a VSERVER at the planet
18:31.13IQhttp://asteriskathome.sourceforge.net/
18:32.15IQis there any way we can receive SMS on our DIDs ?
18:32.24file[laptop]there's guides out there... just read the guides
18:32.43IQfile[laptop]: to receive SMS on DID ?
18:33.06file[laptop]IQ: you can't really...
18:33.32*** join/#asterisk dudes (~dudes@12-215-34-6.client.mchsi.com)
18:35.22*** join/#asterisk linagee (~linagee@netblock-66-245-227-138.dslextreme.com)
18:35.31DyOSmarks you have a vserver at theplanet.com
18:35.51marks_indeed...
18:36.04linageeack. i can't get inbound sip calls working. :(   my grandstream phone can do it all correctly. that is, when i punch the right user/pass and such into the grandstream i can receive calls, but i must be doing something wrong with asterisk
18:37.08DyOSwhat os is installed on it
18:37.24linageeDyOS: centOS. this is with asterisk@home. gah.
18:37.25linagee:(
18:38.37linageewhen i call the number with sip debug on, i do see stuff scroll by, but it's not picking it up.
18:39.21Zeeeklinagee the grandstream is registered with the asterisk box?
18:39.30file[laptop]marks_: you should just hire someone to set it up... because if you can't follow the guides out there, you're lost
18:39.33linageeZeeek: yes. now it is.
18:39.36marks_i kno
18:39.39marks_how muc
18:39.39marks_h
18:39.43linageeZeeek: i think i need nat traversal on
18:39.44marks_we are a non-profit
18:39.56marks_and will give them credit on the pbx and an ad on each page of the site
18:39.58Zeeeklinagee so what happens when you dial a number? What does the CLI say (with NO debug on)
18:40.24linageeZeeek: when i dial out? it works perfectly. it has the caller id i put in and everything! :-)
18:40.26Zeeeklinagee if you are traversing NAT yes
18:40.39Zeeekwhere can you not receivecalls FROM ?
18:40.43linageeZeeek: i guess asterisk@home does not have that on by default
18:40.52file[laptop]marks_: find someone who will do it under the terms... there's a list at voip-info.org of consultants
18:41.07linageeZeeek: i can't get inbound calls. my grandstream does not ring, my upstream provider voicemail kicks in.
18:41.25Zeeekwhere is the GS phone, where is asterisk ?
18:41.32Zeeekand who is calling you?
18:41.39linageeZeeek: they are both behind a nat. my cell is calling me
18:41.49bkw_yo yo yo
18:41.51Zeeekso the GS can dial
18:41.54linageeyes
18:41.55_Vilebkw YO
18:41.56Zeeeklike the echo test
18:42.07blitzragebkw_: yo hoe
18:42.09linageeZeeek: and it registers apparently because i get sip debug junk when i ring in.
18:42.17Zeeeklinagee your cell is calling you how?
18:42.18Qwellbkw_: You're just in time.  file was about to take his muffins out of the oven.
18:42.26linageeZeeek: my finger is pushing the numbers. :)
18:42.47*** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com)
18:42.57Zeeekand after the nulbers, presumably there is a network connecting somhow to asterisk? How?
18:43.13linageeZeeek: my voip provider does the pstn to voip
18:43.25linageeZeeek: i am calling the number assigned to me
18:43.48Zeeekso it sounds like you havea problem at the incoming context you had @home set up for the voip provider
18:43.51linageeZeeek: as i said, this is all fairly irrelavent as when my grandstream is setup, it does ring in.
18:44.09file[laptop]my muffins are hot and big
18:44.13Zeeekno it's not irrelevant
18:44.19file[laptop]would anyone like to butter them?
18:44.21linageeZeeek: yes. this is true. my config here is probably foobar
18:44.35Zeeekwhat is irrelevant is trying to understand what's going on if you choose to use a precooked setup
18:45.03Zeeekusually you would know there is an area in the sip.conf where that voip thing is set up
18:45.07linageeZeeek: i know a bit about asterisk
18:45.13Zeeekand you'd go there and see what is wrong with it
18:45.17linageeZeeek: i just like asterisk@home's interface
18:45.23Shido6AMP
18:45.33Zeeeksip show peers shows the voip prov?
18:45.37linageeShido6: and that cool SWF thingy? :)
18:45.39Shido6I should just fix AMP
18:46.29Zeeeklinagee ok, since you know a little about asterisk you can post the relevant sections of the conf stuff for us
18:46.33ariel_Shido6, there is an #amportal on here that can help to fix what you need in amp. There working on the new release right now.
18:46.38Zeeekand the CLI messages
18:46.46linageeZeeek: Zeeek ok, it shows this
18:46.48linageevoipprovider_out/username  voip.ip.addr                 255.255.255.255  5060     Unmonitored
18:46.48linagee200/200          192.168.2.108    D          255.255.255.255  5060     Unmonitored
18:47.04linagee!!! now you know my internal ip. shoot. lol
18:47.13Zeeekwho is 200?
18:47.16Zeeekyour GS?
18:47.17linageemy GS
18:47.29Zeeekso your peer entry for voip provider isn't set up
18:47.35Aze`Anyone have experience with HFC-S PCI Billion ISDN Card ?
18:47.50Zeeekdo you register with that provider? - I guess so if the call comes in
18:47.51linageeZeeek: should i have voipprovider_in?
18:48.16Zeeekhow does the voip provider find you? With 192.168..... ? :)
18:48.21linageeZeeek: well i don't know how far it comes in. i know that when i call i get voip junk
18:48.37linageeZeeek: i have my NAT routing port 5060 TCP/UDP to the asterisk@home box
18:48.41linageeneeding more ports?
18:48.47Zeeekmaybe reading the junk would say something like "ALERT: Intruder trying to break in!"
18:48.54linageelol
18:49.11Zeeekhow do you expect the unknow provider to reach you?
18:49.28Zeeekwhat is there in sip.conf (it is SIP?) to identify it
18:49.36linageesecond
18:49.43Zeeekwho is this mysterious provider?
18:49.46ZeeekFWD?
18:49.54linageeit. hehege
18:50.12linageeZeeek: a friend of mine. :)
18:50.26linageeZeeek: he has too much money laying around or something so he has his own PRI.
18:50.28linageelol
18:50.39Zeeekwell, whatever *it* is, it needs to be identified by asterisk
18:50.48ZeeekStarter tutorial:
18:50.48Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
18:50.48Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
18:50.48Zeeekhttp://www.automated.it/guidetoasterisk.htm
18:50.48ZeeekTHE reference of the moment:
18:50.48Zeeekhttp://www.asteriskdocs.org
18:51.29linageeok. it puts everything into sip_additional.conf
18:53.25linageeok
18:53.27Zeeeklinagee I tried to help you. You answer questions with jokes. Do some reading and come back when you can express the problem in terms someone can understand
18:53.33linageei have a cooked config that it generated
18:53.49linageeaha!
18:53.52linageenat=never? wtf???
18:54.19Shido6.
18:54.24Zeeek..
18:54.30Qwell,.
18:54.31Zeeek....
18:54.34Shido6!
18:54.36Zeeek>>>>>>>>>>>>>>>>>>
18:54.52*** join/#asterisk marlowe (~marlowe@bmw.princetonhost.com)
18:56.07Aze`Anyone have experience with HFC-S PCI Billion ISDN Card, pls?
19:01.50Blackvelwhat is the problem? i have a hfc (different). should not really matter with zaphfc
19:02.54*** join/#asterisk IQ (~iq@70-59-165-40.omah.qwest.net)
19:04.12Aze`Blackvel > in PTP mode, where try to dial Dial(ZAP/g1/xxxxxxx), it's open channel and i listen tone but it dont compose number.. why ?
19:04.21*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
19:04.58Aze`Blackvel > same problem with HiSax
19:05.19*** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com)
19:05.26IQanyone using Festival with * ?
19:05.37marks_FCUK
19:05.40marks_FUCK
19:05.54niZonIQ: I tried...
19:05.58niZonand failed
19:06.07marks_FUCK
19:06.10marks_ok
19:06.10drumkillaI use it - but it soudns terrible :)
19:06.13*** part/#asterisk marks_ (~Marks_Dis@cpe-70-112-81-84.austin.res.rr.com)
19:06.15IQniZon: so maybe my turn to try :P
19:06.22IQdrumkilla: what version do u use?
