IRC log for #asterisk on 20071017

00:02.54*** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net)
00:04.48bmddoes 1.2 allow for easily adding firewall rules on top of an openvpn tunnel?
00:04.53bmdbah
00:04.56bmdwrong channel
00:07.31drwelbyWhat would cause distortion/static on only for the caller on a SIP phone, and not any of the sound from the called party?
00:07.48tzangeris there any magic involved in getting asterisk manager interface to emit cdr events?
00:07.52tzangerI've got cdr_manager enabled
00:08.01tzangerI have manager enabled, and have a user with 'cdr' events
00:08.21tzangerI can log in, and I can get other events (call, log, etc.) if I enable them
00:08.24tzangerbut no cdr at end of call
00:10.45Bl0w_M0nkdoes anyone use liksys pap2t adapter?
00:11.30Bl0w_M0nkwrong channel
00:12.40codefreezetzanger: what events are cdr events?
00:12.53tzangerhttp://www.voip-info.org/wiki/view/Asterisk+cdr+manager
00:16.02*** join/#asterisk Braxus (n=bhsieh@207.47.21.58.static.nextweb.net)
00:16.41codefreezeok, tzanger, look at how asterisk started up... did the cdr_manager.so load up OK? Any messages?
00:17.34codefreezeAnd, when you "make menuselect", do you see the cdr_manager checked?
00:19.13*** join/#asterisk mvanbaak (i=michiel@vanbaak.xs4all.nl)
00:19.18codefreeze(it's usually #2, "Call Detail Recording"
00:20.57tzangercodefreeze: yes, it's all there
00:21.03tzangercdr status shows it loaded
00:21.18tzangerCDR registered backend: cdr_manager
00:21.52*** join/#asterisk Raky-2 (n=John@220.157.75.246)
00:22.11tzangerlooking at the source it doesn't have much in the config
00:22.16tzangerjust enabled and mappings if I so choose
00:23.27codefreezeNext step, run asterisk under gdb, and break in manager_event. Make a call and see what happens for "Cdr".
00:23.39*** join/#asterisk dlynes (n=dlynes@d154-20-9-152.bchsia.telus.net)
00:24.34tzangerI did the next best thing, edited the code to emit a log_warning
00:24.55codefreezeand?
00:25.05*** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net)
00:25.24tzanger[Oct 16 20:25:03] WARNING[6208]: cdr_manager.c:120 manager_log: ABK: manager_log, enablecdr=-1
00:25.27tzangerwtd
00:25.28tzangerer wtf
00:25.31tzangerthat'd do it
00:26.04tzanger[general] should have enabled
00:26.21tzanger[general]
00:26.21tzangerenabled = yes
00:26.25tzangerlooks good to me
00:26.27*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
00:27.45codefreeze-1 is non-zero, so it's enabled
00:28.15tzanger[Oct 16 20:27:45] WARNING[7966]: cdr_manager.c:84 load_config: ABK: enablecdr=-1, v->value is "yes"
00:30.07tzangercodefreeze: hmm is there magic to using gdb with asterisk
00:30.12tzangerI set a breakpoiunt to custom_log
00:30.24tzangerand it ran, I got my warning printed to the console but gdb did not break
00:30.40tzangerdo I need to thread apply all b custom_log or something?
00:30.55tzangeroh wait, wrong custom_log
00:32.43*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
00:34.23tzangerstill not breaking
00:35.39*** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-82d13e965a5e039b)
00:37.48tzangerasterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_manager.so: undefined symbol: ast_strftime
00:37.51tzangerinteresting
00:38.23*** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ee4f2ca9c72208f1)
00:38.41*** join/#asterisk rva (n=rafael@200-158-236-45.dsl.telesp.net.br)
00:39.05rvado i have to use any specific version of zaptel+asterisk to get a x100p working?
00:39.19Qwellrva: define "working"
00:39.39rvawhen i try to compile, i get "The Zaptel installation on this system appears to be broken."
00:39.46rvabut i just compiled zaptel 1.4.0
00:39.53Qwell1.4.0?  that's a bit...old
00:40.06hmmhesaysshould I update my poly's to the 2.2 sip firmware?
00:40.06Qwells/old/very old/
00:40.27tzangercodefreeze: any ideas?
00:40.30rvaok, i'll try a new one.
00:40.47rvabut someone told me that i have to use a specific zaptel version for this board
00:40.54rvamaybe the person was wrong
00:41.13Qwellrva: is it an actual x100p clone, or is it one of those cloned clones?
00:43.04tzangercodefreeze: looks like that strftime is what's killing me
00:43.14tzangerit's used everywhere though
00:44.42rvaQwell, i dont know. I got it from a friend. lspci shows Communication controller: Individual Computers - Jens Schoenfeld Intel 537
00:44.59*** join/#asterisk _pepo_ (n=Pepo@190.10.187.20)
00:47.20[TK]D-Fenderhmmhesays, yes
00:48.50*** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net)
00:49.07hmmhesaysI don't see that firmware anywhere
00:51.12[TK]D-Fenderhmmhesays, have to get it from your reseller
00:55.10*** join/#asterisk Visual_E (n=easy@unaffiliated/visuale/x-000000001)
00:56.42tzangerbah humbug
00:56.50tzangerthat's working onw, but still no cdr messages
00:56.52tzangeron AMI
01:00.34codefreezetzanger: so you cleared up your ast_strftime issue?
