irclog2html for #asterisk on 20041010

00:00.07bkw_install openssl
00:00.11bkw_thats a simple one
00:00.16devnu112heh
00:00.38bkw_and if you use packages install openssl-dev too
00:00.40tmarshi have it installed but am still getting the error that was the first thing i check
00:01.03docelm0I created a simple agi...  And when it runs it says it ran but it didnt execute it.  Can someone tell me why it didnt do anything?
00:02.25bkw_run ldconfig
00:03.06docelm0who me?
00:03.46syslodtmarsh- if ldconfig doesn't work your libs path is wrong.
00:03.57bkw_correct
00:05.26sudoeris there a way to randomly dial out from a different line like either dial through zaptel or voip or different provider
00:05.47Q-At-Homebbiab, must feed
00:08.14tmarshthe ldconfig looks like it worked thanks
00:08.14bkw_sure
00:08.50Chujidocelm0: what is your agi written in?
00:09.04bkw_ewww agi
00:09.07bkw_how like last year
00:09.23tmarshi spoke too soon same error
00:09.38bkw_path is wrong then
00:09.41bkw_look for libssl*
00:09.48sudoerme?
00:09.50Chujibkw_ : When are you guys going to document res_perl better
00:09.56Chujiintimidating for newbies
00:10.07ChujiThat's why they go to AGI first
00:10.20tmarshhow do i check the paths i did a default install based on asterisk docs?
00:10.24Chujimod_perl has good documentation
00:10.32bkw_Chuji when someone documents it
00:10.35bkw_haha
00:10.38Chuji:P
00:10.39bkw_Chuji give it time
00:11.12rollotomnasitmarsh look in /etc/asterisk/asterisk.conf
00:11.24bkw_what the hell would that have to do with his ldpath
00:12.03rollotomnasibkw_ absolutely nothing. should have scrolled.  going to go crawl away now. :p
00:12.03*** part/#asterisk rollotomnasi (~rollotomn@delmar-209-137-161-171-dsl.cavtel.net)
00:12.07bkw_haha
00:12.12*** join/#asterisk rollotomnasi (~rollotomn@delmar-209-137-161-171-dsl.cavtel.net)
00:12.15bkw_haha
00:12.16justinnnnnnsomeone how do i mysqldump all the tables
00:12.17bkw_ok
00:12.23rollotomnasihehe
00:12.24justinnnnnnlike the hole database ??
00:12.27bkw_justinnnnnn just shut it down and copy it all
00:12.33justinnnnnnhow so ?
00:12.45Chuji/join #mysql
00:12.51*** join/#asterisk iMediax (~user@00d0a8003aa3.click-network.com)
00:12.56bkw_the /var/lib/mysql/DIRNAME
00:13.02justinnnnnnooh col
00:13.02bkw_or something like that
00:13.03justinnnnnnthanks :))
00:13.04bkw_just lock em all and back it up
00:13.13bkw_then unlock the tables or shut it down and backup..
00:15.10*** join/#asterisk Darwin35 (~darwin35@pool-68-162-159-172.pitt.east.verizon.net)
00:15.13*** join/#asterisk z_smurf (~z_smurf@jamlikheten-77-34.ip-pluggen.com)
00:18.50tmarshok i am reay missing something cause the /etc/asterisk/ diretory doesnt exist why would this be i followed the new version install doc?
00:19.13bkw_ignore that
00:19.20bkw_you haven't done a make install
00:19.24bkw_go get openssl and install it by hand
00:19.27bkw_problem solved
00:20.55pfnwhat requires openssl
00:20.58pfnoh, the rsa stuff
00:21.04blacksmallis it possible to have a softphone and * on the same pc?
00:21.28pfnblacksmall it takes a bit of clue, but it's possible
00:21.41blacksmallpfn> want to start me off somewhere?
00:21.47pfnno
00:21.52blacksmallwhy not?
00:22.23bkw_because anyone with a right mind wouldn't do it
00:22.24pfnbecause at this point, it's an exercise left up to you
00:23.36blacksmallwouldn't do it? why? I just want to try to connect to asterisk
00:23.48pfnbecause it isn't a recommended configuration
00:23.58pfnand we aren't going to waste our time trying to encourage it
00:24.04bkw_zactly
00:24.31pfnif you have specific questions/help that you can't answer by reading the wiki, docs, whatever, we can help
00:24.37pfnbut beyond that, we aren't here to handhold
00:24.53bkw_*HINT* run each thing on a diffrent port :P
00:24.56syslodAnyone using * in a large CLEC or large pbx enviroment?
00:27.10bkw_brb
00:29.07*** join/#asterisk Moc (~mochouina@modemcable212.49-80-70.mc.videotron.ca)
00:30.13blacksmallthank you. enough to go on.
00:31.43*** join/#asterisk brc_ (~john@brc.base.supporter.pdpc)
00:33.42tmarshok here is what sequence i ran and i still get the error at the make stage cd /usr/src/asterisk make clean make
00:34.00bkw_go install openssl from src
00:34.03bkw_www.openssl.org
00:34.29tmarshill try that again but have done that twice
00:34.36bkw_installed from src?
00:34.41bkw_cat /etc/ld.so.conf
00:34.50pfnpastebin your make error
00:35.06bkw_add the pat to ld.so.conf to your libs
00:35.10bkw_maybe /usr/local/lib ? then ld-config
00:35.14bkw_er ldconfig
00:35.31tmarshinclude ld.so.conf.d/*.conf
00:35.51bkw_what distro do you run?
00:36.42pfnwhere'd you install openssl?
00:37.19pfn-lssl has nothing to do with ld.so
00:38.00tmarshi ran cat and i get /usr/X11R6/lib /usr/lib/qt-3.3/lib
00:38.24tmarshfedora cat 2
00:38.25pfnwhere'd you install openssl?
00:39.23bkw_pfn yes it would if it can't find it
00:39.37pfnno, ld.so does not determine search path for linking
00:39.38bkw_if it can't find the libssl stuff then you be fucked
00:39.50pfnfor linking, you must specify -Lpath
00:40.00pfne.g. -L/usr/local/openssl/lib -lssl
00:40.07bkw_if you say so
00:40.14pfnof course, runtime is another matter
00:40.31bkw_if it were in ld.so.conf it owuld work
00:40.33pfnif ld.so cannot find it at runtime, then ld.so.conf must be fixed
00:40.42pfnunless binutils changed, it shouldn't....
00:40.45bkw_oh ya
00:40.47pfnbinutils should never look at ld.so.conf
00:40.47bkw_you might be right
00:40.53bkw_its been a long time since I have fucked with that
00:41.46tmarshhere it is  /usr/bin/openssl
00:42.37bkw_no
00:42.38bkw_the libs
00:42.52tmarshfedora 2
00:42.55bkw_locate libssl
00:43.25pfnyou haven't installed openssl-devel
00:43.27pfnyum openssl-devel
00:43.38bkw_er ago
00:43.44pfnsounds like he didn't do it
00:46.49tmarshfedora 2
00:47.03*** join/#asterisk flewid (~flewid@CPE0050ba8c9a95-CM000f9fac6da2.cpe.net.cable.rogers.com)
00:48.25*** join/#asterisk nassy (~mark@24-193-231-136.nyc.rr.com)
00:48.49*** part/#asterisk z_smurf (~z_smurf@jamlikheten-77-34.ip-pluggen.com)
00:49.04bkw_yes yes we get you're using fedora core 2
00:50.15pfnthen gonna head over to the mall and look at crappy mall diamonds
00:50.15pfnheh
00:50.29pfnso I can be thatmuch more impressed with the diamond I ordered that's coming next week
00:51.35bkw_haha
00:51.44bkw_you mean the glass you paid 20000% for
00:54.23tzangerpfn getting engaged?
00:57.52*** join/#asterisk Cheng29 (~cheng29@d57-83-90.home.cgocable.net)
01:00.16pfntzanger already engaged, just haven't gotten the ring, heh
01:00.28pfnwell, not officially, but same difference
01:00.29tzangerpfn congrats :-)
01:00.39pfnthanks
01:00.47pfnbkw nah, mall shit sucks
01:03.10pfngetting a half-carat, ags0, hearts & arrows, e, si1
01:12.22czeroany of you using cisco to terminate pri's then SIP over to asterisk?
01:14.10syslodczero - we tried that but ended up just getting a PRI card for asterisk.  Much simpler.
01:15.01iMediaxbeen awhile.. meetme needs zap for timing correct?
01:15.29syslodYep or at least ztdummy but I haven't seen that work.
01:15.52iMediaxoh ya ztdummy
01:16.39syslodIf you get it to work let me know.  I have a few test systems we tried ztdummy but no luck.
01:17.28C4thYwe use ztdummy
01:17.59syslod:) any pointers on getting it to work/compile?
01:18.11C4thYwe used all 1.0.0 versions
01:18.40C4thYasterisk zapata and zaptel
01:20.13sudoeris it normal for fxs cards to have ports that fit ethernet?
01:20.22*** join/#asterisk hades_ (~hades_@200-180-177-229.paemt7006.dsl.brasiltelecom.net.br)
01:20.30sudoeri got my first fxs tdm card and the ports are ethernet size
01:21.20cypromisyes
01:21.32cypromisat least digium fxs modules are all rj45
01:21.44bkw_you can plug rj11's into that
01:22.05sudoeryes, its digium
01:22.41sudoerif its a 4 port fxf, do i need to plug power into the power hole on the card
01:22.42file[laptop]bkw_: should I buy this workstation?
01:22.48sudoerfxs i mean
01:22.59sudoerfile[laptop] i want to see link too
01:23.12file[laptop]http://cgi.ebay.ca/ws/eBayISAPI.dll?ViewItem&category=11218&item=5723438330&rd=1
01:23.47sudoerlooks nice
01:24.39bkw_file haha I can't copy it
01:25.20file[laptop]it's all like... nifty!
01:25.25bkw_ok
01:25.38file[laptop]of course I'd change it some
01:25.38bkw_I wouldn't buy it
01:25.49pfnxeon 1.0?
01:25.51pfnyuck
01:26.18file[laptop]I don't need a new workstation anyway
01:26.23bkw_I would rather have a Sun Ultra
01:26.27pfnit'd be cool if it had a nice case
01:26.27bkw_10 or something
01:26.29file[laptop]but it's fun to look at everything
01:26.32pfnultra10 is ultra slow
01:26.35bkw_ya that case is ass
01:26.40bkw_pfn slow but solid
01:26.46file[laptop]ultra160 mmm
01:26.47*** part/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net)
01:26.47pfnif I wanted an ultra, have to go pizza box, ultra5
01:27.03bkw_ultra5's are great just don't expect to compile alot on them
01:27.04bkw_haha
01:27.22pfnI've always liked sun pizzaboxes
01:27.26file[laptop]people shouldn't be allowed to visit eBay
01:27.34file[laptop]I find stuff that I want, but that I don't need
01:27.58pfn$100, netbooted off of my linux box and worked ok as a 8bpp display
01:28.16pfncheaper than getting a multisync 21" monitor at the time, heh
01:29.26bkw_not a bad deal
01:29.32bkw_http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=51107&item=5129693351&rd=1
01:30.56pfnthat sounds a bit on the expensive side
01:31.03pfnthey were going for under $250 like after the .com crash
01:31.30bkw_I would love a sunfire
01:31.32bkw_haha
01:31.39tmarshthanks all that did the trick i reinstalled openssl-dev i was surprised that locate shoed it installed but it didnt work again thanks..