19:06.30niZonyeah
19:07.36drumkillaheck, i don't know ...
19:07.42IQoh :)
19:07.44drumkillabut check out Cepstral
19:07.48drumkillait's only $30
19:07.55drumkillabut sounds a loooot better
19:08.02IQthe current version that is available on their website is for Free and is a BETA version
19:08.06niZonis it easy to get running?
19:08.18*** join/#asterisk fixitjimmy (~aficionar@dsl82-163-227-225.as15444.net)
19:08.24niZon(directed towards drumkilla)
19:08.31drumkillafrom what I hear, yeah
19:08.36drumkillaI haven't paid for it yet, heh
19:08.37Aze`Blackvel, are u there ?
19:08.40Blackvelaze: hisax? zap and hisax? uh?
19:08.42niZonah
19:10.11IQif Cepstral is better than Festival then :((
19:10.53ZeeekI have Cepstral
19:11.16Zeeekit is easy to get it working - and it sounds *decent* but not great
19:11.23IQI use ScanSoft's Speechify - it sounds better than I speak :P
19:11.43ZeeekI have only used it with pre-digested text files already made into wav
19:11.47niZonhm, is there a guide for getting it working with *
19:11.48niZon?
19:11.56niZonoh
19:12.20IQZeeek: speechify or cepstral?
19:12.40ZeeekCepestral - speechify I couldn't find at the time
19:12.46Zeeekit was bought out by someone
19:13.24Zeeekis there a speechify product to be used with asterisk?
19:13.55*** join/#asterisk IQ (~iq@70-59-165-40.omah.qwest.net)
19:14.12*** join/#asterisk sgaiadog (~josh@24-29-142-30.nyc.rr.com)
19:19.28IQScanSoft's online demo: http://www.scansoft.com/speechworks/realspeak/demo/default.asp
19:19.56Zeeekdo they sell a simple product to be useable with *? I never found one but the voice is superior
19:20.12ZeeekI remember having say "dirty" things :)
19:20.37ZeeekFind the Valley Girl and ask her to ask you anything!
19:21.00IQwe use SpeechWorkd's Speechify at work.
19:21.30IQI wrote a DLL to interface with their engine. but nothing so far for *
19:22.08niZondo it with AGI
19:22.18niZonmake it create a wave then use playback perhaps?
19:22.21*** part/#asterisk sgaiadog (~josh@24-29-142-30.nyc.rr.com)
19:22.33niZondepending on how long it takes to generate the wav
19:22.39*** join/#asterisk sgaiadog (~josh@24-29-142-30.nyc.rr.com)
19:22.52ZeeekGo to Jill and type in "Hey, I saw one once that was a lot smaller"
19:22.54IQniZon: this is not bad - we use this method in few apps that can't call dlls
19:23.06niZonhm
19:23.38linageeZeeek: damn it. i hate it when you're right. :)   "SIP/2.0 403 Forbidden"
19:23.52linageeZeeek: that was in all the "junk" that came in on an incoming call.
19:24.05IQloool
19:24.20IQthey have Indian English :O
19:24.22ZeeekIQ the speechify voices sound great but I've never found a linux product to play them
19:25.00IQZeeek: they do support Linux - call them :)
19:25.07Zeeekbut not for $30 ?
19:25.14IQZeeek: no way
19:25.34Zeeekthe "call us" made me think "starting at $500" or worse
19:25.40Zeeekthe quality is excellent
19:25.41IQZeeek: once I asked them for a development kit - the dude said that it will cost me 1600 to get dev kit
19:25.49Zeeekthat's more like it
19:25.59IQZeeek: then they bought it at work- so I use it for free :)
19:26.11Zeeekmaybe we can all pitch in $32 (50 of us)?
19:26.26IQZeeek: Try "English, Indian - Sangeeta" she sounds a lot better
19:26.38Zeeekwell the demo allows you to put together a whiole set of great sounding files for asterisk
19:27.00IQZeeek: yeah - but there is a limit on length of sentense
19:27.01ZeeekMaybe Dell support should lok into that one ;)
19:27.21IQmax 100 characters
19:27.35ZeeekAll my sex phrases were shorter than that
19:27.41IQloool
19:27.42Zeeeklike "blow me"
19:27.52IQBut youc an always generate multiple phrases and then join them
19:27.56IQ*you can
19:28.07Zeeek"Oooooooh ohhhhh yes, yes, yesy YES!"
19:28.16Qwellyesy?
19:28.35Zeeekyeah well, TTS breaks down sometimes on emotional stuff
19:29.38*** join/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no)
19:29.39IQno sample .wav for Festival ?
19:29.55ZeeekNever tried ti
19:29.57Zeeekit
19:30.47b0efis it possible to change speex encoding parameters during a call from the asterisk cli?
19:31.04ZeeekI had Cepestral's Dave read a tex from a food magazine. "Nothing is harder to make than good fried chicken thant's not greasy."
19:31.26ZeeekAfter the first sentence, it really sounded stupid
19:31.32linageesomeone write a plugin to steal TTS off that one AT&T website. that thing makes very realistic .wav files! :)
19:31.50IQlinagee: link please
19:32.08b0eflinagee: not as long as it's not free;)
19:32.19linageeIQ: google. http://www.research.att.com/projects/tts/demo.html
19:32.57linageeb0ef: just make the module say something like, "you understand and agree to all terms <at the above url> press 'y' to proceed" lol. :)
19:33.01Zeeekthat's not as good as the other
19:33.07linageeZeeek: where is the other
19:33.22Zeeek<PROTECTED>
19:33.38IQlinagee: ATT sound good :)
19:33.50linageeZeeek: does the other one generate .wav files? :)
19:33.54IQand no 100 char limit either
19:34.06ZeeekI think they're way, but I don't remember
19:34.09linageeoh. i guess it does.
19:35.14ZeeekThe ATT has better French though
19:35.22IQdistortion in ATT wav
19:36.17*** join/#asterisk KingMorse (~nowhere@201-108-141-208.tranquility.net)
19:36.27KingMorsehi everyone
19:37.44KingMorseanyone around that can help me troubleshoot a CallerID issue?
19:38.05Zeeekgo for it KingMorse
19:39.14KingMorsewell I'm getting some errors when receiving calls
19:39.21KingMorsea series of 3 actually
19:39.33KingMorseI don't know how to use pastebin so I'll just describe them here
19:39.41KingMorseone says "fsk_serie made mylen..."
19:39.48KingMorsethe next is "CallerID feed failed"
19:39.57ZeeekKingMorse what country are you in?
19:40.02KingMorsethen "CallerID returned with error on channel 'Zap/3-1'
19:40.08KingMorseI'm in the United States
19:40.10Nugget"<KingMorse> I don't know how to use pastebin so..."   <-- the point where I lost interest in helping.
19:40.22Zeeektry adding a wait(2) and see if that makes any diff
19:40.27KingMorseI'm sorry Nugget I heard of pastebin and i know it's polite to use but I don't know how to use it
19:40.33KingMorseforgive me for not being an IRC master
19:40.45Nuggetif you're too lazy to learn how to paste then there's no hope for you to ever get asterisk working.
19:40.46Qwell~pastebin
19:40.47jbotwell, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
19:40.50Zeeekjust go there and paste your stuff but first try adding a wait(2)
19:41.03Nuggetbecause asterisk is a lot harder to use than your paste button.
19:41.03KingMorsea wait(2), will try that hold on a sec
19:41.05Zeeekbefore the rest of the extension
19:41.10Zeeekof even 3
19:41.15Zeeekor
19:41.25KingMorsehey Nugget, I have Asterisk running just fine thankyouverymuch
19:41.35Nuggetyou don't have to be an irc master, but you do have to show some initiative that you actually care about solving your problem.
19:41.38KingMorsebut I prefered to get to the point rather than ask about pastebin first
19:41.50Nuggetif you don't care, why should we?
19:42.21KingMorseactually Nugget, I'm putting you on ignore...have a nice day
19:42.33Nuggetgood luck finding people to do you job for you.
19:42.40linageeNugget: if you were an irc master you would not need an irc client. just talk to the servers through telnet. you could do DCC transfers through telnet too. :)
19:42.41QwellNugget: I think he just proved your point for you.