01:01.40tzangeryeah
01:01.40*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
01:01.40tzangerI figured the shit out
01:01.40tzangercdrs use 'call' or 'all' privileges
01:01.40tzangernot 'cdr' as the wiki suggests
01:02.42codefreezefor the sake of the next poor schmoe (not that you are a poor schmoe, but the NEXT guy is sure to be one!), if ya got a minute, you might update the wiki.
01:02.56tzangeryeah I am a schmoe
01:03.06tzangeractually I would like to submit a bug report and patch to give it 'cdr' privilege
01:03.21tzangersince if ALL you want is CDRs via AMI, it's a pain int he ass to have to parse out all the call crap too
01:04.06codefreezewell, you sorta already signed up for them with the cdr_manager.conf file; adding that to the ami config is sorta redundant, dontcha think?
01:04.57tzangereh?
01:05.07codefreezescratch that... I see your point now.
01:05.09tzangercdr_manager.conf enables cdr events altogether
01:05.16tzangerif you JUST want cdr events... you can't
01:05.23tzangerif anything, cdr_manager.conf is redundant
01:05.29tzangerif you don't want it, don't catch the 'cdr' events
01:06.06codefreezediff is this: if you don't want them, don't send them, vs. filter them out. Saves some cpu cycles to nix them early.
01:07.56tzangerfair enough, but why jsut cdr events and not call events in general?  There's more of the latter by far
01:08.07tzangernow I need to write a quick C interface for this, ugh
01:08.07codefreezetzanger: OK, file the request.... we can shove it into the trunk.
01:09.07*** join/#asterisk Swat2 (n=bler@218-215-192-45.people.net.au)
01:11.02Swat2Can you do a call forward unconditional on a ring group?
01:12.13Swat2I've got a bit of a unique situation where it's a home/office with 2 separate sets of lines going to 2 different ring-groups.. (Home (601) or Office (600))
01:12.38Swat2i need to be able to forward the calls differently when im out of the office
01:12.50*** part/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com)
01:14.05*** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net)
01:18.53*** join/#asterisk lokiloch (n=me@203.82.44.179)
01:19.37hmmhesaysit seems the same sip.ld for the ip 501 is not the same for the 320
01:19.52*** join/#asterisk osiris (n=osiris@c-71-205-35-230.hsd1.mi.comcast.net)
01:20.29*** join/#asterisk sharp (n=sharp@c-68-46-126-37.hsd1.pa.comcast.net)
01:27.53*** join/#asterisk circas (n=dom_paq@CPE0015e985d53c-CM0011aec7a4c6.cpe.net.cable.rogers.com)
01:28.33*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
01:35.35*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
01:38.27*** join/#asterisk gardo (n=gardo@121.97.251.62)
01:55.48*** join/#asterisk crudpuppy (n=someone@75.138.61.254)
01:56.43crudpuppySHould meetme show as a registered application under the cli when doing core show applications
01:56.44crudpuppy>
01:56.47crudpuppy?
01:57.09[TK]D-Fenderif you had zaptel installed before compiling *, then yes
01:57.17crudpuppyhmmm
01:57.23crudpuppyI did,  but I don't see it
01:57.36crudpuppyI'm useing ztdummy by the by
01:57.43crudpuppyno zap hardware
01:59.50CBU[^_^]M``hello
02:00.00[TK]D-FenderBEFORE
02:00.08CBU[^_^]M``anyone here uses SPA 3102 for the PSTN?
02:00.52[TK]D-FenderCBU[^_^]M``, Probably
02:01.38CBU[^_^]M``i a bit confused... do i need to add a trunk?? or set it as a device?
02:02.02JTadd a trunk?
02:02.10JTyou need to stop using freepbx
02:02.42CBU[^_^]M``what should i use?
02:02.43CBU[^_^]M``:)
02:02.58Maliutaasterisk
02:03.11Maliutaand vi to edit the config files
02:03.49[TK]D-Fenderuse WHATEVER for the config files as long as you're doing them yourself.
02:04.14[TK]D-Fendervi/emacs/nano/pico/mc/gedit/kwrite/whatever
02:04.27*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
02:05.48crudpuppyok,  I just recompiled * and it still don't seem to be compiling meetme???  what am I doing wrong here?
02:07.49Qwellhttp://it.slashdot.org/article.pl?sid=07/10/16/2334253
02:07.50Qwellbahahaha
02:11.13J4k3did cisco expect anything less from brazil?
02:11.20TrentCreekdid you rerun the make file?
02:12.47crudpuppyI did a make clean; ./configure; make;
02:13.05crudpuppybut it didnt compile meetme this time either
02:13.26crudpuppyI KNOW I've got zaptel in place cause I modprobed ztdummy
02:18.30*** join/#asterisk blq (n=Bl@dslb-088-064-146-088.pools.arcor-ip.net)
02:19.14ectospasmcrudpuppy:  did you try making sure meetme was selected under make menuselect?
02:19.37crudpuppyno,  I didnt know nothing about that
02:19.38crudpuppylol
02:20.13crudpuppyits got XXX next to it
02:20.15crudpuppy???
02:20.49ectospasmmeans you aren't installing some dependency
02:21.03ectospasmor some dependency for meetme isn't installed
02:21.41ectospasmif you hover over it with the cursor it may tell you what you need
02:21.53crudpuppysays zaptel(E)
02:22.12crudpuppywhich is what was stated earlier...but I compiled and installed zaptel
02:22.19ectospasmin what order?