01:31.59pfnlocate didn't show it installed, I bet  :p
01:32.02pfnrpm -qa | grep openssl
01:32.07bkw_http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=20328&item=5724909140&rd=1
01:32.07pfnor whatever the short command to that is...
01:32.23pfnrpm -q openssl-devel woulda shown it not
01:32.51bkw_http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=51238&item=5724018528&rd=1  <-- THEY ARE SMOKIN CRACK
01:33.10pfnhah
01:33.24devnu112mmmm, crack
01:33.26devnu112errr, wait
01:33.29devnu112heh
01:33.54bkw_http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=20328&item=5724856760&rd=1
01:34.41ChujiDoes asterisk have problems with variables with spaces in them?
01:34.46file[laptop]why must I have access to Google...
01:35.00czerosyslod when u tired cisco: do you have much cisco experiece?
01:35.08ChujiI'm having a hard time populating setcidname with a variable that has a space
01:35.19Chujiit's truncating after the space
01:35.52Chujiexten => s,2,SetCIDName,${rf_calleridname}
01:36.15ChujiI've tried it in ()'s too
01:36.20Chujiany idea?
01:36.42sudoercan you have cdr do cdr to 2 places at same time?
01:36.48sudoerlike database and to file?
01:37.00pfnyes
01:37.14Chujisudoer : you can pipe the file to a database too
01:37.18Chujithat is what I do
01:37.24syslodczero - We have used cisco since 1997.  "Back in the day" i guess.  We pretty much have eliminated cisco from our archetecture.  We have a class 5 switch, DAX, and on most DS1 level connects we now use Adtran.
01:37.32pfncdr logs to every cdr module that you have loaded
01:37.35*** join/#asterisk cosmic_ (~yes@ns.serversoft2k.com)
01:37.39pfnif you have cdr_csv and cdr_pgsql loaded, it'll log to both
01:38.12*** join/#asterisk hcir (~ar@209-193-42-30-cdsl-rb1.sit.acsalaska.net)
01:38.38czeroSyslod: I'll tkae that as a yes then :) I've beenusing IOS hardcore since 96/97 jsut figured the more I coudl do in IOS the better since while I know linux routers are more my thing
01:38.39pfnok, I think I'm gonna go cruise the mall now
01:38.47Darwin35I can get spark 5 at 20 bucks each at the local computer recycle warehouse here
01:39.15docelm0ok Anyone in here know anything about AGI?   I have a couple questions.. Namely like why mine will not work..
01:39.44syslodczero - What application do you have passing PRI to cisco to SIP to *?
01:40.00*** join/#asterisk Kb1_Kanob (~johnsmith@border.scrd.bc.ca)
01:40.08czeroSyslod I'm not doing it yet
01:40.31czeroI though about useing a t1 win in a 36XX
01:40.36czerothen SIP to *
01:40.59bkw_sun blade 100's aren't that expensive
01:41.33syslodWe tested with SIP to * but once we started our R&D we just bought the PRI card.  We have used Dialogic and Digium.  Both work well and are simple to setup.
01:41.47docelm0Can someone help me?!?!?   Please..
01:42.02*** join/#asterisk Wi_Fi (~OUT@c-66-229-160-14.we.client2.attbi.com)
01:42.16czeroyeah the digum stull is stright forward, I'm thikning about scaling and with IOS I'll get my 729 licences in IOS
01:42.40docelm0When I call the extension that runs the AGI it says it runs but it doesnt do anything its designed for..
01:43.45Chujidocelm0: I tried to help you earlier
01:43.48Chujiyou didn't answer
01:43.56Chujiwhat is the AGI written in?
01:44.35docelm0sorry.. doing 1000 things..
01:44.38docelm0PHP
01:44.50docelm0Im trying to run the sample.agi on the * AGI PHP page..
01:44.51ChujiAre you just wanting it to return a value to the dialplan?
01:45.09docelm0Im just wanting to get it to work..
01:45.56Chujigive me a link to the AGI you are speaking of
01:46.44syslodczero - Are you a carrier or just doing private trunking?
01:47.35bkw_doh
01:48.06czeroI'm going to be sellign serivce but come form a tier 1 carrier background
01:48.23czeroso I always think long term scalability
01:48.33syslodPSTN service?
01:48.53czeroI work for iMCI during hte growth years :)
01:51.09syslodWe are trying 8:1 oversubscription on our * projects with 200 user limit per system.  This is the testing phase and all looks well.  Our exsisting gateways are upwards of 8000 users per slot in a chassis.  We really don't know what the limits are yet.
01:51.56Wi_Fihey guys
01:52.03Wi_Fii might need help
01:52.04syslodHello.
01:52.10*** join/#asterisk brc_ (~john@brc.base.supporter.pdpc)
01:52.38czeroI've got data on over sub rates from big telco's but not sure what point you have to get to before they hold true
01:53.07czerosyslod r u US based I assume?
01:53.32syslodYes US based.  Our data services are something like 1 T1 per 200 subs.
01:54.28czeroyou should be able to scale much higher then that past a couple of t1's
01:55.16czerobut thats the number I use myself for planning :)
01:55.45syslodbbl
01:55.49czerolater
01:56.27*** join/#asterisk wellington (wellington@201008211120.user.veloxzone.com.br)
01:56.48*** join/#asterisk mitcheloc (trilluser@69-169-54-180.anhmca.adelphia.net)
01:57.06file[laptop]all in favor that I upgrade my wifi network to g say aye
01:57.18czeroaye :)
01:57.33file[laptop]I could upgrade my laptop too
01:57.43czeroI did my QAP last month
01:57.50Chujifile: What for?
01:57.51czerogonna start he devices this month
01:58.01file[laptop]Chuji: because! I want a router that runs Linux
01:58.01Chujifile: What's wrong with B?
01:58.11Chujiohh, ic
01:58.26czeroyeah u can hack the G routers
01:58.34file[laptop]what's the cheapest 802.11g router that runs Linux
01:59.26trogsprobably a linksys wrt54g
01:59.46file[laptop]trogs: tried that already...
01:59.48file[laptop]D-Link is cheaper
01:59.51mitchelocpfn are you around?
01:59.53file[laptop]same hardware internally though...
01:59.58trogswhich one?
02:00.13file[laptop]DI-624
02:00.17trogsoh ok
02:00.45file[laptop]hrm it does 108Mbps
02:01.00PatrickDKI have 3 di624
02:01.05PatrickDKand 6 wrt54gs
02:01.06file[laptop]PatrickDK: how are they?
02:01.11PatrickDKI like the wrt54gs better
02:01.22PatrickDKhmm, one di624 is fine
02:01.29PatrickDKother one crashs under heavy use
02:01.35mitchelocanyone know why..after my ip changes, broadvoice bugs out b/c asterisk doesn't re-register fast enough
02:01.36file[laptop]different revisions?
02:01.38PatrickDKother one hasn't been used enough by me to know
02:01.41mitcheloci'm trying to get it to re register
02:01.47PatrickDKfile, all are B I think
02:01.50trogsthe linksys have rp-tnc connectors though, as opposed to the annoying sma's of the dlinks
02:01.56mitchelocand the register => user:pass@server command doesn't do it ?
02:02.01file[laptop]interesting
02:02.03PatrickDKI perfer the sma
02:02.11file[laptop]PatrickDK: does the D-Link run Linux?
02:02.23PatrickDKhmm, probably
02:02.27trogsthe sma is probably ok, if you don't want to make your own cables :)
02:02.30file[laptop]nmap one and see :)
02:02.36PatrickDKmake own cables? for what?
02:02.49trogsantennas
02:03.00PatrickDKI have alot of antennas with rp-sma
02:03.14PatrickDKthey are actually more common than rp-tnc
02:03.26trogsmine are usually N type, and i have a bunch of N -> rp-tnc
02:03.34mitchelocguys? it's a quick question, how do i use register => user@server in sip.conf to get asterisk to re-register every say 60 seconds
02:03.57bkw_go read the sip.conf.sample
02:04.02PatrickDKN is good, though, not at high freq
02:04.09PatrickDKsma is high freq version of N
02:04.12trogs;defaultexpirey=120             ; Default length of incoming/outoing registration
02:04.12mitchelocbkw i did
02:04.16trogsi assume thats what you want
02:04.22mitchelocit says register can use a defined peerid
02:04.24mitchelocdoesn't work for me
02:04.50bkw_defaultexpirey=120             ; Default length of incoming/outoing registration
02:04.55bkw_you are blind then
02:05.05*** part/#asterisk rollotomnasi (~rollotomn@delmar-209-137-161-171-dsl.cavtel.net)
02:05.59mitchelocno bkw, thats not what i'm talkin about
02:06.10mitcheloci think i have the syntax wrong this: register? => <username>@<sip client/peer id in sip.conf>/<extension> :Register with a SIP provider
02:06.11bkw_I think it is
02:06.14mitchelocthats on the wiki
02:06.25mitchelochow do i say register => peercontext
02:06.37mitchelocit won't use it to register remotely
02:06.40bkw_you don't
02:06.52bkw_what is your goal?
02:06.58mitcheloci have it all working now
02:07.03mitchelocbut my house has  a dynamic ip
02:07.05mitchelocand when it changes
02:07.08mitcheloci can't call out BV
02:07.09file[laptop]bkw_: Ever get that NSLU
02:07.10file[laptop]?
02:07.15bkw_file SOON
02:07.20file[laptop]nifty
02:07.20mitchelocb/c it loses my registration/ip relationship
02:07.27bkw_find a better ISP
02:07.27mitchelocso it needs to re-register to them
02:07.32bkw_but 60 seconds/
02:07.35bkw_your ip changes that fast?
02:07.39mitchelocno every 12 hours
02:07.42bkw_DAMN
02:07.50mitcheloci go to sleep, wake up, and i have to reload in asterisk to use my phones
02:07.51bkw_defaultexpirey would do it then
02:07.52mitchelocit sucks
02:08.12mitchelocok, i got that, just don't get what in the context tells asterisk to go out and register to broadvoice?
02:08.18mitchelocor defaultexpirey does that automatically
02:08.29mitchelocoh
02:08.31mitchelocgot it
02:08.32bkw_yep
02:08.33mitchelocit's for general
02:08.33bkw_it would
02:08.40mitcheloci was thinking it goes under a sip context
02:08.50bkw_haha this is why I said look at sip.conf.sample
02:09.08mitchelocyep got ya, just didn't make sense to me *sorry*
02:09.20bkw_asterisk's peer/user matching in chan_sip is leaves alot to be desired
02:10.33mitchelocyep but it works, i kept expecting somehow to make register point to a defined context and use that
02:10.45bkw_no
02:10.53bkw_you do [peer] and host=ip
02:11.01bkw_or [user]
02:11.09bkw_but its better to match on ip to an entry
02:11.23GodseyI'm playing w/ DISA
02:11.25mitcheloci got it
02:11.36bkw_the best way to learn it is ot look at find_user and find_peer I thin kin chan_sip
02:11.39Godseyit seems no matter what I do, after I get second dialtone I can't dial :)
02:11.42mitchelocdisas fun ;)
02:11.47mitchelocin the code?