19:42.49Zeeekoh, Nugget... you're in the shit now....
19:42.59Zeeekput the shields up
19:43.15Nuggetlinagee: yeah, but I get tired of typing PONG all the time.  :)
19:43.20linageehehehe
19:43.44KingMorsehm, zeeek looks like the same issue
19:43.56ZeeekI've seen that error myself a few times
19:44.13Zeeekit's a frank message : "wtf? I don't get that cid that was sent at all..."
19:44.31KingMorsewell I've confirmed that CallerID is set up on the line
19:44.38KingMorseit works with a regular phone
19:44.43Zeeekwhat are you looking at? X100P ? clone?
19:44.55KingMorsei'm running a wildcard TDM400
19:45.02KingMorsewith 3 fxo modules
19:45.15*** join/#asterisk J[SS] (~jeremy@chaoscon.user)
19:45.31Zeeekand obviously usecallerid and all that in zapata
19:45.40KingMorsecorrect
19:45.58Zeeekwhat version of asterisk?
19:46.18KingMorsetried several, currently the most recent CVS-HEAD
19:46.34Zeeeksame error in like 1.0.5 or 6?
19:46.39ZeeekSTABLE?
19:46.56KingMorseyes
19:47.49Zeeekand the FXO work fine other than this issue?
19:47.54Zeeekno noise ?
19:48.19KingMorseyes although I have had issues with echo, they have been largely resolved through the use of fxotune and the echocancel config settings
19:48.26*** join/#asterisk Sedorox (~Sed@Neptune-W.client.wlgrv.pa.sed6.net)
19:49.01ZeeekI'm trying to remember when I saw those errors
19:49.15*** join/#asterisk ckruetze (~nospam@i3ED63E95.versanet.de)
19:50.56ZeeekKingMorse look this up: cidsignalling
19:52.16KingMorsehm, you mean on google?
19:52.25CoaxDMan, learning ASL is tough
19:52.27Zeeekwhatever
19:52.47blitzrageAdult Sign Language?
19:52.47*** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com)
19:52.48CoaxDblitzrage: That'd be "American", but yea
19:52.54blitzrageerrrr... american :)
19:52.59ZeeekAdult is more fun though
19:52.59KingMorsehm, what should cidsignalling be for US?
19:53.13CoaxDblitzrage: I'm learning how to fingerspell right now
19:53.21blitzrageCoaxD: yah.. that's hard to get good at :)
19:53.23ZeeekKingMorse bell? I don't knwo, but there are three possibilities I think
19:53.25CoaxDblitzrage: First things first.. you know.  (Gotta be able to spell to make words)
19:53.25blitzrageCoaxD: you drive?
19:53.31CoaxDblitzrage: I do drive
19:53.59blitzrageCoaxD: when driving, good practise is spelling all the street names
19:53.59blitzrageyou'll get good fast.
19:53.59CoaxDblitzrage: Oh????
19:53.59CoaxDblitzrage: THANK YOU!!! :)
19:54.02blitzrageCoaxD: np!
19:54.04CoaxDblitzrage: That is an EXCELLENT suggestion!
19:54.13blitzrageCoaxD: just spell any sign you look at.
19:54.16KingMorsei've done the same thing with morse code
19:54.28CoaxDblitzrage; (Plus, you're a geek, so chances are, our brains work just a wee bit more alike than i to other more non-technologically oriented people)
19:54.38CoaxDKingMorse; You know, perhaps thats why I could never learn morse either?
19:54.38*** join/#asterisk D1ng0 (~dingo@26.37.33.65.cfl.res.rr.com)
19:54.46blitzrageCoaxD: I had to take some courses when I was teaching soccer to a deaf child
19:54.50D1ng0anyone here living in the phillipiness  ???
19:55.01CoaxDblitzrage: Oh man!  Wow!
19:55.02ZeeekKingMorse I don't have an answer but a start is to google the error messages. You'll see a lot of stuff from mailing list, and maybe you'll find the asnwer
19:55.07CoaxDblitzrage: Not too many people would do that, sir
19:55.15KingMorseyeah I've been through that already zeeek
19:55.23KingMorseI *do* care about solving my problem ;)
19:55.27ZeeekI wish I had written down what made error message go away each time :(
19:55.35CoaxDblitzrage: So apparently, aside from learning another language, you also have an immense appreciation for children.
19:55.41blitzrageCoaxD: yah. I coached soccer from when I was 16-22
19:55.47file[laptop]blitzrage is O-L-D
19:55.51blitzrageCoaxD: this is true. But I don't want any :)
19:55.53blitzragefile[laptop]: lol
19:56.00CoaxDblitzrage: Hehe :) I have a 4 year old
19:56.02Zeeekblitz is way younger than me
19:56.15CoaxDexcept zeek
19:56.21CoaxDzeek is what, 60?
19:56.25file[laptop]FEAR ME!
19:56.30ZeeekBe afraid
19:56.46file[laptop]blitzrage is also VERY silly
19:57.11blitzrageCoaxD: I like kids... just not my own.
19:57.43CoaxDblitzrage: there is nothing wrong with that, sir
19:59.23blitzrageCoaxD: who knows, maybe some day. But with the way the world is going, its not going to be appropriate to bring a child into the world.
19:59.35CoaxDblitzrage: I concur entirely.
19:59.45file[laptop]doesn't mean you can't have fun in the mean time
20:00.23blitzragefile[laptop]: never said I wasn't having fun :)
20:00.28file[laptop]haha
20:00.32*** join/#asterisk vidia (dgcddert45@ip68-12-230-54.ok.ok.cox.net)
20:00.50blitzragefile[laptop]: hell.. ask #asterisk-doc about that bed I broke once... I don't think I'll ever hear the end of that one
20:00.51Qwell...
20:01.18blitzrageQwell: its funny because we weren't doing anything "fun"
20:01.28Qwellmmhmm
20:01.30file[laptop]was it leading up to "fun"
20:01.35zoahello there
20:01.36Qwell:p
20:01.45file[laptop]zoa!
20:03.55blitzragefile[laptop]: unfortunately not...
20:04.06*** join/#asterisk flewid (~flewid@24.42.244.169)
20:04.08flewidsup all
20:04.09blitzrageactually... it might have... I forget... but she was great.
20:04.09file[laptop]blitzrage: poor you
20:04.10blitzragelol
20:04.12file[laptop]lol
20:04.16file[laptop]she was "great" eh?
20:04.28zoathats like she was nice
20:04.29flewidquick question - with the voicemail alerts that get emailed, i know i can define seperate contexts within voicemail.conf
20:04.37flewidbut can those seperate contexts also have a seperate message?
20:04.52flewid(reason being we run a few companies off the same system, and they want the emails tailored to them)
20:05.01*** join/#asterisk [hC] (~turnerd@8.10.2.4)
20:05.25*** part/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no)
20:05.53[hC]If i neglect to specifically set CIDName and CallerID, will Dial() use whatever my phone passes along instead?
20:06.11[hC]or whatever is set in sip.conf, i suppose.
20:06.31Jer13261wow * seg faults when i call it from my C7960
20:06.41Jer13261yes
20:07.00flewidJer13261: interesting
20:07.17[hC]I suppose it goes in the priority of phone, sip.conf, then manual SetCIDName?
20:07.35flewid[hC]: whatever the last place it was set i believe
20:07.36Mw3i have a serious problem. we want to deploy voip services on our network (about 1000 users on a housing estate) but the local authorites requires that it has to be unspoofable/sniffable. so i think we have 2 choises, change or switches to manageable and do portsecurity and etc... (expensive, we have only about 12 manageable switch on the core points) or find some vpn capable ata devices (cpe). is there anythink which could do vpn ? or do you have any other ...
20:07.43Mw3... ideal to make it right for the fucking authority ? :)
20:07.50niZonJer13261: potential remote exploit? :\
20:08.16Jer13261yea .....
20:08.17flewidMw3: wait for iax2's encryption to work as it should? or yeah, a vpn
20:08.23flewidi think the sipuras support vpn's don't they?
20:08.23Jer13261?
20:08.40Mw3flewid: i'm going to check
20:08.41file[laptop]iax2 encryption, THAT'S what I was going to muck with
20:08.49file[laptop]...later
20:08.55Nuggetno!  do it now!  :)
20:08.57flewidfile[laptop]: i knew i had a purpose!