02:22.35crudpuppyzaptel then asterisk then asterisk sounds
02:22.40ectospasmhrm...
02:24.45ectospasmcrudpuppy:  I'd suggest doing this:  "make clean && ./configure && make menuselect && make && make install"
02:24.50ectospasmand with that, to bed.
02:28.09crudpuppywas just looking through configure....trying ./configure --with-zaptel now
02:28.43crudpuppychecking for ZT_DIAL_OP_CANCEL in zaptel/zaptel.h... no
02:28.50crudpuppytheres my prob
02:28.55crudpuppywhere is it looking for that
02:28.56crudpuppy?
02:30.13crudpuppywiat a min
02:34.10*** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt)
02:37.30crudpuppyhow the heck
02:37.40crudpuppythere was a mistype int he makefile of zaptel
02:37.44crudpuppyso it wasnt finishing properly
02:37.44crudpuppylol
02:38.06[TK]D-Fendercrudpuppy, gott watch out for 1.4.5 ;)
02:39.45crudpuppymaybe this will fix it we will see in a bit
02:39.46crudpuppylol
02:42.26*** join/#asterisk _pepo_ (n=Pepo@190.10.187.20)
02:42.31_pepo_hi friends
02:44.07_pepo_How do I can get statistics about voicemail boxes? like space, message's number, etc, of each one
02:45.30crudpuppywoohoo,  it compiled meetme this time
02:46.54[TK]D-Fender_pepo_, "voicemail show users"
02:47.06*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
02:49.55_pepo_show user only give me the number of messages, How do I know the real space on disk
02:49.58_pepo_?
02:51.03[TK]D-Fender_pepo_, You'll have to write some custom stuff for that
02:51.58_pepo_How do I can? I can develop but I dont know where or how
02:52.39*** join/#asterisk PepOSX (n=pepOSX@190.72.151.134)
02:53.54_pepo_Do I have to use just scripts out of Asterisk in the gnu/linux filesystem?
02:54.03[TK]D-Fender_pepo_, /var/spool/asterisk/voicemail
02:54.11[TK]D-Fender_pepo_, Get to work.
02:54.20*** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net)
02:54.24[TK]D-Fender_pepo_, its all jsut files.
02:55.18*** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net)
02:56.16_pepo_yes
02:57.03*** join/#asterisk RageMax (n=RageMax@pool-72-77-120-39.pitbpa.east.verizon.net)
02:58.00*** join/#asterisk red9012 (n=marc3234@76-10-149-62.dsl.teksavvy.com)
02:58.15red9012ms just released ocs, how is this going to affect asterisk?
03:00.00*** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net)
03:01.12[TK]D-Fenderred9012, not in the slightest
03:01.19crudpuppyis there a way a internal user can call in from say a softphone to a conference?
03:01.38[TK]D-Fendercrudpuppy, Any call in * is jsut a call.
03:01.41crudpuppyI've got my conference room setup and working from outside
03:01.50[TK]D-Fendercrudpuppy, You can do whatever you want
03:01.54crudpuppy[TK]D-Fender,  what do they call on the softphone
03:02.05crudpuppythe conf room #?
03:02.09[TK]D-Fendercrudpuppy, its you dialplan... GIVE TEHM SOMETHING TO DIAL!
03:02.21crudpuppytrue
03:02.25crudpuppyI'm still new
03:02.30crudpuppybut this is nice so far
03:02.44[TK]D-Fenderexten => 666,1,Meetme(100) ; Yippy-kia-yay
03:02.52crudpuppyhehe
03:03.04crudpuppythanks
03:03.06[TK]D-Fendercrudpuppy, Now 666 = meetme room 100
03:03.12crudpuppyyeah I know
03:03.19crudpuppyI know a bit about reading that stuff now
03:03.39[TK]D-Fendercrudpuppy, Learning the dialplan it 100x more important than meetme
03:04.17crudpuppyall the functions and applications are way confusing I have like over 150 applications in here to learn about what they do!!!
03:04.59crudpuppyI just wanted to get meetme setup for now as that was my main goal at the moment for a conference call on thursday
03:04.59crudpuppyheeh
03:05.00red9012now that ms ocs. is asterisk going to be obsolete?
03:05.23crudpuppyred9012 thats like asking if linux is obsolete
03:05.24crudpuppyhehe
03:05.28[TK]D-Fenderred9012, Did you jsut hear my answer?
03:05.51red9012tkd-- your answer was once sentence, with no remark/explanation. Hence worthless.
03:06.18[TK]D-Fenderred9012, nobody is going to care about MS's solution in the big picture.  Nothing revolutionary.  * = CONTROL
03:06.53[TK]D-Fenderred9012, MS's solution just like every other proprietary vendor just ties people down.  * will thrive becuase its OPEN.
03:07.12[TK]D-Fenderred9012, Who wants M$ owning their ass for everything?
03:07.20[TK]D-Fenderred9012, why are YOU here?
03:07.34J4k3I think he's here to troll
03:07.45peanut-weird channel to troll..
03:07.53[TK]D-Fenderred9012, Thinking maybe "Hey I know... I should look at * so I can go buy a toaster from Avaya!"
03:07.56J4k3its like asking "now that microsoft released vista, why would anyone want linux or bsd?"
03:08.11peanut-anyone use the linksys WIP300 phone?
03:08.16red9012I am here to know if further investment of time is worthed knowing that now a new solution is available.
03:08.20[TK]D-Fenderpeanut-, BLEH <-
03:08.30peanut-oh?