02:11.48Godseyafter 1 or 2 numbers I get error tone
02:11.57bkw_yes in the code
02:12.03bkw_best place to learn how something works
02:12.03mitchelocheh i don't know C
02:12.10bkw_I didn't either when I started witha sterisk
02:12.14mitcheloci could look though, but i wouldn't understand it
02:12.29bkw_know php/
02:12.30bkw_or per?
02:12.32bkw_er perl?
02:12.42file[laptop]is 802.11g worth it...
02:12.46bkw_file yes
02:12.49mitchelocphp, c#, vb,asp, html, java, those stuff, yea i know it's someone similar
02:12.56bkw_then C isn't that far off
02:12.58bkw_you'll have it down
02:13.04mitchelocbkw, how much would a decent webconfig utility be useful for everyone?
02:13.04file[laptop]it isn't, I know it isn't
02:13.05bkw_my first C project was cdr_odbc
02:13.07mitchelocthat fully configs asterisk?
02:13.35bkw_mitcheloc learn the new realtime stuff
02:13.46bkw_brb
02:13.52mitchelocnew realtime? can you link me somewhere?
02:15.14Godseymight any of you use DISA? :)
02:15.33mitcheloci do
02:15.37mitchelocwell i've set it up
02:15.38mitchelocit's easy though
02:15.43mitchelocnothing to it?
02:15.48Godseyit seems like it should be!
02:15.49Godsey:)
02:16.06GodseyI do 1,Answer()... 2,DISA,1234|trunkld
02:16.20Godseyi dial the #, dial 1234# and get second dialtone
02:16.29mitcheloci don't think you pass disa parameters
02:16.29GodseyI enter any digit and get busy
02:16.31mitchelocjust go DIsa
02:16.31Godsey(single)
02:16.36mitchelocand it gives the user a dialtone
02:16.38mitchelocnothing to it
02:16.44sudoeris the AMA flag in cdr equivalent to account number , so if i had an account number for each entry in sip.conf or iax.conf, i could assign their own ama #?
02:16.58Godseyyou don't give it a password or context?
02:18.00mitchelocwhatever context it runs in i presume is the context it uses
02:18.18bkw_show application DISA
02:18.22mitchelocoh i was using no-password
02:19.27mitchelocbut thats cause i used it differently
02:19.28sudoeri cant find more info abouthow exactl ama works,voip-info.org didnt have enough information, can anyone tell me more about ama?
02:19.40mitchelocbasically i would have someone call in, dial 9 then it would dial them back on a predefined number
02:19.44GodseyI have a DID setup to answer and give a dial tone
02:19.46mitchelocand then send them straight into DISA
02:19.57mitchelocthat way it's more secure then a password
02:20.52GodseyI have [disadial]
02:21.06*** join/#asterisk _mwoodj_ (~MWoodJ@hyper-eye.digium.sponsor.pdpc)
02:21.06Godseyexten => 1,1,Dial(SIP/95551212@as5400,60)
02:21.22Godseyexpecting that when I dial 1, it'll ring the 555# :)
02:21.26GodseyI also have _1,1,
02:22.32mitchelocpost your example fully to pastebin can you?
02:22.44Godseytrying 1 more thing :)
02:22.53Godseyhaving the Anser() bit go to disadial first
02:24.33Godseythat worked
02:24.35Godseyyay :)
02:25.02Godsey[disadial]
02:25.03Godseyexten => s,1,DISA,1212|disadial
02:25.03Godseyexten => _.,1,Dial(SIP/9${EXTEN}@as5400,60,t)
02:26.44mitchelocbkw, are you back/
02:28.23czerohas anyone test the ZyXEL WiFiSIP phone yes?
02:28.25czeroyet even
02:30.41bkw_czero its shit
02:30.44bkw_mitcheloc yes
02:30.51mitchelocbkw: what do you mean by the realtime stuff?
02:31.01bkw_go read the cvs mailing list
02:31.18czerobkw have you tired anythere where any good?
02:31.38mitchelocbkw: http://www.google.com/search?hl=en&q=asterisk+cvs+mailing+list+real+time&spell=1
02:31.39mitcheloc?
02:31.40sudoerhas anyone played with those xml phone books incorporated inside of phones
02:31.43mitcheloc<PROTECTED>
02:31.53sudoerlike in cisco's, they have an xml browser?
02:32.15czeroI've used the Cisco one, at Cisco does that count :)
02:32.20czerobut I've never set one up
02:32.52*** part/#asterisk wellington (wellington@201008211120.user.veloxzone.com.br)
02:33.15bkw_mitcheloc go check out latest cvs
02:33.16bkw_check config.c
02:33.20bkw_you'll have to check it out
02:33.23bkw_I can't really explain it
02:33.30bkw_its real time db stuff for asterisk
02:34.31Godseyguess I have some work to do on echo cancelation :)
02:35.36Godseythe newer cisco as5400 modem code is suposed to have good echo cancelation, I just couldn't figure out how to do it in ios
02:35.47mitchelocheh bkw, sorry i checked it out and i'm looking at config.c but your talking about code? i don't know enough of how the code is to know whats going on
02:35.51mitcheloccan you tell me the gist of it?
02:36.05mitchelocload realtime?
02:36.09mitchelocso no more "reloads?"
02:38.57bkw_correct
02:39.06bkw_look for the word realtime on all the stuff
02:39.07bkw_config.c
02:39.10bkw_chan_sip.c
02:39.12bkw_app_voicemail.c
02:39.14bkw_chan_iax2.c
02:39.22bkw_look at app_realtime.c
02:39.58*** join/#asterisk mshades (~mshades@67.176.22.85)
02:40.22*** join/#asterisk Legend` (~Legend@24.244.142.133)
02:40.26mitchelocok very cool
02:40.36mitchelocso what does that have to do about the web config tool i was talking about?
02:40.50bkw_tiz gonna make it EASY
02:40.51bkw_:P
02:40.58bkw_README.extconfig
02:41.48*** join/#asterisk mitchel (trilluser@69-169-54-180.anhmca.adelphia.net)
02:41.56mitchelwhoops got disconnected
02:42.06mitchelbkw: what did that have to do with my question about a web config tool?
02:42.09bkw_README.extconfig
02:43.35mitchelok
02:43.40mitchelso say i wrote that web configuration
02:43.43mitcheluse databses?
02:43.53bkw_ya
02:44.10bkw_you could pull data from anything
02:44.11bkw_LDAP
02:44.14bkw_DNS
02:44.15bkw_what ever you want
02:44.18bkw_just write the driver for it
02:44.57*** join/#asterisk mitcheloc (trilluser@69-169-54-180.anhmca.adelphia.net)
02:45.02mitchelocdamnit, i keep getting d/c
02:45.04mitcheloci understand that
02:45.16mitchelocso do you think i should write a well indepth config ui?
02:45.17bkw_haha
02:45.25bkw_mitcheloc sure you could
02:45.54mitchelocbut your recommendation is?
02:46.02bkw_give it a few more weeks before you do anything
02:46.35mitchelocok, i'd like to seriously dedicate some time to making an official user interface though if you think thats possible
02:46.43bkw_yes its possible
02:46.46mitcheloci promise it would be good
02:47.21mitcheloci think i'm going to do that
02:47.24bkw_you could build an iax, sip and voicemail stuffs
02:47.25mitchelocthing is i think everyone would hate it
02:47.27mitchelocsince i like c#
02:47.33bkw_ewwwwww
02:47.36bkw_write it in perl or php
02:47.39mitchelocsee
02:47.44mitcheloctheres nothing wrong with c# though
02:47.47mitchelocthats whats stoping me
02:47.54mitchelocit's quick, efficient
02:47.59bkw_how?
02:48.05bkw_M$ came up with it.. haha
02:48.10bkw_I have never seen it
02:48.12mitchelocit's practically 100% compiled code when run on the server
02:48.15mitcheloccompared to php
02:48.23mitchelocoh c# code is easy nice syntax, hold on a sec
02:48.26bkw_I can't comment since I haven't written anything in it
02:49.10mitchelochere...look at some of this code, it's not my best work but it's something to look at
02:49.14mitchelochttp://www.titaniumsoft.net/downloads/tsweblib.0.2.src.zip
02:50.20Chujiwhere's Corydon when I need him
02:50.33debaserok, everyone needs to go watch Team America: World Police
02:50.40debaserits quite possibly the funniest movie i've ever seen
02:50.44Chujigawd that looks stupid debaser
02:51.06debaseryou're a fucking douche
02:51.08ChujiAnyone good with AGI::Asterisk?
02:51.11debaserit was hillarious
02:51.48dan2C# == java
02:51.59sudoeris cisco the only phoen that has a 'xml mini browser'?
02:52.02mitchelocC# ~ java
02:52.09mitchelocit's similar, not the same though
02:52.25dan2mitcheloc: ocaml > C# | Java
02:52.28dan2:-P
02:53.22mitchelocit's perl though? lol
02:54.40mitchelocugh! it's ugly =p
02:54.44dan2ocaml?
02:54.46dan2ocaml is great
02:55.32mitchelocthe reason i wouldn't use it is
02:55.36mitchelocit doesn't look widely supported
02:55.42dan2*COUGH*
02:55.48mitchelocis it?
02:55.50dan2ocaml is ported to many more platforms than C#
02:55.54dan2MANY
02:56.05mitchelocaye well i have no say so as i don't know anything about it
02:56.28mitchelocbkw: did you see the source yet?
02:56.30dan2mitcheloc: ever heard of the file sychronization tool called unison?
02:56.32*** join/#asterisk gregwood (~lynnwood@pcp01073653pcs.andrsn01.tn.comcast.net)
02:56.48dan2mitcheloc: its based after rsync protocol, but more powerful, written in ocaml, on every platform
02:58.56mitchelocnope haven'theard of it
02:59.05mitcheloci'm new to this linux stuff/that area
02:59.11mitcheloci was a windows geek before ;)
03:00.30*** join/#asterisk astk_tester (nathan@68.114.199.229)
03:00.48astk_testerhm, anyone from NuFone here?
03:01.17astk_testerI'm having some trouble registering I believe
03:03.48astk_testeranyone at all here?
03:04.04czeroyes but I'm not from nuphone
03:04.11astk_testerthat's alright
03:04.14gregwoodIm here as well
03:04.45astk_testeranyone know of any good asterisk-compatible VoIP providers with pay-as-you-go and offer toll-free DIDs in the US?
03:05.02*** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc)
03:10.36*** join/#asterisk B0ngFrOg (~wsmith@67.176.78.68)
03:10.40B0ngFrOglo
03:11.22mitchelocastk i guess nufone
03:11.29mitchelocbut i don't ever get good quality from them
03:12.17iMediaxthey've been great for me
03:12.42czerohow many minutes amonth do you guys push?
03:12.46czeroif I may ask
03:12.49mitchelocless then 10 lol
03:12.55mitchelocbut thats cause i prefer broadvoice
03:12.58mitchelocbetter quality etc
03:13.00czero:)
03:13.12iMediaxbout 120/mnth
03:19.15*** join/#asterisk telme (~non@c-24-8-57-124.client.comcast.net)
03:19.38telmeso i installed my t100p and nothing happens. neither the read or the green light come on. what gives?
03:20.02mitchelocheh
03:20.04mitchelocyou have to configure it
03:20.07mitcheloczapata.conf
03:20.15mitchelocand then you do a modprobe
03:20.18mitchelocto get lights up
03:20.50telmebut even if zapata.conf and modprobe was wrong wouldn't the green light come on? (by the way i've already modprob'd successfully).