20:08.58flewid:)
20:09.24file[laptop]hehe
20:09.24Jer13261how can i debug this err?
20:09.32file[laptop]okay fine people
20:09.34flewidwhat's the error
20:09.40file[laptop]lemme get a test environment setup
20:09.44Jer13261* seg faults
20:09.51Mw3flewid: by the way, whats wrong with iax2 encryption ?
20:09.53Jer13261when called via sip
20:10.02flewidJer13261: what are you doing, is debug on? did it work before? have you upgraded recently?
20:10.19flewidMw3: i couldn't tell you, i haven't had much time to actually play with it yet
20:10.20flewid:(
20:10.31flewidbeen more involved to switching our systems to use realtime
20:10.35flewidwell, our test systems
20:10.39Jer13261Asterisk CVS-HEAD-03/26/05-07:04:19
20:10.48flewidokay
20:10.57flewidso you're using head, have you tried reverting back to a version that worked?
20:11.02Jer13261i thikn this could be a realtime issue since realtime doesnt work via odbc i had to use mysql :(
20:11.03flewidhead isn't guaranteed to be stable in any way, nor even work
20:11.24Jer13261true
20:11.56flewid%#> show version
20:11.56flewidAsterisk CVS-HEAD-03/26/05-03:10:04 built by root@asterisk on a i686 running Linux
20:12.05flewidi'm using that, and it's working with realtime
20:12.14flewidmaybe just try checking out again
20:12.20Jer13261alright
20:13.12Jer13261doing that now :)
20:13.16flewid:)
20:13.30flewidalso, go to logger.conf, and change console to include 'debug'
20:13.38flewidthen asterisk -rx "stop now"
20:13.45flewidthen asterisk -vvvvvvvvvvvvvvgc and check out the debug outpu
20:15.03Jer13261compiling now
20:15.32flewidwe finally got our portage server, now i get to see how installing gentoo on a raid1 system goes remotely
20:15.33flewid:)
20:16.37KingMorsehm, still no luck on the callerid front
20:17.03KingMorsedoes anyone know what the proper settings are for cidstart and cidsignalling in the US?
20:17.14flewidPRI? or ZAP?
20:17.18KingMorsezap
20:17.24KingMorsewildcard tdm400
20:17.29flewidhmm, i'm in canada, and i didn't have to change anything
20:17.29flewidsec
20:17.33*** join/#asterisk ikey (ikey@220.226.52.252)
20:17.51flewidusecallerid=yes
20:17.51flewidhidecallerid=no
20:17.53flewidfrom zapata
20:18.15flewidthen callerid=asreceived
20:18.20Jer13261ok now no crash :)
20:18.28flewidwithin the channel definition for my x100p card
20:18.42flewidand then i think i had to add a 2 second wait to the main menu so it would grab that callerid info
20:18.56flewidJer13261: glad to hear it :)
20:19.24KingMorseunfortunately I'm getting these errors
20:19.25KingMorsehttp://www.mypastebin.com/?code=738791200
20:19.28KingMorseany ideas flew?
20:20.39flewidsec
20:20.55flewidhmm
20:21.14flewidyou have a callerid on that line right? :) (i know, but sometimes people forget the easy stuff)
20:21.23KingMorseyes, I do :)
20:21.26flewidand what version of * you using? HEAD ?
20:21.28Jer13261can someone tell me what USE_POSTGRES_VM_INTERFACE= does?
20:21.32KingMorseyes, HEAD
20:21.42flewidJer13261: i think that's the oldschool method of doing mysql-voicemail
20:21.46flewidi didn't have to change that with realtime
20:21.55flewidi could be wrong tho - it might be for the web interface
20:22.02Jer13261ok i need the filename of the voicemail in a db someplace
20:22.07Jer13261can that be done?
20:22.15flewidyeah use the ast_config table
20:22.19ikeydoes any one know how many calls per second can asterisk handle in realtime
20:22.25flewidand the add config files scripts that bkw made
20:22.27flewidi think it was bkw
20:22.34ikeyon sip/h323/pstn
20:22.38flewidikey: depends on servers i believe
20:22.46flewidand redundancy, and codecs, and etc
20:22.59flewidKingMorse: hmm, they just changed the callerid stuff to have that cdr_custom.so
20:23.03flewidperhaps it's something ot do with that?
20:23.10flewidi put noload cdr_custom.so in my modules.conf
20:23.15flewidcause it wasn't working for me
20:23.17ikeysay dual processor server with intel xeon + 2gb ram on failover?
20:23.25ikeyand G.711 codec
20:23.32*** join/#asterisk anachron (~phr34k@ip70-176-146-245.ph.ph.cox.net)
20:23.34flewidikey: i think on voip-info they have 350 simultaneous calls on a server like that
20:23.41flewidbut it depends on everyone's situation
20:23.57zoadont trust voip info for how many calls you can do
20:24.02flewidlol
20:24.11flewiddon't trust anyone
20:24.14flewidtrust yourself and your testing
20:24.23QwellDon't even trust yourself.
20:24.38flewidwell, hehe, trust yourself, but add leeway
20:24.39ikey:)
20:25.08ikeywill it support atleast 200?
20:25.11flewidKingMorse: did you try adding debugging in?
20:25.19flewidikey: it should on a server like that
20:25.22flewidthat's pretty beefy
20:25.29Jer13261that allows to store the users in a table i want to store messages
20:25.39flewidJer13261: voicemail messages in db?
20:25.45Jer13261yeppp
20:25.52flewidnot gonna happen afaik
20:25.58Jer13261well not the file itself but a record of the file
20:26.01QwellShouldn't be hard to code up
20:26.06flewidoh
20:26.16flewidyeah you should be able to add something in there as to which file was written
20:26.19flewiddon't think it's possible right now tho
20:27.12KingMorseflewid:  what do you mean add debugging in?  at compile time?
20:27.15Jer13261ok well my problem is i cant work out the filename of the voice mail that got left for a user heh
20:27.38flewidKingMorse: in logger.conf, toss 'debug' behind the 'console' line
20:27.43flewidso you can see the debug output as you run it
20:29.40cjkhi, lets say I do a dial sip/username1&sip/username2 . username1 is on the phone (busy) username2 is unavailable.  what will be the value of dialstatus?a
20:31.05KingMorsehm
20:31.16KingMorseok well nothing looks useful
20:31.27KingMorsealso, the error messages in that pastebin aren't happening
20:31.35KingMorsethey come and go
20:31.44KingMorsebut at no time does CID seem to want to work
20:33.20linageeasterisk inside NAT, clients inside NAT is hard?
20:33.32linagee(two different NATs)
20:34.32niZonsounds painful
20:36.38cjkis there a way to disable in voicemailmain the busy and temporary message options. i want to disable them
20:38.59Moceverything is possible
20:39.17drumkillait wouldn't be easy ...
20:39.20drumkillai'm not sure why that would be worth it
20:39.41BuckRogershey has anyone incontered a problem with the grandstream Handytone 486 with ip registration, it will not take a static or dynamic ip scheme
20:39.47flewiddrumkilla: is it possible to have the email voicemail alerts be different for each voicemail context?
20:39.58flewidlike [default] has email1, [companya] has email2
20:39.59flewid?
20:40.05drumkillaooh ... good question
20:40.13drumkillaI don't think so, but that would be a nice feature
20:40.33zoadrumkilla: digium support doesnt reply to my email :(
20:40.48BuckRogersjust call digium support
20:40.52flewid:) yeah we have a client asking for his emails to be different than the ones we're sending out currently
20:40.52BuckRogerstheir great guys
20:40.55flewidi wasn't sure if that was possible or not
20:40.59flewidguess now i know :)
20:41.00flewidthanks man
20:41.08drumkillanp
20:41.21BuckRogershey has anyone incontered a problem with the grandstream Handytone 486 with ip registration, it will not take a static or dynamic ip scheme
20:41.27zoai dont want to call them :)
20:41.33zoai want an email back :(
20:41.39drumkillathey will eventually
20:41.40drumkillait's the weekend
20:41.42BuckRogerswhy not they have helped our technical staff a load
20:41.43flewidyou can't always get what you want! :)
20:41.46flewidand it's easter right now
20:41.48zoai sent it 3 days ago or so
20:41.50zoa:)
20:41.55drumkillahm .. :(
20:42.00BuckRogersdont for get that it is a holiday weekend for some
20:42.05zoaits just some hard questions on te410ps etc :)
20:42.07drumkillai'm sure they'll get back to you early this week
20:42.14BuckRogerssuch as?