03:08.35J4k3peanut-: I hear that wifi-equipped "pda phones" work best
03:08.37[TK]D-Fenderred9012, A new solution is ALWAYS available.
03:08.38J4k3or smartphones
03:08.51*** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net)
03:08.55peanut-well the WIP300 is $100..
03:08.57J4k3most dedicated voip phones lack CPU.
03:09.12crudpuppyisnt the wip300 tied to skype anyway?
03:09.15J4k3you can get an XV6700 for $150-175ish off ebay, mildly used.
03:09.16peanut-no
03:09.20peanut-that's the 320
03:09.20J4k3nah thats the 320 iirc.
03:09.23J4k3the 300 and 330 are sip
03:09.23crudpuppyoh
03:09.30J4k3the 330 is a wm5 phone, iirc.
03:09.32peanut-which is More expensive
03:09.32[TK]D-Fenderred9012, Linux will never make it on the desktop.  Your mom will never be able to use it.... blah blah blah.
03:09.47NivexMy mom runs Ubuntu, so :-P
03:09.53[TK]D-Fenderoh...
03:09.55J4k3my mom is insane.
03:09.56[TK]D-Fender</sarcasm>
03:09.59J4k3(and runs XP Pro)
03:10.22J4k3and I'm likely retarded because I'm running vista (and not hating it anymore)
03:10.35crudpuppyJ4k3,  same here
03:10.36crudpuppylol
03:10.39red9012the analogy with xp/linux may very well be not applicable in this case.
03:10.45crudpuppyI have to have it for some of my WORK apps
03:10.46J4k3red9012: it is, completely.
03:10.50NivexJ4k3: I think that's called masochism
03:11.00J4k3any feature that microsoft adds can be completely replicated in the OSS community
03:11.06J4k3and the price tag is a whole shitload better.
03:11.27crudpuppyare we really gonna argue Open source vs proprietary here?
03:11.34J4k3I'd hope not.
03:11.45crudpuppythat seems to be what red9012 is trying to do
03:12.32peanut-so what's wrong with the WIP300?
03:12.32[TK]D-Fenderred9012, Tell you what.. how about you extrapolate why the world will dump the awesome control of * at such a low cost to be shoved down M$'s myopic solution and getting taken to the cleaners doing it...
03:12.33J4k3my girlfriend just IM'd me and told me "you >>>>> *"... so I'm thinking what Microsoft and asterisk should be concerned about is *me*, not each other.
03:12.58J4k3but I think she just meant I was greater than everything, which most likely doesn't include any microsoft or digium products.
03:13.13crudpuppylmao
03:13.14[TK]D-Fenderred9012, Yup.. That IIS is gonna KILL Apache.  Firefox?  Nah, that'll NEVER work.
03:13.28J4k3peanut-: I'm not sure... I've never used one
03:13.34NivexJ4k3: awwwwwwwwwwwwww
03:13.46peanut-[TK]D-Fender seemed to have a stong opinion and few details
03:13.51J4k3I was tempted to try the 330 til I saw the price of mildly used wifi pdas
03:13.58J4k3well, this channel has a lot of those
03:14.00J4k3see this
03:14.01J4k3~gs
03:14.01jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
03:14.04J4k3~wifi
03:14.05jbotwifi is, like, see wireless or for a small compact non-port-blocking card, get one of these a) linksys wcf12 for only $65 shipped from buy.com b) netgear MA701NA for $65 shipped from buy.com c) socket LOW POWER wlan (amazing battery life) for $160 + shipping on buy.com, or better than nothing
03:14.09J4k3err thats not it
03:14.25[TK]D-Fender~wifisip
03:14.26jbotWi-Fi SIP phones suck.  All of them.  HARD.  Some only slightly less than others...
03:14.28J4k3yeah, thats it.
03:14.32[TK]D-Fender:D
03:14.33JTpeanut-: an opinion backed up by fact
03:14.34J4k3alas
03:14.37J4k3whats fucked up about this is
03:14.39NivexNokia N800 with BT headset?
03:14.58JTmobile wifi is unsuitable for reliably quality voice calls
03:15.04JTreliable
03:15.04*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
03:15.04J4k3I have a GRANDSUCK BT 101 behind a openwrt-hacked wgt634u router, across the house from a wrt54g v2 running openwrt, connected to my lan... and it runs beautifully well
03:15.12J4k3now, both F1000G's I had come through here were completely worthless
03:15.13L|NUXHello every one
03:15.17J4k3a lack of CPU was pretty evident in every respect.
03:15.40L|NUXcan some one help me with South Korea PRI
03:15.58L|NUXwhen i try to call out using PRI i got this error message
03:15.58L|NUX> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: Private network serving the local user (1)
03:15.58L|NUX>                  Ext: 1  Cause: Invalid number format (28), class = Normal Event (1) ]
03:16.00peanut-I need anothe wrt54g AP for my network...
03:16.12J4k3the new ones are crappy
03:16.14red9012for one thing, the open source you refer to are all much much bigger projects involving a whole lot more people than asterisk.
03:16.15J4k3really crappy
03:16.17peanut-one for 802.11 SIP and one for 802.11 computers..
03:16.18J4k3the antennas aren't even swappable
03:16.20peanut-different VLAN
03:16.21J4k3I suggest a wrt54gl
03:16.34JTL|NUX: then check with the provider for the correct number format, and send numbers correctly
03:16.34J4k3which is basically a wrt54g v4
03:16.38J4k3same hardware level
03:16.42J4k3same removable antennas
03:16.46JTred9012: this means shit why?