03:21.02telmesorry not green but any??
03:21.43Kb1_Kanobtelme: you've got zaptel in, but is wct1xxp also loaded?
03:22.09telmemodprobe wct1xxp doesn't return any errors
03:22.19Kb1_Kanobdoes it show in lsmod?
03:22.42telmeas unused yes
03:23.05Kb1_KanobHmmm.... and entries in dmesg are all reasonable?
03:23.13*** part/#asterisk ranliv (~root@203.172.11.239)
03:23.44mitchelocno it wouldn't come on i doubt it
03:23.54mitchelocred might though
03:24.11Kb1_Kanobunless the span isn't declared in /etc/zaptel.conf?
03:24.16telmemitcheloc: that's what i'm askin, neither light comes on
03:24.19telmeloadzone=us
03:24.20telmedefaultzone=us
03:24.20telmespan=1,0,0,esf,b8zs
03:24.20telmefxoks=1-24
03:24.20telmefxsks=1
03:24.26telmethere's zaptel
03:24.27Kb1_Kanobthat will not load.
03:24.38telme???
03:24.39Kb1_Kanobchannel 1 is double-declared.
03:24.53Kb1_Kanobs/b fxoks=2-24
03:24.53telmeoks vs. sks
03:25.05telmeno no
03:25.16telmethere is a t100p and an x100
03:25.24telmethe x100 is the sks
03:25.58telmehere, i'll kill the sks and see what happens
03:26.18Kb1_KanobNot tried t100p w/x100p in same box. Likely need something else between the fxs and fxo lines to make wct1xxp understand it's for a different drive.
03:26.23Kb1_Kanobs/drive/driver/
03:28.02telmei killed the fxsks and still no light (red or gree)
03:28.06telmegree -> green
03:28.46Kb1_KanobSounds like the x100p hardware is confusing something.
03:29.14B0ngFrOghave you tried to put a loop back in the t1 port ???
03:29.31telmewhat i the opposite of modprobe (to uninstall)
03:29.37telmei -> is
03:29.40B0ngFrOgrmmod
03:29.56Kb1_Kanobin reverse order of usages (rmmod wct1xxp; rmmod zaptel)
03:30.13telmei'm gonna remove the wcfxo
03:30.15Legend`or rmmod -r wct1xxp to remove zaptel as well
03:30.35B0ngFrOg;p
03:30.36Kb1_KanobLegend`: heh - you learn something useful every day. Thanks.
03:32.57mitchelocno it shouldnt
03:33.02mitchelocjust do it in order
03:33.05mitchelocyou modprobe wct`1xxp
03:33.12mitchelocthen it's fxsks = 1-24
03:33.22mitchelocthen the next one is fxoks=25
03:33.25mitchelocsomething like that
03:33.41Kb1_Kanobmitcheloc: will the x100p need a 'span' declaration?
03:34.32mitchelocnot sure
03:34.34mitcheloci don't think so though
03:35.27mitchelochey you want to buy a t100p ? =) i've got an extra one =/
03:35.36Kb1_Kanobhow much?
03:36.17Kb1_KanobAlso, country of origin?
03:36.20Kb1_Kanob(of shipment)
03:36.27telmestill nada you guys, thanks for trying to help me
03:36.51mitchelocmake me an offer
03:37.00mitchelocits new unused
03:37.05Kb1_Kanobshipping out of the USA?
03:37.20mitcheloccheck fedex.com i don't know it
03:37.29mitchelocbut if the moeny clears why would i care
03:37.32B0ngFrOgtelme --- are you sure there is no odd refrences in /var/log/messages when you modprobe???
03:37.33mitchelocwhere it goes to
03:37.42Kb1_KanobI would care - taxes, duties etc.
03:38.00telmeB0ngFrOg: i'll check
03:38.05astk_testeranyone here a VoIP provider?
03:38.10Kb1_KanobI meant 'is it shipping from the usa?', sorry. :-0
03:38.19mitchelocyes
03:38.24telmeastk_tester: why, you lookin?
03:38.31astk_testeryes
03:38.59telmecheck out teliax, good prices, I have no complaints
03:39.17Kb1_Kanobmitcheloc: Let me ponder - don't dissapear.
03:40.37astk_testerlooking for around 3000 min/mo for now, with potential for much much more if we decide to deploy asterisk over many of our commercial and residential properties
03:40.43mitchelocok just msg me if i'm not here on aim: mitcheloc
03:40.45astk_testergoing to try it in our office first
03:40.59mitchelocastK: whats your business?
03:41.11astk_testerI'm a real-estate developer
03:42.02astk_testerwe'll have about 100,000 sq. ft. of class-a medical office space this time next year, and about 300 units of residential rentals
03:44.52andy_bhey everyone - question if anyone could help me out - just reloaded Asterisk and reloaded the conf files from my prior working config - not I can't log into FWD (times out - failure to register) - any hints from anyone?
03:45.13andy_bnot = now
03:45.48mitchelocheh where at?
03:46.07heragis there an impedance difference between the headset jack on my cell phone and the headset jack on my cordless phone? the earbud that I have for my cell phone works fine, but if I try to use it for in the jack for my cordless phone, the sound is totally
03:46.22heraglow
03:46.39*** join/#asterisk Sickning (~sickning@transkore.com)
03:46.52astk_testermitchel:  missouri
03:46.53heragattenuated...it's like there's just an impedance mismatch...if that's so, how would I go about finding an earbud that would work for my phone?
03:46.54SickningAnyone around to answer a question ?
03:48.02Kb1_KanobSickning: I take it you haven't been here before?
03:48.08Sickningcorrect
03:48.11Sickning;-)
03:48.32mitchelochey if they want to hire me on the west coast (orange/la county) let me know =)
03:48.33mitcheloci've got to go, be back in 20-30 minutes
03:48.34Kb1_KanobRight - best to come right out with the question.
03:49.21Kb1_Kanobsooooooo?
03:50.17*** join/#asterisk silug (~steve@osiris.silug.org) [NETSPLIT VICTIM]
03:54.18file996
03:56.47*** join/#asterisk PhilM (~a@r42h98.res.gatech.edu)
03:58.47gregwoodNeed Help, Have installed asterisk but can not get the demo to work. Can't hear any voice played from the box. iphone to iphone call works fine but echo and demo and voicemail don't play any voice.
03:59.08*** join/#asterisk B0ngFrOg (~wsmith@67.176.78.68)
04:03.02*** join/#asterisk andrew` (~andrew@adsl-67-119-25-173.dsl.snfc21.pacbell.net)
04:04.03gregwoodis this thing working
04:04.31Kb1_Kanobyes - no one knows the answer to your question at the moment. Or they're all sleeping.
04:04.41gregwoodok thanks
04:15.36docelm0What is the easiest way to create new extensions?   Can I use the management API to do this?
04:15.50florzgregwood: Created the echo extension yourself or default config?
04:16.04gregwooddefault config
04:16.23gregwoodactually copied the /etc/asterisk directory from another machine that is working fine
04:17.01gregwoodip phone call work fine also.  it would appear to be a problem with playing back recorded audio.
04:17.22florzgregwood: How about creating a simple test context with only one extension that answers and then starts the echo app?
04:17.46gregwoodcan try
04:18.44docelm0Can I create new extensions via the Management API?
04:19.44florzdocelm0: I think someone in here said something like that recently ...
04:23.09docelm0I have been out and about working on other things..   But Im guessing from looking at the wiki it cant be done..  Then what is a efficient way to do it?   Create the files and have them included in the main conf's?
04:23.15docelm0or edit the confs directly?
04:25.50gregwoodflorz:  ok, the echo test actually works if I don't do the playback(demo-
04:25.57gregwoodechotest)
04:26.34gregwoodif I include that not only do i not hear the playback it would appear it never goes beyond that point.
04:26.34*** join/#asterisk andreg (~andreg@204.249.177.82)
04:27.54filehttp://www.jonessoda.com/
04:28.27filehttp://www.jonessoda.com/files/sodas.html
04:28.59florzgregwood: Tried to start asterisk with -vvvvvvvv
04:29.01florz?
04:29.11gregwooddone
04:30.02florzgregwood: Then what does it say when you try a playback?
04:31.41gregwood<PROTECTED>
04:31.42gregwood<PROTECTED>
04:31.42gregwood<PROTECTED>
04:32.18gregwoodThe weird thing is I can hear when it begins to play. Sounds like someone picking up a phone but not talking.
04:32.28florzhmmm
04:32.30gregwoodiaxphone also show I am receiving audio
04:32.44*** join/#asterisk docelm0 (~me@116-39.202-68.tampabay.rr.com)
04:33.12florzweird, indeed =:-)
04:33.41florzalready tried a different client (protocol)?
04:34.05iMediaxlol what speed is that? 38081864 bytes received in 3.33 secs (1.1e+04 Kbytes/sec)
04:34.12Kb1_Kanobgregwood: codec problem?
04:34.36*** join/#asterisk ahyanne (yahnee@dialup-222-126-69-37.infocom.ph)
04:34.50florziMediax: about 1.1 MB/sec
04:34.53gregwoodOne the working machine when the deom-echtest payback is done it goes into the echo app.  on this machine it never goes to the echo app, just stays in the playback.
04:34.59iMediaxahh duh
04:35.11florziMediax: really ;-)
04:35.17iMediaxheh smack
04:35.20florziMediax: What software is that? ;-)
04:35.25gregwoodyes I have tried several iax phones and also tried sip
04:35.35iMediaxits off my 1and1 server
04:35.42iMediaxsimple ftp
04:36.26florziMediax: So it's your FTP client telling you that?!
04:37.26*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
04:37.29iMediaxya.. reason why i said that.. i was d/l new kernel, and kernels are big... so i thought it was an error since only took 3.3 secs *shrug* long day.
04:39.17*** join/#asterisk WilliamK (~wkeller@c-24-0-130-60.client.comcast.net)
04:40.18florziMediax: IC - the speed looks much like a 100 Mbps connection ...
04:40.36iMediaxthat was from kernel.org
04:40.39gregwoodKb1_Kanob: the gsm codec works fine when I make a ip phone call. and the echo app works fine. It seems to just be the playback.
04:41.10gregwoodvoicemain does not work either but I assume it is using the playback command under the covers.
04:41.10Kb1_KanobHmmm.. that's interesting.
04:41.10Kb1_Kanobyes, it does.
04:41.19Kb1_Kanobwhen you make an IP call via asterisk?
04:41.31gregwoodcorrect
04:41.55florziMediax: Ah, so were logged in via ssh/telnet/whatever to your server and downloaded the kernel there? Then it even might be that this speed might be correct without a local proxy ... =:-)
04:42.11*** part/#asterisk ahyanne (yahnee@dialup-222-126-69-37.infocom.ph)
04:42.33iMediaxya i'm logged in viz ssh
04:42.40iMediaxvia*
04:42.44florzgregwood: When making a call through *, are you using different codecs with both phones?
04:43.31gregwoodIf I understand you question correctly, No. Both phones are using the gsm codec.
04:43.57Kb1_Kanobis this a recent download from CVS? If so did you pull v1-0 or head?