20:42.29zoat1e1override
20:42.38zoalspci output for all the digium cards
20:42.41zoasuch things :)
20:43.18BuckRogerszoa: t1 e1 overfide is the jumper setting?
20:43.21zoafirmware numbers for the cards
20:43.23BuckRogersi beleive
20:43.29zoaits an override for the jumper setting
20:43.33zoai think i figured it out
20:43.41zoathat it accepts a digit between 0 and 15
20:43.45zoaand those are bits
20:43.48BuckRogersa software over ride for the hardware
20:43.52zoawhere 0 = t1 and 1 = e1
20:43.56zoajust want a confirmation
20:44.16BuckRogersin the zaptel drivers..?
20:44.21zoanah
20:44.25zoawhen loading the module
20:44.27BuckRogerswhere about
20:44.33zoaits not in there
20:44.45zoaand i think the firmware == the FALC version
20:44.46zoabut unsure
20:45.07BuckRogersyou want to pass in parematers while loading the zap moduel?
20:45.22zoawhen loading the driver module
20:45.36zoai can use it
20:45.39BuckRogersis that apzap?
20:45.40zoajust want confirmation
20:45.50zoathat would be like wct4xxp
20:46.06Blackvelhey
20:46.13BuckRogersliterally the hardware specific drivers
20:46.17zoaim also documenting all errors you might get with asterisk
20:46.19zoaand the solution
20:46.31BlackvelI get always this reply from AGI (I didn't send any specific command like this): 510 Invalid or unknown command
20:46.35Blackvelwhat's wrong?
20:46.36BuckRogerssounds like an intresting post
20:46.56zoait will take some more months
20:47.14BuckRogersi could imagine
20:47.48Mw3another problem. is there any signalling gateway which do ss7 <-> sip ?
20:48.21BuckRogersyeah asterisk
20:49.03Mw3hm, i've read that asterisk-ss7 just in progress
20:49.11BuckRogershey has anyone incontered a problem with the grandstream Handytone 486 with ip registration, it will not take a static or dynamic ip scheme
20:49.17BuckRogersso what is the problem
20:49.30zoaasterisk-ss7 is there
20:49.30BuckRogerswe do it now
20:49.35zoabut the cost is unknown
20:49.53*** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net)
20:50.08BuckRogersss7 singnaling to and from sip
20:50.13BuckRogers..?
20:50.22BuckRogersits the purpose of diguim
20:50.26zoato and from anything
20:50.27BuckRogershardware
20:50.31zoaits not digium stuff
20:50.45BuckRogersyeah but they support it
20:50.48BuckRogersso does cisco
20:50.50zoai dont think they do
20:51.10BuckRogersi dont think i understand what he is saying, could you re phrase it
20:51.34zoai mean that the ss7 for asterisk implementation is not done by digium
20:51.38zoaand not sold by digium
20:51.42zoabut by a 3rd party
20:51.50zoabut you will need a license for asterisk
20:51.54BuckRogerswhom is this 3rd party
20:51.55zoaif you want to buy the ss7
20:52.00zoaa german guy
20:52.04zoamarku something
20:52.35BuckRogersso what happens when you buy a te405p and have a data and a pri t1
20:52.48BuckRogersand want to go from ss7 to sip
20:53.26zoayou need to buy a license from digium
20:53.31zoaand one from marku
20:53.38BuckRogersis it included in the price of the card?
20:54.01BuckRogersim assuming?
20:54.09zoano no no no
20:54.10*** join/#asterisk dudes (~dudes@12-215-34-6.client.mchsi.com)
20:54.18zoait has nothing to do with the card
20:54.46dudesAnyone know why asterisk CLI programs may not work?  hangup,dial,softhangup, ect.
20:54.48BuckRogersreally so how does a company like voicepulse that uses diguim hardware and sip with pri functino they have bought the lincense
20:55.06zoayou dont need ss7 to use a pri card
20:55.12zoapri also exists without ss7
20:55.14BuckRogersahhh i see
20:55.21zoapri cards also do t1 or e1
20:55.36BuckRogerst1 carry ss7 signaling
20:55.59BuckRogersNorth american standard
20:56.21zoano they dont
20:56.33zoathey can carry ss7
20:56.35zoabut they dont need to
20:57.39BuckRogersyeah thier is common channel and out of band also
20:58.05BuckRogersbut ss7 is old school still in use
20:58.45BuckRogersanyhow has any one hooked up a grandstream 486 here
20:59.15Blackvelwho can help me with AGI?
20:59.16Blackvel<message>Sending [192.168.1.2] : EXEC DIAL ZAP/g1/90|30|tr
20:59.22Blackvelwhen I send this
20:59.32Blackvel<message>Got BINARY [192.168.1.2] : 200 result=0
20:59.34BlackvelI get this reply
20:59.44Blackvelwhich indicates that everything is okay (my zap telephone rings)
20:59.56Blackvelbut when I read the input stream, another messages appears
21:00.07Blackvel<message>Got BINARY [192.168.1.2] : 510 Invalid or unknown command
21:00.30Blackvelbut I don't send any other commands after dial, so, why do I receive a 510 invalid?
21:01.46Blackveldo I get another reply when I hangup the telephone?
21:02.02Blackvellike when the dail returns?
21:02.12BlackvelI receive this at the very end after the pickup
21:02.16Blackvelerr hangup
21:02.18Blackvel<message>Got BINARY [192.168.1.2] : 200 result=-1
21:04.46Shido6zZzz
21:05.00Blackvelyes
21:05.01Blackvel:)
21:05.09Blackveldo you feel like this?
21:05.20*** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net)
21:08.18Blackveloh
21:08.24Blackvelcould fix it
21:08.40BlackvelSAY NUMBER 12345 #
21:08.43Blackvelseems not working
21:09.03Blackvelwhat is in asterisk the default audio file? invalid?
21:11.15*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
21:11.57zoathere is no default audio file
21:11.57*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
21:12.01*** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl)
21:12.46Blackvelany audio files which I can display to the caller?
21:12.53Blackvellike there is the "invalid"
21:13.01zoalots of them
21:13.08zoajust look at the sounds directory
21:13.15zoaor the sounds.txt
21:13.33BlackvelI can believe it
21:13.40Blackvelmy java application can control * :)
21:13.44Blackvelcan't believe it
21:15.13lesouvageI have a x100p clown card. When I connect my phone and make a call after a while I hear "something terribly wrong, goodbuy".  The output of the cli is Mar 26 16:08:26 NOTICE[1144]: app_dial.c:749 dial_exec: Unable to create channel of type 'Zap'. I disabled usb in the bios so there are no shared irg's. What should be my next step?
21:15.49zoano support sorry
21:15.59zoago buy one from digium and i will be happy to help you
21:16.02zoaand so will they
21:19.09bjohnsonthat's what you get for buying a clown card
21:19.32bjohnsondid you get about 20 of them in a tiny little box?
21:19.58lesouvagezoa: Digium stops selling this card otherwise I would have bought one in there webshop.
21:20.13zoathey have tdm cards now
21:20.28zoawhich are lot better
21:20.34zoathe x100ps suck
21:20.56Blackvelhm
21:21.00Blackvelwhat could be the reason for this:
21:21.10BlackvelI only receive one AGI message from *
21:21.33Blackvelbut not all variables (multiple lines) in one read-call. I always have to read twice
21:21.46Blackvelagi_network: yes
21:21.49Blackvelthat is the first
21:22.45Blackvelbut the others I don't get in one read, e.g
21:22.45Blackvelagi://192.168.1.100
21:22.45Blackvelagi_channel: SIP/100-2583
21:22.45Blackvelagi_language: de
21:23.05Blackvelwould be better to receive all in the first read
21:26.35lesouvagebjohnson: I just have one, just to test the concept of a mini asterisk box based on an epia m5000 motherboard.