03:16.55L|NUXJT : their support pathetic :(
03:16.56[TK]D-Fenderred9012, True.... do you think MS is going to matter to Cisco's telecom business?  How about Nortel?  Avaya?
03:16.57JTred9012: you still haven't made any case at all for this ms crap
03:17.10[TK]D-Fenderred9012, We all love * and we are growing.  MS offers NOTHING for us.
03:17.10peanut-J4k3: what's the purpose of getting a gl?
03:17.16L|NUXJT : i was looking some one who already worked with south korea pri's
03:17.28J4k3peanut-: more ram, more flash, rp-tnc jacks instead of perminantly affixed antennas.
03:17.28JTL|NUX: i don't think you'll find that here
03:17.29[TK]D-Fenderred9012, Do you realize that *'s audience isn't MS'?
03:17.36L|NUX:(
03:17.36peanut-ah
03:17.40L|NUXJT: ok
03:17.43J4k3peanut-: the WRT54G V8 is a total POS
03:17.48peanut-why
03:17.52J4k3its like the worst $20 router you've ever seen, except its $49.95
03:18.26*** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net)
03:19.04J4k3the OS is quirky, the ram and flash are too small to load any linux-based OSes on it, the antennas aren't removable
03:19.25J4k3the only improvement I've seen from the original product back 5 years ago is they finally include a switching-type wall wart, instead of an ugly power-wasting transformer.
03:20.14JTwhen the switching ones are too crappy, they are worse as they can make very noisy power
03:20.27J4k3JT: this one seemed pretty decent.
03:20.44J4k3but its only 12V/0.5A.. typically for the transformer type that was too small to keep the unit happy
03:20.51J4k3and eventually things get unstable.
03:20.56JTas good as polycom are
03:21.04JTtheir switching PSUs are trash
03:21.20JTthey make awful power that renders a headset with amplifier inopperable
03:21.22*** join/#asterisk i3inary (i=i3inary@ip72-207-113-253.sd.sd.cox.net)
03:21.41J4k3ick
03:22.16J4k3for wifi stuff icky power can lead to all sorts of crappy operation
03:22.29J4k3from unstable ethernet to dirty RF transmit.
03:22.47peanut-icky power, sounds highly technical
03:22.57J4k3it is.
03:23.22peanut-next time I describe a high SWR on a line I'll refer to it as "icky"
03:23.38J4k3dirty input power isn't going to lead to a high swr.
03:23.56JTpeanut-: dirty power is a pretty basic concept.
03:24.01J4k3if you want a consultant you can pay for one, prick.
03:24.01JTit doesn't need a fancy name
03:24.19peanut-... it was a joke
03:24.30J4k3:P
03:24.32peanut-unknot those panties
03:24.40J4k3but thats how I prefer my thongs.
03:24.45J4k3knotty
03:25.14J4k3(I've had a shitty day for a lot of non-technical reasons.  sorry for being an ass)
03:25.43[TK]D-FenderJ4k3, Yeah... you need a good server crash to help give yourself some context :p
03:26.53neaxgood afternoon, ladies and gentlemen
03:27.07J4k3[TK]D-Fender: I heard a HD raising hell in the closet a few minutes ago
03:27.16J4k3that shrill sound that can only be made by an HD.
03:27.20J4k3:|
03:28.06*** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net)
03:28.36J4k3and theres the text message.
03:28.39J4k3verizon must be slow tonight.
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03:33.17hmmhesaysi've had a shitty day for technical reasons, now I'm drinking
03:33.57neaxi had a good day, but i'm still drinking
03:34.06neaxi just feel that my liver could do with the workout
03:34.25hmmhesaysyeah
03:34.29hmmhesaysdrinking is just good
03:35.02J4k3I'm waiting for everybody to go to bed
03:35.05J4k3then I'm getting out the bottle
03:35.13J4k3if I get intoxicated first, I might slap somebody I care about.
03:35.56J4k3I much prefer technical problems, ugh.
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04:23.08_pepo_is there some way to configure language using the CLI?  I've configured my sip.conf with language=es, I copy the sound in /var/lib/asterisk/sounds/es but still is in en
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05:21.00i3inaryanyone know much about voice trading or similar services?
05:21.18*** join/#asterisk Alowishus (n=jpenix@ip72-199-253-51.sd.sd.cox.net)
05:22.42AlowishusI've got a simple TDM400P with a single FXO card in port 4... ztcfg sees it fine (modules are loaded fine), but Asterisk insists "Unable to open channel 3: No such device"
05:23.16Alowishusmy zapata.conf is just 4 lines... signalling=fxs_ks and channel => 3
05:23.20Alowishuswhat am I missing?
05:23.25Strom_Muh
05:23.33Strom_Mthe FXO module is in port 4, not port 3
05:23.42Alowishusoh I thought it counted from 0
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05:23.44Strom_Mno
05:23.48Strom_M1
05:23.52Alowishussorry
05:23.57Alowishusbut ztcfg sees it as 3
05:24.07Strom_Mwhere did you get the idea that it's on port 3?
05:24.19Alowishusztcfg
05:24.26Strom_Mwhat does zaptel.conf look like?