04:44.38florzgregwood: Then it still might be a codec problem. As long as a stream goes out in the same format it comes in, no recoding is needed, data is basically passed through transparently
04:44.49Sickninganyone know of any good softphones besides DIAX
04:45.26gregwoodI have two version currently. One is a recent download from about a week ago. the other is a copy from another machine. complete copy of /etc/asterisk from other machine. Other machine works fine, this one does not.
04:46.09Kb1_KanobIf you're trying to run cvs head you might have problems.
04:47.02gregwoodI could be wrong but I don't thing it has anything to do with asterisk directly but maybe something to do with the os. os is redhat, same version but different install packages,etc.   Also on different type machine.
04:47.37Kb1_KanobShouldn't affect it. It's all just packet shuffling - no hardware unless you specifically add it.
04:47.44Kb1_Kanob(meaning audio hardware)
04:48.13gregwoodI would agree, but it the same binary and config files on both machines
04:48.23gregwoodit is
04:49.08Kb1_Kanobin that case - are the config files modified correctly in /etc/asterisk ?
04:49.18Kb1_Kanobcould be a networking problem w/the audio streams.
04:50.04Kb1_Kanobwhat is the client?
04:50.24gregwoodthe /config files are identical. No sure what I would modify.  Don't know about the audio stream but again the echo works and I can make a phone call.
04:50.30gregwoodiaxphone
04:51.21Kb1_Kanobplayback is very simple. If it 'hangs' on that step for longer than the duration of the actual audio file it implies there is a problem pushing the data out - ie. that it's waiting on something to move.
04:51.32*** join/#asterisk Mike (~mike@201.135.48.52)
04:52.34gregwoodit does appear to hang. it never comes back from the payback unless I hang up the phone. I have stayed on as long a 10 mins and it never returns from the playback to next step in the dial plan.
04:52.48Kb1_Kanobwhat happens if you use background() instead?
04:52.58gregwoodhave not tried
04:52.59gregwoodhold on
04:54.48Kb1_Kanobgregwood: any chance there is a firewall issue? compare 'iptables -v -L' on both boxes.
04:54.50gregwoodit hangs as well
04:55.15B0ngFrOggergwood .. are you sure the file you are playbacking is on the machine???
04:55.33bkw_are you answering the channel?
04:55.35*** join/#asterisk rollotomnasi (~rollotomn@delmar-209-137-161-171-dsl.cavtel.net)
04:56.01Kb1_Kanobbkw_: would not answering stall the playback? not just prevent the audio from making it out?
04:57.01gregwoodiptables are the same on both boxes. iptables not being used on either.
04:57.37gregwoodbuth return INPUT FORWARD and OUTPUT chains are empty
04:57.55Kb1_KanobYou're getting audio on demo, but not a custom playback and custom voicemail?
04:58.12gregwoodno audio on demo either
04:58.42Kb1_Kanobno audio on voicemail implies something other than a missing Answer() then.
05:00.37gregwoodsorry, in answer to the playback file, yes it is there. I also tried to rename it and I did get a file not found error then.
05:03.33gregwoodIf I dial 8500 for voice mall and enter an mailbox number(even though I can't hear it as for it) It does switch to playing the playback for the password (which I also can not hear).
05:05.08Kb1_Kanobthat really sounds like a codec negotiation issue between the ends. but I don't know iaxphone. sorry,.
05:07.49gregwoodcould it be a timing issue with the codec and the actual hardware I am running on. The machine is an smp compaq proliant with 2 cpu's. I have the quad t1 care installed. have tried removing the quad t1 and using a x100p but the result is the same.
05:11.10gregwoodalso could it have anything to do with the language. Don' t know how languages work as I can only find one version of the sound files.
05:11.35Kb1_Kanobshouldn't be language.
05:12.03Kb1_KanobPlayback should work w/o timing source. It's needed for meetme and other coordinated services.
05:13.18Kb1_Kanobplayback is a tiny application - 42 lines of code.
05:13.22*** join/#asterisk Fpl (~Fpl@201.248.181.147)
05:13.45Kb1_Kanobit starts the stream on the channel and then waits for it to finish before returning.
05:14.04Kb1_Kanobso the stream isn't ever finishing.
05:14.28Fplhi guys
05:14.30Kb1_Kanobwhich implies the audio channel from * -> client that carries the stream isn't ok.
05:14.42Fplanyone using QUICKNET BOARDS ? ( linejack ) ?
05:14.50Kb1_Kanobyou're not using g729 are you?
05:14.58gregwoodWould the echotest and the playback be carried on the same audio channel?
05:15.02gregwoodno to g729
05:15.23Kb1_Kanobthey should be.
05:15.36gregwoodand the echotest works fine
05:15.39Fplanyone using Linejack from Quicknet ????
05:15.40Kb1_Kanobdo you have silence supression turned on somewhere?
05:15.48Kb1_Kanobon the client?
05:16.04gregwoodno
05:16.07Kb1_Kanobdamn.
05:16.58Kb1_KanobCan you try a different client, like diax or x-lite?
05:17.04gregwoodsure
05:17.07gregwoodI have both
05:17.09Kb1_Kanobx-lite is sip, but it would prove out the engine.
05:18.09*** join/#asterisk rene- (~rene-@201.135.255.155)
05:18.20rene-hello boys and girls
05:18.41Fpli have about 7 Linejack boards and would like to know if this is possible. : I had a standard siemens PBX that had capability for 4 PSTN LINES AND 8 extensions. ( all analog ). The PBX burned.....!!! now, i have a box with 7 ISA Linejacks that i wish to implement as my pbx using the standard analog RJ-11 in the linejacks. Use two for incoming lines and on the same token the remaining to transfer the calls to the different extensions..Has anyone done this ?
05:19.07rene-is anyone experiencing problems with voicepulse where the call gets dropped randomly after 1 or 5 seconds with no-apparent reason
05:19.09Fplusing quicknet linejacks ??
05:19.32Kb1_KanobFpl: would the rolm phones work plugged directly into a telco wallplug?
05:20.09Kb1_Kanobie. they're not anything fancy for signalling?
05:20.12Kb1_Kanoblike 4 wire?
05:22.36gregwoodOk, diax has the same results works fine on old server, can do echotest on new server but can't do playback.
05:22.49gregwoodtrying x-lite
05:22.58Kb1_Kanobthat's extremely wierd.
05:27.09*** join/#asterisk drmatrix (~sean@cdm-68-228-9-62.laft.cox-internet.com)
05:28.04drmatrixanybody using a sangoma?
05:28.48gregwoodsame with x-lite except that I tend to hear more static/hissing/background noise during the echotest.
05:29.04Kb1_Kanobnow that's got me stumped.
05:29.10bkw_dfasdfasdfasdfasdfasdf
05:29.33Kb1_KanobIAX clients and SIP clients don't get audio from asterisk, even demo?
05:29.54Kb1_Kanobnoise in echo test is probably just delay/compression artifacts
05:31.35*** join/#asterisk justinnnnnn (~justinm@solid.mpa.net.au)
05:31.37justinnnnnnhey guys
05:31.38justinnnnnnguess wat
05:31.41gregwoodalso playback and background both hang and never return to next step in dial plan
05:31.49justinnnnnnmy server isnt screwed after the gr8 hdparm incident of yesterday
05:31.51justinnnnnnreboot fixed her :)
05:32.05justinnnnnncan someone please help me with my sata hdd's tho.. i cant enable dma.. but i need to do something to speed em up or something
05:32.10justinnnnnnto stop em interfering with my e100p
05:32.21justinnnnnnim getting heaps of event 6 warnings and each time it comes up the pstn person can hear a click..
05:32.46Kb1_Kanobwhich sata drivers?
05:32.52justinnnnnnnot sure
05:32.55justinnnnnnhow do i tell ?
05:33.07Kb1_Kanoberm. lsmod?
05:34.15justinnnnnni think its this..
05:34.17justinnnnnnext3                   73376   2
05:34.51Kb1_Kanobnope.
05:35.03Kb1_Kanobwhat kernel are you using?
05:35.07justinnnnnn2.4-smp
05:35.08justinnnnnnredhat 9
05:35.20justinnnnnnhid                    22308   0  (unused)
05:35.25justinnnnnnusb-uhci               27404   0
05:35.25justinnnnnnehci-hcd               20456   0  (unused)
05:35.28justinnnnnnany of thse 3 ?|
05:35.42Kb1_Kanobperhaps dump output of lsmod into pastebin.ca and bring back the link.
05:36.13*** join/#asterisk BoRiS (boris@wnpgmb01dc2-31-184.dynamic.mts.net)
05:38.15*** join/#asterisk coppice (~Steve_Und@218.198.17.210.dyn.pacific.net.hk)
05:38.36justinnnnnnhttp://www.pastebin.com/108890
05:39.32Kb1_Kanobodd - looks like its monolithic (ie. not a module)
05:39.46Kb1_Kanobgrep dmesg for ide and paste it in a bin.
05:40.37heragI have a curious problem with my current setup, cell phone users who have t-mobile and cingular can't call me, all other types of incoming calls seem to work just fine...i'm pretty sure it's a configuration problem on my side, because if I connect my sipura directly to my vsp, then these celll users can call me just fine, any ideas?
05:41.18justinnnnnnhttp://www.pastebin.com/108891
05:41.27justinnnnnntheres the grep ide
05:42.33*** join/#asterisk habakuk (~chatzilla@adsl-64-168-20-150.dsl.snfc21.pacbell.net)
05:42.41drmatrixherag, who owns the number they are dialing?
05:42.42Kb1_Kanobhmmm... which drives are joined w/jbod?
05:42.52justinnnnnnnot sure..
05:42.56heragdrmatrix: huh?
05:43.08heragdrmatrix: the number is mine, it's a broadvoice line
05:43.22drmatrixand only tmobile/cingular users can't call?
05:43.49heragdrmatrix: yes, but they can't call only when I use my asterisk server, if my sipura goes directly to broadvoice, they can call
05:43.54drmatrixwhat's a call to you sound like to them?
05:43.58drmatrixwith asterisk
05:44.04heragdrmatrix: a bunch of little quick beeps
05:44.27drmatrixdo you have a verbose log of asterisk when the call comes in?
05:44.30heragdrmatrix: I can even see their call come in on the cli, it terminates after the third exten => command, usually
05:44.49drmatrixwhich is/
05:45.08herag-- Executing Answer("SIP/147.135.0.129-08137430", "") in new stack
05:45.08herag<PROTECTED>
05:45.08herag<PROTECTED>
05:45.28heragafter the waitexten, I get a spawn extension...exited non-zero
05:45.54grrantI think I accidentally deleted a line in sip.conf now Xlite doesnt work with asterisk it says "No compatible codecs" and then plays a busy signal
05:46.46gregwoodyou can turn the codecs on and off in x-lite by clicking them in the display window of x-lite.  make all the codec black instead of white or grey.
05:46.54heragit's really peculiar, cause verizon cell users can call, at&t can call, all the land lines I've tried can...it's just really bizarre
05:47.02*** part/#asterisk Fpl (~Fpl@201.248.181.147)
05:47.10drmatrixwhat is the next step when other users call
05:47.39heragdrmatrix: normally? assuming they don't push a new extension on the stack?
05:47.58drmatrixi assume the beeping occurs before 2 secs pass...
05:48.07Kb1_Kanobherag: how is the call coming in? what interface/hardware?