21:27.26*** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com)
21:27.45jakepdevblackvel - read until the buffer is exhausted
21:28.10jakepdevor until a blank line might be safe
21:28.15jakepdevthen post your commands
21:29.21jakepdevIt is acceptable for TCP messages to be split among packets.  You have to do your own parsing
21:29.26KingMorsetrying again - anyone have any ideas why callerid is not working properly?  I get some errors occasionally like those found here:  http://www.mypastebin.com/?code=738791200
21:29.50KingMorseCVS-HEAD, Digium TDM400 w/ 3 FXO modules
21:30.24dantKingMorse, where are you?
21:30.34KingMorseUS
21:30.53dantDoes it work on any of your lines?
21:31.05KingMorsenone of them that I can tell
21:31.25KingMorsebut callerid is active on the line, i checked with a regular phone
21:31.29dantDo you have callerID on the lines?
21:31.30dantahh
21:31.39danterm
21:31.47KingMorsei have usecallerid=yes, cidstart=ring, cidsignalling=bell
21:32.02KingMorseI've tried playing with txgain/rxgain, to no avail
21:32.12KingMorseI've tried turnning callprogress off
21:32.26KingMorseI've also tried with/without echo cancellation
21:33.13*** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu)
21:33.28danthmm
21:34.00KingMorsei've also tried immediate=no and a Wait(3)
21:34.42KingMorsethe error messages are intermittent
21:34.47KingMorsethey don't always show up,
21:34.51dantmight help if someone else in the US who uses callerID on analogue lines could help
21:34.52KingMorsebut CID never works regardless
21:35.19dantI'm in the UK, CID here is pretty different
21:35.50KingMorseah.
21:36.13dantI'm not sure what to suggest next for your problem I'm afraid
21:36.29KingMorsethat's quite alright, hopefully someone will see this and respond
21:38.20*** join/#asterisk jhiver (~jhiver@AStDenis-103-2-3-232.w81-248.abo.wanadoo.fr)
21:39.00jhiverguys... can somebody send me a test email at jhiver@ykoz.net? I'm trying to figure out wether my mail server works... sorry 4 OT ::
21:39.24*** join/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no)
21:40.33b0efanyone tried using asterisk with jackplug?. I'm not getting it to select the jackplug device. I get snd_pcm_open failed: No such file or directory. I am able to use the jackplug device with other applications.
21:41.15b0efjack is started as the same user as asterisk, btw
21:41.47b0efI just don't get why it's not able to _see_ the jackplug device
21:42.31Blissexb0ef: if it works with an ALSA logical device name, it will work with any...
21:43.36b0efBlissex: not sure what you mean; jackplug is a device defined in asound.conf
21:44.45Blissexb0ef: then if you can use 'jackplug' from any other ALSA app, it must work in Asterisk too. They all use ALSA lib, which interprets logical device names in exactly the same way for all apps.
21:45.08Blissexb0ef: note: try another ALSA app when logged in as the Asterisk login...
21:45.27b0efBlissex: yes, I've done it and it works
21:45.51b0efaplay -D jackplug foo.bar
21:45.56Blissexb0ef: logged in with the Asterisk login?
21:45.56b0efas the asterisk user
21:46.12b0efyes, I'm running it all under the asterisk user
21:46.21Blissexb0ef: then perhaps you have having hallucinations... :-)
21:46.23*** join/#asterisk vlan (~iq@70-59-165-40.omah.qwest.net)
21:46.40b0efasterisk is able to use the emu10k1 device specified in asound.conf aswell
21:46.40cactus1is there anything i can do with asterisk @ home that would be neat before i get the card to connect to the pstn?
21:46.45b0efbut it won't use jackplug
21:46.54*** part/#asterisk vlan (~iq@70-59-165-40.omah.qwest.net)
21:49.19*** join/#asterisk vlan (~iq@70-59-165-40.omah.qwest.net)
21:49.40cactus1b0ef is there anyway i can dial into my asterisk @ home box and call out to a FWD user or something to that effect?
21:50.00b0efsure
21:50.21cactus1would you recommend asterisk @ home?
21:50.34b0efI don't even know what it is;)
21:50.39*** part/#asterisk sezuan (sezuan@port-212-202-202-204.dynamic.qsc.de)
21:50.42cactus1LOL
21:50.46b0ef, but asterisk can do that; it's a pbx
21:51.18b0efasterisk @ home sounds like you got a asterisk box in your home;)
21:51.26b0efs/a asterisk/an asterisk/
21:51.29cactus1heh
21:51.30*** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net)
21:51.31*** part/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net)
21:51.34*** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net)
21:51.46cactus1asterisk @ home is simplified and idiot proof compared to the full asterisk
21:51.53bjohnsonasterisk@home is a CD iso that installs Centos and asterisk
21:51.57b0efahh
21:52.02bjohnsonit is full asterisk
21:52.21cactus1sorry bjohnson i worded it wrong
21:52.22bjohnsonwith the AMP gui .. which attracts idots .. but doesn't prevent them from screwing it up
21:52.38b0efI'm no idiot so I don't need it;)
21:52.56bjohnsonit's a quick way to get a asterisk box running
21:53.12bjohnsonbut unfortunately people try to use the AMP gui that comes with it
21:53.13b0efyeah, but I got my own distro;), so I need to use that
21:53.17vlanHi, how to play .gsm files ?
21:53.26bjohnsonand then come here wanting us to solve their problems
21:53.34b0efvlan: use sox
21:53.35bjohnsonvlan: soxplay is one way
21:53.41vlanbjohnson: thanks
21:54.00bjohnsondon't forget beef
21:54.01bparkeranyone have any experience with Netlogic (VoIP Provider)
21:54.04vidiahi - where do change the email text that is sent along with voicemail messages?
21:54.08bjohnsonerr .. le boef
21:54.12bjohnsoner b0ef
21:54.17b0ef;)
21:54.27b0efit's a recursive acronym
21:54.39b0efb0ef 0wnz eats and fscks
21:54.50b0efand it's sexadecimal;)
21:58.15cactus1theres a bunch of people that are just dead weight sitting on this channel
21:58.35b0efdamn, it would be so nice to get asterisk running through jack
21:58.46PatrickDKdead weight? that's me
21:58.47cactus1?
21:59.14b0efimagine the nice phonecalls;)
21:59.19mgthjack?
21:59.22vidia?
21:59.23file[laptop]I never noticed but Voicepulse Connect upped their price on inbound DIDs...
21:59.31mgthto what?
21:59.32Nuggetinteresting.
21:59.34file[laptop]$11
21:59.37mgthdang
21:59.38cactus1http://cactus.dacnomm.com/calls/bh/product.wav <<caution may be innapropriate for some people
21:59.44Nuggetthey seem to have finally fixed the problems with area code 512 DIDs
22:01.17vidiaanyone know where to change the email message that accompanys voicemail nessages??
22:01.18*** join/#asterisk booyeah23 (~afdas@cpe-24-175-29-253.houston.res.rr.com)
22:01.33file[laptop]vidia: voicemail.conf
22:02.00booyeah23there are a lot of bugs dealing with transfers over bridged connections
22:02.58*** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4089458.sympatico.ca)
22:03.48*** join/#asterisk phsdshft (~phsdshft@66.103.13.10)
22:03.51roamer323~trig
22:05.24linagee~asterisk
22:05.25jbotwell, asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org
22:06.06Chuji~asstricks
22:06.08jboti heard asstricks is #asstricks, the underground gay Asterisk channel. Be afraid, very afraid
22:06.09linageelol
22:06.16linagee<PROTECTED>
22:06.28IQloool
22:06.43linageesecret, underground, XOR-ed, channel. :)
22:07.18b0efmgth: http://jackit.sf.net
22:07.43booyeah23dtmf handling is also buggy over bridged connections
22:09.26Essobichan_sip is coring on rtp.c in -head on me.. Anyone want to comment?
22:09.59linageeis there any way to have a grandstream make those annoying, "you left me off the phone too long and now i'm going to make annoying sounds" sound?
22:11.06b0efI'm trying to get asterisk to work with jackplug. Both jackplug and emu10k1 are defined in /etc/asound.conf, but asterisk is only able to see and use the emu10k1 device. I'm able to aplay as the same user with jackplug. What could possibly be the problem here?.