05:24.40Alowishusfxssks=3
05:24.44Strom_Mno
05:24.47Strom_Mfxsks=4
05:24.50Alowishusdur
05:25.21Alowishusah tha'ts where I got it
05:25.31Alowishuswhen wctdm loads it lists Modules 0-3
05:26.17AlowishusMUCH better
05:26.19Alowishusthank you
05:26.24Alowishussanity check :)
05:27.55Strom_Myou're welcome
05:27.59Strom_Mnext time:  read the manual plz :)
05:28.33Alowishushey I'm working through the TFOT book... I just didn't adjust the examples correctly
05:29.00Strom_MTFOT != manual
05:29.03Alowishustrue
05:29.36*** join/#asterisk mLx (n=mlx@217.151.231.18)
05:29.56Alowishusactually now that you mention it... which manual would discuss this numering issue?  Man for the TDM400P itself?
05:30.07Strom_Myes
05:30.13mLxHi there. I have a problem withh call transfer. When I pressing # or * a hear only the beep.
05:30.34mLxDoes anybody can say anything about this trouble?
05:30.43Strom_MmLx: forget inband transfers and do them the right way
05:31.24mLxStrom_M, Can you say how can I fix it?
05:31.34Strom_MmLx: what kind of phone are you using
05:32.09mLxOn the first side I use X-Lite and on the second one using analog phone via FXS
05:32.24Strom_MX-lite should have a transfer feature
05:32.37Strom_Mthe FXS phone you transfer just like you'd do with Centrex service from the telco
05:33.34mLxYes. In X-Lite I see button "X-fer" but it's hint is "Upgrade to eyeBeam 1.5 Transfer feature :("
05:34.14mLxI know that asterisk can transfer calls by pressing # or other buttons. How ca I make this feature?
05:34.18Strom_Muse a softphone that doesnt suck
05:34.31Strom_MmLx: the # transfer is a really nasty knudge
05:34.33Strom_Mkludge
05:34.35Strom_Mdon't use it
05:35.23mLxI think that I will not have toubles with soft phones but in our company we usign basic analog phones without transfer features
05:35.47mLxI afraid that it will be a troublw
05:35.51Strom_MmLx: hence why I said "just like you'd transfer a call using Centrex service"
05:36.00Strom_Mi.e. hookflash, dial destination, hang uo
05:36.28mLxStrom_M: I'm sorry. Can you say more about Centrex service?
05:36.46Strom_MmLx: I just told you how to do it :)
05:37.18mLxI can't understanding what is that
05:37.25mLxhow can I use it?
05:37.37Strom_Myou hookflash
05:37.46Strom_Myou dial the destination for the transfer
05:37.52Strom_Mand you hang up once the destination starts ringing
05:37.56Strom_Mit's quite simple
05:38.12mLxAh! Sorry. I'm understand
05:38.46mLxBut this feature will work with attended transfer only. And I can't transfer call blindly
05:39.56Strom_Mif you need feature-rich telephone sets, you should use SIP phones instead of analog phones
05:40.37mLx:) Would be great, but I think that our managment will not aprove this idea
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05:40.52L|NUXhave any one used pri from TDX10A
05:40.55L|NUXswitch ?
05:41.12Strom_Mthis must be Terrible English Night in #asterisk
05:41.18mLxStrom_M, What soft phones will you recommend for use?
05:41.26*** join/#asterisk bintut (n=bintut@203.125.63.150)
05:41.32Strom_MmLx: "zoiper" is well-regarded
05:42.51mLxStrom_M, Thank you!
05:43.08mLxCan you say also where can I find "zoiper"?
05:43.37Strom_Msearch.yahoo.com
05:44.05mLxHehe :) Okay
05:44.18mLxThank you once more. Good buy
05:44.39Strom_Mwhat am I buying?
05:46.25*** part/#asterisk mLx (n=mlx@217.151.231.18)
05:47.17Strom_ML|NUX: what is your question re PRI?
05:48.54peanut-so does skype have an ANI associated with each user that has skypein?
05:49.36peanut-on outgoing calls
05:49.52*** join/#asterisk chendy (n=chendy@121.76.132.123)
05:50.50Strom_Mpeanut-: does this look like #skype to you?
05:54.45peanut-ah the ANI for all outogoing is 1-202-580-8200, just called an ANAC then googled the number..
05:55.05peanut-Strom_M: no, but it doesn't look like #shitheads either, you seem to be lost.
06:01.06*** part/#asterisk Alowishus (n=jpenix@ip72-199-253-51.sd.sd.cox.net)
06:02.06peanut-I didn't ask about CPN, that's generally 0123456789 or 0000000000, I asked about ANI
06:02.08*** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com)
06:03.10Strom_Mjust wanted to make sure you knew the difference; so many people don't have a clue what "ANI" really is
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06:23.38*** join/#asterisk MacDeath (n=davidn@hobbit.tsol.co.za)
06:23.43MacDeathmorning all
06:23.59MacDeathor evening to some
06:24.09MacDeathi wonder if someone can help me
06:24.17*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
06:24.23MacDeathi have recently reinstalled my * box
06:24.57MacDeathand my sound playback starts normal, but then gets slower and slower
06:25.06MacDeathwith a big stutter
06:25.08*** join/#asterisk WilliamK (n=noc@static-71-170-144-3.dllstx.fios.verizon.net)
06:25.53Strom_MMacDeath: what kind of equipment are you using
06:26.40MacDeaththe phones or the pc?
06:26.51Strom_Meverything
06:26.58MacDeathit does it on grandstreams and snoms
06:27.03MacDeaththe pc itself is new
06:27.15MacDeathits an Intel core 2 duo
06:27.23MacDeathon an intel motherboard
06:27.24Strom_Mand the phones?