05:48.21heragdrmatrix: I'm not sure about that, it all happens so fast, I can't be sure
05:48.27grrantGregwood: I tried that but it still has the sameproblem
05:48.41heragKb1_Kanob: it's all sip, broadvoice number gets dialed, goes to my asterisk server
05:49.00drmatrixgrrant: try allow=gsm in the sip.conf for that user
05:49.12Kb1_Kanobhow long before the hangup (from the time the caller hits send)? and is it consistent?
05:49.16heragdrmatrix: the next step in the extension, assuming no new extension gets pushed on is stopplaytones, and then a Dial(...)
05:49.38drmatrixherag, msg me with the number; i can try with my cingular
05:49.41heragKb1_Kanob: I believe it's happens as soon as the caller connects
05:50.23habakukanyone know what "realtime" support is referring to in chan_sip?
05:50.47heragKb1_Kanob: and I cannot be sure if the duration happens consistently, because generally a cell phones takes a varying amount of time to connect to a call
05:50.58Kb1_KanobI'm having a problem with allstream/rogers, but it's an analog answer supervision problem on my channel bank. Manifests itself as hangup after exactly 60 seconds. Not a problem for other carriers.
05:50.59grrantallow=gsm is already there
05:51.14grrantI just typo'd something in sip.conf and this happened
05:51.34heragdrmatrix: you connected?
05:51.42drmatrixyep... music?
05:51.46heragyes
05:51.51heraghmm
05:51.58heragthis is curious
05:52.11drmatrixonce more; another cingular
05:52.12Kb1_Kanobherag: if it was a telco supervision problem w/broadvoice it would be timered from the pressing of send, not connecting to you.
05:52.29grrantWeird I just called my DID it rang my softphone Xlite
05:52.31drmatrixworks there 2
05:52.36heragyup
05:52.40grrantBUt when I try to call out with Xlite it says "no compatible codec"
05:52.49heraghmm
05:52.55heragnow I don't know what to make of this
05:52.57grrantMy account is at only $1.90 but that should be sufficient plus Im calling 800#s
05:53.32grrantWhat happens in xlite when you select (or make black) all the codecs, how does it select which one it wants to go to
05:53.34heragsee, I only know what my friends tell me, I'm on verizon, and it works...but I have friends on t-mobile and cingular that say they can't get through
05:53.40heragand they have similar problems
05:53.59heragthanks for testing that drmatrix
05:54.15heragthough this makes my problem a little harder
05:54.30drmatrixthey just dont hear the music i'd bet.
05:55.04drmatrixyour ring runs into the original ring... it's hard to tell when asterisk picks up
05:55.23grrantHow could xlite and asterisk work to receive calls through a DID but not to make calls from asterisk?
05:55.25drmatrixi'd change the ring tone with a background voice prompt, just to be sure they can tell
05:55.44heragdrmatrix: no, that music is what was coming out of my speakers
05:55.52heragdrmatrix: I physically picked up that call
05:56.08drmatrixinstead of the waitexten?
05:56.45grrantAnyone else having problems making outbound calls with Connect.Voicepulse
05:57.03drmatrixah, they don't support all the codecs they say they do
05:57.05heragdrmatrix: the waitexten only lasts for two seconds, if you don't push anything on the stack, it'll go to dial
05:57.22drmatrixgrrant, i'd try ulaw everything until it's working
05:57.32heragdrmatrix: now here's an even more curious situation, I just told one of my friends who can't connect to dial me
05:57.41heragand I took out the waitexten stuff
05:57.53heragso all I have is 1,Answer() and 2,Dial()
05:58.08heragand it rang my phone for a second, and then just hung up
05:58.13grrantDo i just add disallow = all & allow = ulaw in sip.conf
05:58.18heragit's almost like something's timing out
05:58.21grrantOr in iax, and everywhere else I can including xlite
05:58.23drmatrixgrrant, yes
05:58.41grrantiax?
05:59.07Kb1_Kanobherag: can you debug where the hangup is occuring in the console? Ie. is broadvoice hanging up the call?
05:59.08drmatrixsame
05:59.23heragKb1_Kanob: I don't know how to determine where the hangup is happening
05:59.28heragKb1_Kanob: how would I check?
05:59.34drmatrixwhat was the log on that last call?
05:59.53grrantMe?
05:59.58drmatrixno, herag.
06:00.01heragdrmatrix: you mean the cli output?
06:00.04drmatrixyou: same for iax and sip .conf
06:00.07drmatrixyes
06:00.12grrantYea I didit, still not working
06:00.12drmatrixset verbose 5 perferred
06:00.17grrantBut my DID still rings to xlite
06:00.19grrantHow can this be
06:00.24drmatrixstill the xlite error?
06:00.24heragoh, verbose 5, ok I'll try that
06:00.52Kb1_Kanobherag: or sip debug peer .....
06:00.58grrantNo xlite error
06:01.01grrantWell yea
06:01.04grrant404 not found
06:01.22drmatrix?
06:01.37grrantThe error is in xlite
06:01.44grrantit says Call failed: 404 not found
06:01.45grrantXlite error, before I did these modifications it would give me an CLI error saying "no compatible codecs" and everythin was working fine butI
06:01.56grranttypo'd something in sip.conf and saved/closed it
06:02.07drmatrixsip reload from the console
06:02.08grrantI've looked through sip.conf for syntax errors but nothing
06:02.19grrantWhat does sip reload do?
06:02.24drmatrixrereads the file
06:02.28grrantyuh
06:02.34grrantI already did that with the changes you said
06:02.39drmatrixno error?
06:02.59grrantNot in CLI
06:03.10grrantUnless I need to set verbose to 100 first
06:03.14grrantNow the error is in xlite
06:03.19grrant404
06:03.28riksta"there is no error"
06:03.37rikstas/error/spoon/
06:03.42grrantHuh?
06:03.48drmatrixset a verbose, and see if you are dialing an invalid extension
06:03.57rikstait's the matrix
06:04.11heragok, progress has been made...if I have just a single dial command in my extensios     1,Dial(...) they can get in
06:04.32grrantits set to verbose
06:04.36grrantI did asterisk -rvvvvc
06:04.43grrantWhat level is that considered: 4?
06:04.46drmatrixyes
06:04.58heragbut if I want to do anything else, it won't work....I'd like to pick up the phone...1,Answer   2,Dial()....why on earth won't it work with something like this?
06:05.38Kb1_Kanobherag: Paste the exact (unworking) in pastebin
06:05.44heragok, one sec
06:06.15drmatrixherag, you might want to add the obligatory wait,2 right after the answer
06:06.18bkw_kram
06:06.25heraghttp://pastebin.ca/1358
06:06.28bkw_I swear he doesn't pay attention at all sometimes
06:08.34Kb1_Kanobherag: they are always hung up on, even without trying to hit a dtmf key?
06:08.52habakukhi bkw_ can you tell me what you mean by "REALTIME" extensions to chan_sip.c ? how is that better than MYSQL_FRIENDS? is it just a more modular approach?
06:08.58heragKb1_Kanob: yes, they never try to push a dtmf
06:09.07heragKb1_Kanob: yet they still are dropped
06:09.52heragKb1_Kanob: asterisk would tell me if they pushed a dtmf and are now on a different extension, I mean, the calls are dropped even if all I have is 1,answer 2,dial()
06:10.10Kb1_Kanoband if you simply change 3 to 'exten => s,3,noop(WaitExten(2))'?
06:10.15bkw_habakuk go read
06:10.18bkw_I can't really explain it
06:10.19heragbut the calls _work_ if all I have is 1,dial()
06:10.22bkw_its alot more modular
06:10.24Kb1_KanobHericom: ah, sorry.
06:10.33Kb1_Kanobherag, even.
06:10.35drmatrixherag, just for giggles, does 1,answer; 2 dial work without the ,r ?
06:11.19bkw_its ,,r
06:11.21heragdrmatrix: I'm not sure, I will have to try
06:11.22bkw_recall the timeout
06:11.33drmatrixhe had a 30, bkw
06:11.34bkw_IAX2/blah,,r
06:11.38bkw_oh ok
06:11.40bkw_just checkin
06:11.40drmatrixi'm happy leaving that
06:11.44heragKb1_Kanob: I nooped it, nothing
06:11.48heragKb1_Kanob: still dropped
06:11.49habakukbkw_, sure. do you a have a bug# where I can read more about? My main beef with MYSQL_FRIENDS, is there doesn't appear to be a way to create a sip user of type "friend"
06:12.02habakuki.e. just user or peer
06:12.20bkw_friend is a user and a peer at the same time
06:12.25Kb1_Kanobherag: try that wait(2) as s,1,... suggestion.
06:12.48drmatrixi like that one
06:13.21habakukbkw_, right, but I can't figure out how to set type=friend from a database. Unless I add a new field to my sipfriends table or something..
06:13.22heragKb1_Kanob: is the wait(2) before or after the answer?
06:13.24Kb1_Kanobdrmatrix: you think far end is getting confused by the fast answer?
06:13.29Kb1_Kanobherag: before.
06:13.35bkw_habakuk it implys friend
06:13.35heragdrmatrix: I tried the removal of the ,r      no good
06:13.38bkw_no need to do type=
06:13.40heragKb1_Kanob: ok, I'll try it
06:13.45bkw_thats why its called "sipfriends"
06:13.48drmatrixyea... i had an extention that played a gsm instead of ringing...
06:13.58drmatrixworked for everyone but a few callers until i answer()ed
06:14.29drmatrixand if i remember right, it was particular cellular users...
06:14.30habakukbkw_, but if you look closely at the code,  it doesn't allow you to things like callerid and mailbox for example
06:15.16heragdrmatrix: sorta like what I'm experiencing?
06:15.25drmatrixhabakuk: and res_odbc doesn't do everything you want?
06:15.32grrantThanks for you help I found out it was just a typo in extensions.conf
06:15.34drmatrixherag: it rhymes....
06:15.41grrantinstead of NXXNXXXX i had NXXNfNXXXX
06:15.56bkw_habakuk the realtime does
06:16.00heragKb1_Kanob: I tried the wait(2) like you said
06:16.03bkw_MYSQL_FRIENDS no
06:16.03grrantIs there a way to setcallerID for each call you make without opening sip or reloading asterisk
06:16.03heragKb1_Kanob: nothing
06:16.05bkw_realtime YES
06:16.09bkw_realtime supports all fields
06:16.13bkw_you just add them to the database
06:16.15bkw_and they are there
06:16.23bkw_so if you want callerid
06:16.26bkw_you add the field
06:16.28bkw_and its there then
06:16.31heragdrmatrix: your description really sounds like my problem...how did you get around it?
06:16.31Kb1_Kanobherag: and the sipura behaves for the same callers. Wierd.
06:16.32habakukbwk_ at least callerid, as this is not part of a peer, but a user.
06:16.42drmatrixanswer, wait, then send audio
06:16.43heragKb1_Kanob: yes, the sipura works for everyone
06:16.46bkw_I use friends and it works fine
06:16.52bkw_the relatime friends that is
06:17.31heragdrmatrix: would it matter if I do wait() answer, playtones... or answer, wait() playtones...?
06:17.33habakukbkw_, hmm so is there a new struct defined? How do you combine attributes of user and peer? is what I'm getting at...