22:11.13booyeah23linagee: feedback
22:11.29linageebooyeah23: you mean there's actually a way? yuck. :)
22:15.37booyeah23linagee: those phones are really cheap and crappy so its possible to get feedback from the two sides of the handset
22:16.01linageebooyeah23: grandstream is a cheap and crappy phone? :(
22:16.07linageebooyeah23: what would you suggest?
22:16.37Moclinagee, polycom, or it being told the new buisnes grandstream aint that bad, but nothing is confirm on that part
22:16.37linageesomething that at least costs $300?
22:17.05Mocgo with the polycom IP 600 then
22:17.11*** join/#asterisk poli (~poli@200-168-30-125.dsl.telesp.net.br)
22:17.34Shido6?
22:19.50ChujiThe new Grandstream I hear is OK
22:20.20ChujiLots of people came back from VON talking about them
22:20.39file[laptop]I liked it.
22:20.51*** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu)
22:22.36*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
22:27.12tainted-do variables span across contexts?
22:27.13tainted-or no
22:27.31Essobiumm
22:27.37Essobidepend on the variables..
22:28.11EssobiThere's global, and channel as far as I can remember
22:28.27EssobiSo anyone else getting cores on -head?
22:28.46*** join/#asterisk Secretive (~polarisx@c-67-161-5-149.client.comcast.net)
22:28.53SecretiveHey guys, anyone know what the hell this would mean: Mar 26 16:33:07 WARNING[2109]: chan_sip.c:739 retrans_pkt: Maximum retries exceeded on call 8332fb4f-61eedb89-2b337dd2@192.168.1.125 for seqno 106 (Non-critical Request)
22:29.03file[laptop]it couldn't send the packet
22:29.22SecretiveHow would you suggest I fix it?
22:29.35file[laptop]it's probably due to NAT
22:29.46SecretiveShould I do some sort of Port Forwarding to my phone?
22:30.00file[laptop]do you have nat=yes in the the entry in sip.conf for the phone?
22:30.14SecretiveNot sure it's in a database I believe.
22:30.35file[laptop]well if you don't have it set to yes, asterisk is going to send replies to the phone's internal address
22:30.51file[laptop]so I'd suggest turning it on...
22:31.12file[laptop]and depending on the NAT setup, you may need qualify turned on for it to receive calls
22:31.18SecretiveWell -- Some guy setup this system and I can't get a hold of him.
22:31.22file[laptop]and yes, you may have to forward ports 5060 and 10000-20000 UDP
22:31.59bkw_ok I gotta get this off my chest
22:32.03bkw_EVERYONE CAN FUCK OFF...
22:32.23bkw_people are such idiots at times
22:32.31bkw_getting on to me for top posting.. OH FUCK THEM
22:32.31Chujihmmm
22:32.47file[laptop]deep breaths bkw, deep breaths
22:32.48EssobiAND FUCK YOU, AND FUCK YOU, and you're cool, and IM OUT
22:32.54bkw_haha
22:32.56bkw_Essobi, ya that
22:33.17file[laptop]it's okay dear
22:33.23bkw_next time i'll just let cdr_mysql be broken
22:33.23Chujidid you get critch'd ?
22:33.26EssobiD'oh.. Stick.
22:33.51*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || Quick everyone top post to the -dev and -biz list..
22:34.01Jer13261lol
22:34.35bkw_thosee can do h323
22:34.36Chujiamazing how telecom gear goes down in price
22:34.44Essobi:)
22:34.50Chujithat thing was 15,000 a few years ago
22:34.59Jer13261$20???????????
22:35.17EssobiYou can find all kinds of stupid stuff on Ebay like that.
22:35.31file[laptop]they're cheap...
22:36.07Chuji96-56k ports and 4T1 module
22:36.15file[laptop]bkw_: so, should I plan on attending cluecon?
22:36.16bkw_they can do h323 and voice
22:36.18bkw_NEXT!!!
22:36.21bkw_file[laptop], yes
22:36.29file[laptop]but daddy do I have to
22:36.37bkw_yes
22:36.44file[laptop]fine then
22:36.51MikeJ[Jayden]no, thats cheap: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=15635&item=4711941394&rd=1
22:37.04file[laptop]I want my own little apx
22:37.35bkw_why?
22:37.46file[laptop]bkw_: no reason, but it's nifty for smashing people over the head with
22:37.47MikeJ[Jayden]file[laptop], here you go http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=47064&item=7309411810&rd=1  :D
22:38.09Chujibkw_ : Did you use max's when you worked at the ISP?
22:38.48bkw_no
22:38.51bkw_PM3's
22:38.55bkw_I had friends that had a max
22:39.09file[laptop]bkw_: all your pbx are belong to me
22:39.10*** join/#asterisk pimpsmart (~spam@cpe-24-175-29-253.houston.res.rr.com)
22:39.22_Sam--i just threw away a bunch of pm3s
22:39.47_Sam--bkw i didnt realize youre famous
22:39.48_Sam--http://www.sineapps.com/news.php?rssid=378
22:39.58file[laptop]ha
22:40.03file[laptop]_Sam--: I'm on there too ya know
22:40.45pimpsmarthey, anyone have trouble with dtmf when doing iax to iax?
22:40.48file[laptop]but enough about me, who are you and what do you want! how do you know my name...
22:40.50pimpsmarti'm using 1.0.6
22:41.01cjkhow cpu intensive is sip to iax and iax to sip translation
22:41.22*** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com)
22:41.37file[laptop]pimpsmart: do an iax2 debug and see if you're getting DTMF type packets... and if they're going back out
22:42.28_Sam--bkw you're quoted "I would rather work on making the software more stable first.
22:42.36_Sam--"....does that mean you feel the software isnt stable?
22:42.45file[laptop]okay let's not get into that
22:42.47pimpsmartfile[laptop]: will do and report back
22:43.29MikeJ[Jayden]this is what I need: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=94879&item=5762790525&rd=1&ssPageName=WDVW
22:44.02file[laptop]MikeJ[Jayden]: crazy!
22:44.52MikeJ[Jayden]they are fun toys
22:45.11pimpsmartfile[laptop]: no dtmf type packets
22:45.27_Sam--what is the name of that software that is like ipswitchboard but done in flash?
22:45.29file[laptop]then it's not getting the DTMF from the other side...
22:45.37file[laptop]_Sam--: Flash Operator Panel
22:45.41*** join/#asterisk crash3m (crash3m@crash3m.user)
22:45.45_Sam--thankf forgot
22:45.45file[laptop]pimpsmart: SIP -> IAX -> whatever?
22:45.52_Sam--f=s
22:45.58Essobiokay that's screwed up
22:46.10pimpsmarti am doing Cell phone->IAX->IAX->Cell
22:46.11EssobiI started * up and my 7960 starts ringing out of nowhere
22:46.33file[laptop]pimpsmart: well then, someone isn't sending you DTMF I'm afraid...
22:46.35ChujiMy sipura does that
22:46.44pimpsmartiAX is nufone or voicepulse connect
22:46.49pimpsmarti tried both
22:47.16file[laptop]pimpsmart: there should be DTMF packets then...
22:47.29file[laptop]try from a different phone?
22:47.30pimpsmartyeah, that is what i thought
22:47.49pimpsmarttried sip->IAX->IAX->Cell
22:48.07pimpsmartbut cell->IAX->SIP works fine
22:48.15pimpsmartcould it be because of nat issues?
22:48.33file[laptop]I doubt it
22:49.04file[laptop]look on the first both where the call is coming for DTMF packets...
22:49.08file[laptop]er both = box
22:50.21pimpsmartfor an iax2 debug, does it show other information other than the call setup?
22:50.36file[laptop]it'll show tons of info
22:51.32file[laptop]you should see something like this: (only two line paste)
22:51.32file[laptop]Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: DTMF    Subclass: 2
22:51.32file[laptop]<PROTECTED>
22:51.36file[laptop]for transmitting DTMF
22:51.42file[laptop]and an Rx-Frame for receiving
22:55.30pimpsmarti still don't get that
22:55.37pimpsmarti get this though when i send dtmf when it works
22:55.38pimpsmartMar 26 16:55:19 DEBUG[5434]: res_features.c:537 ast_bridge_call: Read from IAX2/NuFone@NuFone2/3 (1,49)
22:56.14zoawhy am i not in sineapps ? :p
22:56.31zoai also look gay on those last pictures
22:56.44_Sam--thats ok zoa i still like ya :)
22:56.47file[laptop]zoa: you're sooooooo sexy
22:56.52zoahaha
22:56.54pimpsmartbut when i did a transfer i got the dtmf from iax
22:56.57pimpsmartRx-Frame Retry[ No] -- OSeqno: 056 ISeqno: 072 Type: DTMF    Subclass: 0
22:56.57pimpsmart<PROTECTED>
22:57.32_Sam--big saturday night eh zoa?