06:27.31Strom_Many interface cards?
06:27.49MacDeathgrandstreams 102, grandstream GXP2000 and snom 300
06:27.53Strom_Many equipment between the phone and the pbx?
06:27.58MacDeathzaptel TDM400P
06:28.10MacDeathand a B410P
06:28.31MacDeathonly a switch between the phone and pbx
06:28.42MacDeathvoice calls are perfect
06:29.08MacDeathits only when the pbx "creates" the voice
06:29.29Strom_Mare you playing back the included sound files?
06:29.44*** join/#asterisk af_ (n=getsmart@81-174-44-210.dynamic.ngi.it)
06:29.53MacDeathyeah
06:29.56MacDeathas an example
06:30.10MacDeathif i dial *77 (for call waiting)
06:30.33Strom_M....*77?
06:30.36MacDeaththe first word "call" sounds ok
06:30.42Strom_Myou wrote that feature code yourself?
06:31.29MacDeathno, that came from trixbox
06:31.36MacDeathbut it happens with anything
06:31.48Strom_Mugh
06:31.51Strom_Myou're running trixbox?
06:31.52MacDeatheven when you call to give you the time
06:32.01Strom_M~tribox
06:32.08Strom_M~trixbox
06:32.09jbothmm... trixbox is a full linux distro that includes FreePBX, and other 3rd party add-ons. It is all this extra stuff which makes trixbox seriously painful to support and hence you will find little help here for it.  Try asking in #trixbox or on their forums & WIKI at http://www.trixbox.org
06:32.31MacDeathi am now, i installed trixbox to try get rid of the problem
06:32.59Strom_M*77 is supposed to be for activating anonymous call rejection
06:33.09Strom_Mone wonders how badly they've screwed up all the other VSC assignments
06:33.11MacDeathit happened before i installed trixbox though
06:33.34Strom_Mwell, if you go back to not trixbox, then I can try helping you
06:33.48MacDeathi can do that very easily
06:34.17MacDeaththe example i did before trix
06:34.21MacDeathwas dialing for the time
06:34.44MacDeathit starts announcing the time normally
06:34.51MacDeaththen gets slower and slower and slower
06:34.59Strom_Mwhen does asterisk include a time announcement number?
06:35.53MacDeathi've had that since i first installed asterisk which was nearly 2 years ago
06:36.10Strom_Mdid you write it yourself?
06:36.35*** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it)
06:37.31MacDeathno
06:37.46Strom_Mwhere did it come from then?
06:37.50MacDeathit was so long ago, i cannot even tell you
06:38.20MacDeathits hardly the point though. everything has been working until i reinstalled on new hardware
06:38.22Strom_Mcan you share the code with me?  i'm curious to see what it does
06:38.50*** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net)
06:40.16MacDeathmy question is though, can a network card cause the stuttering of the sound
06:40.23*** join/#asterisk felix_da_catz (n=felix@c-98-198-197-85.hsd1.tx.comcast.net)
06:40.29MacDeathor a problem with cpu / motherboard
06:40.42MacDeathas that is all i initially changed before the problem started
06:40.51Strom_MMacDeath: if you have some irq fighting going on, then possibly
06:41.08MacDeathhow would i know if there is?
06:42.21MacDeathif i cat /proc/interupts i get
06:42.25MacDeath<PROTECTED>
06:42.25MacDeath<PROTECTED>
06:43.08Strom_Mwhat happens if you take the cards out of the system and disable the drivers
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06:49.02MacDeathim going to try that now
06:52.20phixhey, I am having some call quality issues
06:52.38Strom_Mphix: please, be less specific
06:53.31phixok, A and B: A -> ~B, B->A
06:54.15Strom_Mi was being sarcastic
06:54.40phixPerson behind Asterisk box has poor quality / crackily noises while the person that they are ringing does not notice any quaility loss
06:54.54Strom_Mwhat kind of equipment are you using
06:55.00Strom_Mwhat kind of PSTN link?
06:55.16phixPerson behind asterisk box is using a TDM analog phone which goes out Internet using SIP
06:55.26phixhmmm
06:55.28phixTDM card even
06:55.31Strom_Mwhat, pray tell, is a "TDM Analog Phone"
06:55.32Strom_Mok
06:55.42phixand an analog phone connected to it
06:55.48Strom_Mwhat does your internet link look like?
06:56.16Strom_Mare you sharing the internet link with anything else?
06:56.28phix1500/256 connected by ADSL router modem, Asterisk box is behind NAT
06:56.47Strom_Mdo you get crackly calls when calling between two phones behind the asterisk box?
06:56.57phixone line
06:57.08Strom_Mwhat does "one line" mean
06:58.19phixI cannot call a phone behind the asterisk box because the TDM card only has two modules, one for PSTN and the other for internal phone
06:58.25Strom_Mok
06:58.39Strom_Mdo you get crackling when using the POTS circuit?
06:58.59kiscokidyou could configure and call a softphone
06:58.59phixGood question
06:59.16phixkiscokid: I am not at the site though, it is a good 40 min drive
06:59.34Strom_Mphix: how the hell do you expect to diagnose the issue if you're not in front of the machine
06:59.52phixStrom_M: via proxy
07:00.00Strom_M"via proxy"?
07:00.14phixI will call the person at the site :)
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07:00.29peanut-I hate being that person onsite..