06:18.11drmatrixi dunno. i had to answer, then wait(2), then playtones
06:18.24heraghmm
06:18.35heragdrmatrix: well, I'll try it
06:18.40drmatrixi've looked through it, and it's not in my current setup anymore, i was playing with stuff then
06:18.42bkw_habakuk you must be misunderstanding the code or something
06:18.48bkw_hold please
06:18.52Kb1_Kanobherag: sorry, you're beyond me with this one. You might need to talk to broadvoice for a trace but have a 100% non working cellphone handy.
06:19.03heragKb1_Kanob: : /
06:19.23heragKb1_Kanob: that's hard to manage...considering that it's not my cell that's experiencing the problem
06:19.28drmatrixthe closest i have now is http://pastebin.ca/1359
06:19.44bkw_yep
06:19.50bkw_habakuk you can do all that stuff with realtime
06:19.59heragdrmatrix: and that works for those people that were having problems?
06:20.01bkw_callerid and all settings
06:20.17Kb1_Kanobherag: yes, that can be a pain. But without repeatable fault telco debug isn't going to happen. :-/
06:20.29drmatrixit works for everyone, with some hearing nothing, some hearing a full ring tone before connecting
06:20.43drmatrix0.9 was the lowest i could put it
06:21.05drmatrixtiming.
06:21.31herag?
06:21.40drmatrixthe answer to your wonders
06:22.00habakukbkw_, you could be right. My point is now with the code there is mysql_user() and mysql_peer(). at registration time, it's always calling mysql_peer() only and doesn't grab the callerid from my database
06:22.11heragwhat does timing have to do witht his?
06:22.13herag*this
06:22.36drmatrixwith the complications? or with your problem?
06:22.38habakukbkw_, hmm maybe I should check out the realtime stuff. Is there a bug # I can reference? or is it already in CVS? (I'm using 1.0.1)
06:22.43heragmy problems
06:23.02drmatrixi think your asterisk is too fast for some providers
06:23.11heraghmm
06:23.14Kb1_Kanobherag: wierd things start to happen with inter-carrier calls. Many things that would never be an issue otherwise crop up. you're probably spanning three carriers.
06:23.17drmatrixi had some callers that could hear music BEFORE i answered the phone
06:23.28drmatrixothers would get disconnected
06:23.35heraghaha    ; P
06:23.38drmatrixand still others heard nothing until the answer
06:24.08docelm0can someone tell me where I can get more info on how to setup the conf's in the database?   I think it was res_config?
06:24.22heragI think the very fact that we can do this is causing so many problems on the administration on the telco side, I don' think they know what to do with all of it
06:24.43drmatrixres_config_odbc or res_odbc; not sure which
06:24.46Kb1_Kanobheh... that's an understatement.
06:25.00docelm0thanks matrix..
06:25.14habakukdrmatrix, I thought there were more options besides odbc ?
06:25.16docelm0Now if I can only get ODBC to work on linux..
06:25.18docelm0ack..
06:25.27heragthose old monster bells need to release some docs that explain how their networks are configured so we can acutally interface with them...but they'll never do that, so we're just gonna have these fun problems
06:25.33drmatrixsure, if you want to read the source yourself
06:25.53Kb1_Kanobherag: oh, they do.
06:25.53drmatrixherag: id be scared if I were them
06:26.12habakukdrmatrix, is this checked into cvs yet, or is this still a patch?
06:26.19drmatrixuh, good question
06:26.33drmatrixi'm on CVS-HEAD-10/03/04-17:32:36,
06:26.35Kb1_Kanobherag: it's getting them without paying > $1k for a copy...
06:26.52Kb1_Kanobherag: http://www.voip-info.org/wiki-Telco+Engineering+Information
06:26.52bkw_habakuk realtime is cvs-head only
06:26.53heragscrew them...that should be public info
06:27.04Kb1_Kanobgoogle is your friend. ;-)
06:27.16drmatrixbut 1.0.0 has them both too
06:27.29docelm0Anyone in here good with ODBC and Linux?
06:27.32bkw_yes
06:27.35habakukdrmatrix: the one I was talking about was here: http://svn.asteriskdocs.org/res_data/ast_data/ not sure if this is the same thing you are referring to
06:27.45bkw_docelm0 whats the problem?
06:27.46docelm0You wrote the damn hack.. I hope so.. :)
06:27.52bkw_hack?
06:27.53bkw_what hack?
06:27.55docelm0I cant get ODBC to work
06:28.01bkw_haha
06:28.02docelm0well..  res_odbc
06:28.09bkw_I didn't write that
06:28.13bkw_tony did
06:28.15docelm0I thought you did..
06:28.15bkw_I wrote cdr_odbc
06:28.15docelm0ohh
06:28.23bkw_so I do know odbc
06:28.28bkw_but what are you trying ot get working?
06:28.32bkw_cvs-head or 1.0.1?
06:28.35drmatrixdocelm0: whacha need, whatsit worth?
06:28.46BoRiSshow me the money!
06:29.00bkw_and in the recent changes in res_odbc and res_config_odbc
06:29.12herags,1,dial() all by itself works...
06:29.12drmatrixbkw_: it's looking nice
06:29.14drmatrix(er)
06:29.14heraghow is it working?
06:29.21docelm0well either way.. I am figuring in stead of using the flat files to add/delete users I thought I would use the DB aspect..  Faster and my code to interface w/ it will be cleaner..
06:29.23heragit must be sending something that the other sutff isn't
06:29.24drmatrixwith the ,r?
06:29.28heragdrmatrix: yes
06:29.32docelm0but I want to dump my sip.conf and extensions.conf into the DB
06:29.40docelm0so I can work on the fly..
06:29.51docelm0add users and dialplans etc..
06:30.03bkw_www.bkw.org/load.txt
06:30.06drmatrixtry 1,answer,2,ringing;3,dial
06:30.11bkw_that should load into ast_config for you
06:30.25heragringing...what does ringing do?
06:30.32bkw_change the indication
06:30.32drmatrixsame as ,r
06:30.33bkw_to ringing
06:30.40bkw_no accually its not relly the same
06:30.49bkw_Ringing can just provide ringing.. ie fake it
06:30.52drmatrixer, r just does ringtones?
06:31.00heragoh...hmm
06:31.03bkw_you can smack a wait on that and play like you called someone
06:31.05docelm0ok bk.. what am I supposed to do with this perl script?   Is there anything I need to setup?  install etc?
06:31.07bkw_totally fool people
06:31.12heragwhat the difference between ringing and playtones(ring) ?
06:31.18bkw_www.bkw.org/~brian/res_config.doc
06:31.21bkw_have fun
06:31.29bkw_er
06:31.33bkw_www.bkw.org/~brian/doc/res_config.doc
06:31.40bkw_BEEFCAKE!!!
06:31.40*** part/#asterisk gregwood (~lynnwood@pcp01073653pcs.andrsn01.tn.comcast.net)
06:31.41drmatrixplaytones is just noise; ringing is status.
06:31.54bkw_well Ringing DOES NOTHING but fake it
06:32.00bkw_,r on a dial is accually dialing something
06:32.05bkw_yes it does ringing
06:32.25bkw_but its diffrent in the way you can just do Ringing with a wait and totally never dial someone
06:32.36heragwell, it's worth a shot
06:32.36bkw_but to the caller it sure would sound like it
06:32.46drmatrixwhat i do is immaterial, i'm talking about the channel's picture here
06:33.13bkw_Thank you Mr. Mayor
06:33.36drmatrixit's a nitpicky thing, with a nitpicky difference
06:33.41drmatrixi've been bit by it
06:33.58bkw_so why are you needing the r anyway?
06:33.59bkw_whats up?
06:34.00heragdrmatrix: I hope that it's nitpicky enough to make a difference    ; P
06:34.25drmatrixi do too, because the headache isn't worth it
06:34.28heragwhy do I need the ,r? cause otherwise my callers think nothing's working and just hang up
06:34.38bkw_and you're going out sip?
06:34.40bkw_or iax?
06:34.55drmatrixsneaky
06:34.59heragbkw_: I don't know who you're talking to, but I'm going out sip
06:35.05bkw_ok
06:35.07docelm0so bk..  The perl script..  It will dump sip.conf or whatever conf I need to dump into the db?
06:35.09bkw_going to cisco gear?
06:35.17bkw_docelm0 yes
06:35.27heragbkw_: me? cisco?
06:35.37bkw_herag yes going to a cisco gateway?
06:35.40docelm0sweet..  this is gonna be MUCH FREAKING easier than what I have been racking my brains about..
06:35.46heragbkw_: broadvoice -> asterisk linux -> sipura device
06:35.54bkw_OH broadvoice
06:35.59drmatrixheh
06:36.09bkw_have you looked at sip.conf.sample?
06:36.10bkw_progressinband=no              ; If we should generate in-band ringing always
06:36.29heragI don't have that setting in there
06:36.31bkw_you should try that
06:36.32heragshould I?
06:36.34heraghmm ok
06:36.39bkw_in /usr/src/asterisk/configs/sip.conf.sample
06:36.41bkw_it sure is
06:36.42bkw_:)
06:36.48bkw_ya I would try it
06:37.01bkw_but if you don't get ringing then broadvoice needs to get that fixed
06:37.06bkw_they might not be responding correctly
06:37.11bkw_a sip debug would be most helful
06:37.12bkw_er helpful
06:37.24heragbkw_: ?
06:37.33heragbkw_: my problem doesn't have to do with ringing
06:37.50bkw_you said they don't hear anything right?
06:37.52drmatrixthat may be why i needed my ringing.
06:38.06bkw_no ringing?
06:38.06heragbkw_: no, no...their calls get dropped
06:38.11*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-218-236-246.red.bezeqint.net)
06:38.14bkw_god be clear about it then
06:38.23BoRiSlol
06:38.26drmatrixheh
06:38.37bkw_its like pulling teeth in here sometimes..
06:38.42heragbkw_: no offense, but I was...go back to the beginning, drmatrix and Kb1_Kanob are quite clear about my issue
06:38.46drmatrixlmao
06:38.53rikstawith the default configuration of installing asterisk, can one test if it can receive a connection on iax2? i just want to see if i have set up my routers port forwards correctly, before i start to continue
06:38.55bkw_you think I have time to read everything
06:39.01bkw_this channels S/N is too high
06:39.02BoRiSthen ask them instead.
06:39.06drmatrixherag: i think bkw_ does know where you are
06:39.18bkw_I thought I knew exaclty what you were talking about .. but thats ok
06:39.20drmatrixjust some details are missing
06:39.27bkw_inbound not outbound right?
06:39.31drmatrixbkw_ his calls drop from some users inbound
06:39.34heragbkw_: yes, inbound only
06:39.46bkw_ok if you get dropped calls we need a sip debug
06:39.50bkw_of a dropped call
06:39.56bkw_I suspect maybe its coming from a diffrent IP
06:40.05drmatrixi think the callers' providers are unhappy by call progress from broadvoice+herag's * setup
06:40.07bkw_and not matching a context/peer and gets rejected maybe?
06:40.12Kb1_Kanobherag: ringing means two things - sound in channel (to indicate to user) and signal on channel (to indicate to far end pbx). The audio will be ignored by far end, but the signalling may be confusing things.
06:40.26herago...hmm
06:40.44Kb1_Kanobbkw_: some cingular wireless inbound calls are disconnected but not all when answer() is in the dialplan.