22:57.43zoanah just trying to stay awake
22:57.48file[laptop]pimpsmart: *shrug*
22:57.55file[laptop]hold me hold me HUG ME HUG ME
22:57.58zoatsss
22:58.02zoawatch it sister
22:58.11file[laptop]just don't give me any germs
22:58.18zoahehe lol
22:58.30*** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net)
22:58.36file[laptop]down below or up above :p
22:58.41zoagrr
22:59.32zoacould someone arrange me a picture of an iaxy wrapped in a condom for my next astricon speech about asterisk security ? :)
23:00.09file[laptop]what, can't do it yourself?
23:00.24zoai dont have an iaxy
23:00.35file[laptop]Qwell: lol
23:00.36zoathat makes it kinda hard
23:00.37SecretiveAnyway to have Playback or Background play a GSM file louder ?
23:00.41file[laptop]Qwell: I'm not even gonna ask why
23:00.47Qwellfile[laptop]: good man
23:01.37zoanow get me that picture :p
23:01.50Qwelltrust me, if I had a picture like that, you would've heard about it months ago :p
23:03.13file[laptop]alas, I have not an IAXy
23:03.45Jovuasterisk seems to shut itself down immediately, http://www.firenzee.com/full is the logfile but it doesnt seem to give any clues, i assume one of the sub threads is dying but i dont know how to debug that
23:04.02Jovuanyone have any ideas on this? i've been messing with this all day
23:05.15*** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com)
23:05.33Shido6err
23:05.35Shido6yeah
23:05.40Shido6whats the log say?
23:07.10*** part/#asterisk crash3m (crash3m@crash3m.user)
23:08.39Jovuthe log is on http://www.firenzee.com/full
23:08.45Jovudoesnt seem to say anything usefull
23:09.07Jovuone minute its initializing, the next its shutting down
23:12.43EssobiHmm.. something in -head rtp/chan_sip is out of wack.
23:14.11poliPeople, I need to find out the cadence parameters (prewink, preflash etc.) to get my X100P working with FXS signal from my PBX. Does anyone know some good source of information for that? thanks.
23:14.48Shido6brb
23:17.53JovuEssobi, i've tried with 1.0.6 and 1.0.7 stable versions too, and same problem.. i thought upgrading to -head might fix things
23:19.17Essobithat comment about -head wasn't meant for you
23:19.49Essobithere's something up with re-invite control in chan_sip -head
23:20.52EssobiTry uh... asterisk -cvvvvvggggg and you might see something useful
23:23.27zoabkw, are you coming to astricon europe ?
23:23.34zoafile are you ?
23:24.36file[laptop]nope
23:24.43zoadamn
23:25.18*** join/#asterisk Shido6 (~greg@d57-87-253.home.cgocable.net)
23:25.59file[laptop]Cluecon I can do
23:26.16zoacluecon is so far away
23:26.19file[laptop]Astricon USA, I can do
23:26.23file[laptop]but Europe - HA
23:26.50booyeah23Cluecon?
23:27.10zoai would like to go to astricon eu, us, and von eu
23:27.10NormAstCan't do Europe.
23:27.19zoaand cluecon
23:27.24zoabut dont think i could do all of em
23:27.38file[laptop]if you don't show up to one I go to, I'll be sad :(
23:27.39_Sam--where is astricon us?
23:27.56zoaatlanta again i think
23:28.03_Sam--doesnt say on their page
23:28.13*** join/#asterisk shidan (~shidan@CPE000e08eaf90e-CM014280007905.cpe.net.cable.rogers.com)
23:28.14zoaif you can go, go there, its really interesting
23:28.15file[laptop]that's because it is too far away
23:28.24zoaits les crap than the other things
23:28.27zoaits more hands on
23:28.50zoamost people speaking are not just talking about how great their company is
23:28.55zoabut talking about asterisk
23:28.56SexyKenI need to hire someone to write a GUI and setup an Asterisk server for me.
23:29.13zoasexyken: find me in private :p
23:30.02_Sam--what exactly is dCAP Certification
23:30.07zoaits a training
23:30.15zoadone by the people behind astricon
23:30.20zoacalled olle and steve
23:30.28zoathey give a one week training on asterisk
23:30.37_Sam--what does dpac stand for
23:30.40zoaand then they give you a certificate
23:30.46zoagood question :)
23:30.51file[laptop]dCAP, Digium Certified Asterisk Professional
23:30.59zoaah voila
23:31.13zoafor the moment its just ipsando giving them
23:31.48shidanhey guys
23:32.16zoahello there
23:32.43file[laptop]zoa zoa
23:32.46zoayes yes ?
23:32.53file[laptop]zoa zoa zoa
23:33.02file[laptop]or rather, Joachim!
23:33.07poliIn zapata.conf faxdetect=incoming means Asterisk will detect fax signal coming from an incoming connection or the "incoming" is the context?
23:33.45zoaits on incoming
23:33.52zoanot the context
23:33.56file[laptop]zoa: you are very silly
23:33.57*** join/#asterisk Darwin35 (~Darin@c-24-3-226-147.client.comcast.net)
23:33.59zoacontext would be the defaultcontext stuff
23:34.03zoathank you file :)
23:34.08shidandoes anyone know of a ser live cd thats free by any chance?
23:34.11file[laptop]but a great guy regardless
23:34.26_Sam--damn zoa, is file your biggest fan, or you have others?
23:34.45file[laptop]be careful, zoa can tackle you and win
23:34.48elriahAnyone running debian with the 1.0.5 asterisk package?
23:34.48polizoa, If I can't select the context a fax call is going to, what is faxdetection good for?
23:34.51file[laptop]he can flip you!
23:35.03jeffikShidan: Chk you pvt
23:35.11_Sam--i will not be in any positions where zoa will be flipping and/or tackling me.
23:35.11shidanoh ok ;)
23:35.17_Sam--but thanks for the heads up.
23:35.42zoaneither will file, but he keeps dreaming :p
23:35.49file[laptop]ha
23:42.13IQHi, whats the command syntax to play .gsm using sox?
23:44.00EssobiIQ:  sox is a player too?  I thought it was just amixer. ;)
23:44.32IQEssobi: so thats why its not playing :P ...
23:44.44IQEssobi: what do u use to play - or u convert it into .wav first ?
23:44.49Essobi<PROTECTED>
23:45.02Essobiquicktime will play GSM.
23:45.26Essobibut that's on windows.. no idea for anything else..
23:46.06IQI'm using Fedora
23:46.16EssobiSo anyone else having trouble with -head SIP, reinvite/monitor/ast_channel_bridge?
23:46.24EssobiIQ:  No idea.
23:47.02*** join/#asterisk JerJer[mobile] (proxyuser@adsl-66-143-63-132.dsl.ksc2mo.swbell.net)
23:47.37EssobiI got a repeatable core in -head where ast_rtcp_read is reading a null pointer and coring.
23:48.09shidanjust out of curiosity has anyone here tried compiling ser on windows using cygwin
23:56.03JerJer[mobile]shidan: why in hell would anyone want to do a thing like that?
23:56.17file[laptop]JerJer[mobile]: because Windows is the best server OS!!!
23:56.22file[laptop]the ads told me so so it must be true!!!
23:56.28shidanhaha
23:56.37shidanbecause when im at work i have to run windows
23:56.44shidanthe corporate image
23:56.49shidanworkstation image
23:56.56shidanand i want to work on my ser scripts
23:57.11FuriousGeorgei think im beginning to get this:  true or false:  at the very least i will need a context for incomming and outgoing calls
23:58.23shidanim running the cygwin port for asterisk and it works fine for dev purposes
23:58.37JerJer[mobile]Essobi:  http://bugs.digium.com

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