06:40.44heragbkw_: it's only some cell phone users, specifically some t-mobile and some cingular users
06:40.45bkw_accually I think its ip issues/peer matching
06:40.58bkw_herag haha good luck I know whats wrong but you'll have hell fixing it
06:41.16heragbkw_: if I do strictly a s,1,Dial() it will work, but if I s,1,answer() first, it'll drop
06:41.23Kb1_Kanobbkw_: cross-carrier answer supervision?
06:41.33bkw_herag sip inbound to sip?
06:41.43heragbkw_: huh?
06:41.44bkw_s,1,Answer then playback a file doesn't work?
06:41.50heragno, that will drop
06:41.56bkw_get a sip debug
06:41.59bkw_something is'nt right then
06:42.24BoRiSwhat is your * version or cvs date?
06:42.32heraghmm, ok...I turned on sip debug at the command line, how do I get the debug?
06:42.43heragBoRiS: it's 1.0RC1
06:42.43bkw_make a call that gets dropped
06:42.46bkw_OH GOD
06:42.47bkw_get 1.0.1
06:42.56bkw_1.0.1 has ALOT of fixes
06:42.58BoRiSalways a GOOD idea to upgrade to the latest
06:43.13BoRiS(well, MOST of the time...hehe)
06:43.25Kb1_Kanobbkw_: has the v1-0 vs. v1-0-1 debate been resolved?
06:43.32bkw_use v1-0
06:43.34bkw_as I said
06:43.38bkw_nothing needed to be resolved
06:43.40Kb1_Kanobnoted.
06:43.52bkw_just people seem to not wanna pay any attention to me
06:43.58bkw_but thats ok
06:44.00bkw_:p
06:44.04BoRiSwe luv you!
06:44.07bkw_I know honey
06:44.17bkw_luv you too
06:44.20BoRiS:)
06:44.32bkw_guess people don't know how close I am to all the dev work on asterisk
06:44.42bkw_and how involved I am in everything
06:44.46bkw_haha
06:44.53bkw_ya when I get going I might not make much sense
06:44.58bkw_but I got alot of info
06:45.07bkw_my fingers can't keep up with my mind sometimes :P
06:45.11Kb1_Kanobinfo is good.
06:45.20cypromisyes and no
06:45.45cypromistoo much info about latest cvs changes
06:45.53*** join/#asterisk af_ (~af@62.94.148.227)
06:45.55cypromiscan make jo average user go nuts
06:45.55cypromislol
06:46.00bkw_ya really
06:46.20Kb1_Kanobbut it's good to understand why they're happening.
06:46.27Kb1_Kanoband tidbits are better than nothing.
06:46.39bkw_it does clue you in on what you might or might not like
06:46.42habakukbkw_, yeah you're a great resource
06:46.44bkw_you can get pointed in the right direction
06:46.55bkw_which is what alot of people need
06:46.55rikstacan someone test my IAX2 connection on       81.178.236.177 please?
06:47.01bkw_need more info
06:47.12bkw_IAX2/user@ip/exten
06:47.16bkw_gotta have info
06:47.26bkw_ie IAX2/guest@pbx.moosepenis.com/996
06:47.29bkw_:P
06:47.31BoRiSHey!
06:47.33bkw_aka pbx.bkw.org
06:47.36BoRiSSssssssssssssh
06:47.45rikstaahh sorry, i dont know what to do, i've just installed asterisk, followed the FWD IAX setup guide
06:47.52Kb1_Kanobjust the philosophy behind decisions is very helpful - like why log levels are split the way they are...
06:47.55rikstabkw_ what do i need to make an ext?
06:48.09bkw_riksta you have alot of reading and playing to do
06:48.19bkw_ok simple
06:48.19bkw_iax.conf
06:48.19bkw_you see [guest]
06:48.22bkw_and it has context=
06:48.32bkw_that context= matches [something] in extensions.conf
06:48.33rikstayes
06:48.35bkw_which has an extension
06:48.35bkw_ie
06:48.42rikstaok, i follow
06:48.46rikstacontext=default is mine
06:48.51bkw_exten => s,1,Answer and exten => s,2,Playback(demo-congrats)
06:48.53drmatrixriksta: what's your fwd number then?
06:48.59riksta479332
06:49.08bkw_riksta ok you're good then :)
06:49.14BoRiSexten => s,3,Playback(moosepenis)
06:49.26bkw_show-me.gsm is better
06:49.32bkw_www.bkw.org/show-me.gsm
06:49.36rikstabkw_ i don't understand that exten => command
06:49.51bkw_riksta look at the examples included in the extensions.conf.sample
06:50.00bkw_its alot to take in
06:50.01heragthat's a lotta output, I'll have it pasted in a sec
06:50.13rikstashould i set context= demo
06:50.19bkw_herag update your asterisk first
06:50.40bkw_show me on the dolly where the bad IVR developer touched you!
06:51.03bkw_ok i'm sleepy
06:51.05bkw_bed time
06:51.06bkw_ttyl
06:51.09rikstaOct 10 07:50:46 WARNING[98311]: requested inkey 'freeworlddialup' for RSA authentication does not exist
06:51.12rikstai got this error
06:51.23rikstaahh wait i missed making the RSA key
06:51.23drmatrixnite, bkw
06:51.37drmatrixi guess from my call?
06:51.47rikstaplease can whoever called, try again
06:51.52riksta65.39.205.121
06:52.05drmatrixit's by your fwd number... still bad
06:52.16rikstahuh?
06:52.23rikstai restarted asterisk just then
06:52.26rikstapls try
06:52.31drmatrixyou are setting iax up with fwd, right?
06:52.37rikstayeah, trying to :S
06:52.46heraghttp://pastebin.ca/1361 this is the sip debug, anyone have a clue what went wrong?
06:52.48rikstaOct 10 07:52:41 NOTICE[98311]: Rejected connect attempt from 65.39.205.121, requested/capability 0x4/0x4 incompatible  with our capability 0xff0
06:52.50drmatrixso anyone that can call your fwd can check...
06:52.59drmatrixer, gsm not there?
06:53.15rikstado i have to enable gsm codec?
06:53.23rikstai think i allowed ulaw (if that is the same kinda thing)
06:53.25BoRiSallow=all in case
06:53.27rikstaok
06:53.31drmatrixi'm gsm preferred.
06:53.38rikstaone moment pls
06:53.50BoRiSjust for testing, when it works then you should be specific
06:54.02rikstaok restarted with allow=all
06:54.17BoRiStry that
06:54.20drmatrixbusy
06:54.28rikstasame error message
06:54.45BoRiSis their anything that says "disallow=all?"
06:54.50rikstasec
06:54.51drmatrix<PROTECTED>
06:54.51drmatrix<PROTECTED>
06:54.51drmatrix<PROTECTED>
06:54.51drmatrix<PROTECTED>
06:55.00Kb1_Kanobherag: what is s,3, currently defined as?
06:55.16rikstajust dissallow=lpc10
06:55.36rikstadisallow*
06:55.36heragKb1_Kanob: s,3,WaitExten(2)
06:55.37drmatrixwhat do you have for context=
06:55.40rikstademo
06:55.51rikstashall  i change back to default?
06:56.07drmatrixcan you set verbose?
06:56.18rikstaoh hold on no thats wrong im sorry
06:56.21rikstai have for context
06:56.26Kb1_Kanobherag: note 'Spawn extension (incoming, s, 3) exited non-zero ....' followed by 'Destroying call 'SD30qg701-d6bc9152f0059d0aa2c8967461798787-js11002'
06:56.28riksta[fromiaxfwd]
06:56.39rikstaand can i paste the 3 lines that are in extensions.conf?
06:56.40BoRiSmake sure that is set in your context=
06:56.52rikstayes it is set in my context=
06:56.53BoRiScontext=fromiaxfwd
06:56.55docelm0you know if this actually works I am gonna be happy and amazed..  I have NEVER had any luck with unixODBC..
06:57.01heragKb1_Kanob: ya, I've been seeing that, what am I supposed to understand from it?
06:57.07rikstacan i sho you the three lines in extentions.conf for [fromiaxfwd] ?
06:57.09Kb1_Kanobherag: that is the call being hung up by ast. note line 047 in pastebin. that is the matching ID.
06:57.14BoRiSyeah
06:57.18rikstaexten => ${FWDNUMBER},1,Dial(${FWDRINGS},20,r)
06:57.18rikstaexten => ${FWDNUMBER},2,Voicemail,u${FWDVMBOX}
06:57.18rikstaexten => ${FWDNUMBER},102,Voicemail,b${FWDVMBOX}
06:57.26heragKb1_Kanob: the cli keeps telling me that it blows up at the wiatexten(2) line...the question is why?
06:57.30BoRiSreplace ${FWDNUMBER} with s
06:57.41BoRiSexten => s,1,Dial(${FWDRINGS},20,r)
06:57.46heragKb1_Kanob: and why only for these certain callers?
06:57.50rikstai had FWDNUMBER defined above
06:58.07Kb1_Kanobherag: agreed.
06:58.07rikstacan i just ask, the FWDRINGS, do i have to have some client set up connected into asterisk?
06:58.20drmatrixherag: try insecure=yes in your sip.conf.
06:58.25rikstai have this set         FWDRINGS=sip/office ; the phone to ring              what is sip/office
06:58.35heragdrmatrix: it already is
06:58.41drmatrixhmph.
06:58.49BoRiSmake sure office is a useraccount defined in sip.conf
06:59.00drmatrixFWDRINGS and FWDNUMBER need to be set in globals for that exten to work
06:59.07rikstaok do i make it like [office]     or [sip/office]
06:59.21drmatrixwhat phone are you using with *?
06:59.24BoRiS[office]
06:59.26heragmaybe it's just a bug, and I have to update...I think I've banged my head against the wall quite enough for tonight, I'll make the upgrade tomorrow
06:59.27Kb1_Kanobherag: before ast gives up gathering digits broadvoice is termianting the call (lines 047  Call-ID: SD30qg701-d6bc9152f0059d0aa2c8967461798787-js11002
06:59.29drmatrixsip/office it is
06:59.36rikstaim going to use a softphone, what do you recommend
06:59.37Kb1_Kanoboops. lines 135 onwards.
07:01.12*** part/#asterisk rollotomnasi (~rollotomn@delmar-209-137-161-171-dsl.cavtel.net)
07:01.51drmatrixsjphone
07:01.56drmatrixbut x-lite is pretty
07:02.25drmatrixKb1_Kanob I was starting to wonder how bad my sip was
07:02.29Kb1_Kanobherag: bkw_ may have been refering to a silence detection issue w/broadvoice (http://lists.digium.com/pipermail/asterisk-users/2004-August/060340.html) but that's not your issue.
07:02.43BoRiSriksta: Maybe try setting this a test in your extensions.conf exten => s,1,Answer   exten => s,2,Playback(demo-congrats), then dial your fwd # and you should hear something
07:02.48Kb1_Kanobdrmatrix: oh?
07:02.57drmatrixwjat
07:03.01drmatrixwhat's the point of line 39?
07:03.35BoRiS(of course as long as you have your fwd setup in your extensions.conf for dialing other fwd users)
07:03.36drmatrixi don't recall completing a call when that happens
07:03.36BoRiSoy
07:03.52Kb1_Kanobdrmatrix: indeed. broadvoice appears from multiple origins?
07:04.10drmatrixthat's what bkw_ alluded to
07:04.24Kb1_KanobAhhh... yes. I see now.