07:00.44Strom_Mthat sounds like chapter 5 of the Big Book of Brain-Dead Debugging
07:01.02phixStrom_M: not really, I have SSH access to the computer
07:01.11phixI can chagne settings and see if they notice any change
07:01.27Strom_Mis there someone on-site now?
07:01.54phixsure it will add on some latency, but any way my debugging methods isn't really want I want to talk about, I want to talk about how to diagnose and resolve this call quality problem
07:01.58phixStrom_M: yes
07:02.16Strom_Mphix: ok, so try a call out the POTS line and see if that's also broken
07:02.37phixok
07:03.26phixjust need to find out what number I assigned to ring through POTS
07:03.35Strom_M?
07:03.44Strom_Mplace an outbound call from the TDM phoe
07:03.46Strom_Mer
07:03.49Strom_Mfrom the analog phone
07:03.56phixyeah, it will go out through SIP
07:03.59Strom_Mdamn you and your mangled terminology
07:04.04phixlol
07:04.04phix:P
07:04.36Strom_Mjust reroute the outbound call to use the pots line
07:04.40phixoh come on, TDM phone, an analog phone connected to a TDM card, how hard is that?
07:04.46Strom_Mbecause that's not what TDM is
07:04.54*** part/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it)
07:05.21i3inarystrom maybe for your birthday ill make you an if this then do this flowchart for all the  most common asked issues you deal with ...and host it in html for you.  then you can just link urls.
07:05.36phixhmmm 28ms ping to SIP provider
07:06.04kiscokidwho is the sip provider?
07:06.05Strom_Mi3inary: um, ok?
07:06.06phixi3inary: good idea ;)
07:06.09phixkiscokid: GoTalk
07:06.22phix(AU)
07:06.33Strom_Mphix: ffs, can you just try one thing at a time when I ask you to, please?
07:06.41phixStrom_M: yes
07:07.47peanut-wow, I just realized I can setup a soft phone to test asterisk instead of waiting for my hard one....
07:08.28kiscokidd'oh
07:09.06peanut-what's one that doesn't suck hard on linux
07:10.22phixStrom_M: ok I just rang them on landline number, they didn't have a quality problem at all
07:10.28phixonly when calling via SIP
07:10.40phixor making a call via SIP
07:10.49kiscokiddropped packets?
07:11.32Strom_Mphix: what happens if you try a different ITSP?
07:11.36i3inarydid you make a call from your house to that sip provider and rule out the sites connection?
07:11.37phixhmmmm, this could be it, load average is 1.97
07:12.02phixi3inary: I called from my land line to their land line (which the asterisk box is connected to)
07:12.39phixhmm also, another problem is when I call via landline there is a redirect message that is played, the end bit of it gets cut off
07:13.10i3inarysame everytime? or just one time?
07:13.35phixsame everytime, I notice alot of the audio messages are cut off short
07:13.51phixhold message, voice mail etc
07:16.51i3inarywell this doesnt seem like a busy site right?
07:17.00phixok so ping from asterisk box to SIP provider is 28ms, and link speed is 1500/256, what else would influence bad SIP phone quality?
07:17.19phixi3inary: no it isn't, still annoying that it cuts off though
07:17.50phixThe computer is a cele 850MHz, it doesn't really do much, current CPU usage is about 5%
07:17.51SparFuxcodec
07:17.56phixg729
07:18.05SparFuxok its good
07:18.26i3inarythis just started? or has it been like this since you installed?
07:18.29phixthe quality was even worste on ulaw actually. don't know if that is significant since ulaw is a bandwidth whore any way
07:18.59DRTHMany transocding?
07:19.02i3inaryif it was worse on ulaw then i would guess your having a bandwidth issue
07:19.12i3inaryis that used for other things or just asterisk?
07:19.17phixi3inary: Recently it has been bad, it has been on and off though
07:19.22i3inaryhow many people on site?
07:19.34i3inary1 person with a torrent is all it takes
07:19.52i3inaryi screw my self all the time with a torrent up while im on the phone
07:19.55phixi3inary: 2 people, no torrents, little traffic (HTTP and SMTP mostly)
07:20.21phixI do have a shaper / prioritiser  running (well I think it is running, I will check :))
07:20.25peanut-oh awesome, the box I was gonna use likes to turn itself off randomly..
07:20.33i3inaryyou sure they dont have any active virus....have you checked your router logs for the inbound/outbound traffic?
07:20.37peanut-that won't make a good asterisk box..
07:20.58DRTHMphix: why dont you get a pcap capture of a call and check if rtp packets are missing/coming out of sequence?
07:21.13phixqdisc sfq 10: dev eth1 parent 1:10 limit 128p quantum 1514b perturb 10sec
07:21.14phixhmmm
07:21.20phixpeanut-: lol
07:21.26phixpeanut-: VIA or SIS chipset?
07:21.33peanut-no idea
07:21.39i3inaryor even better if you could capture after the router
07:21.40peanut-it's my old windows gaming machine..
07:21.41phixpeanut-: or an old prescott CPU?
07:21.46peanut-amd64
07:22.11phixi3inary: I don't have any bandwidth monitoring software running at all, perhaps I should run some
07:22.19peanut-it actually powered off.. maybe it's just a shitty PSU
07:22.23phixhmm actually snort may be on there
07:22.23peanut-that's generally the first thing to go
07:22.33phixnah snort is not on there
07:22.45i3inaryi have also had my soho router take a shit many times and ijust have to cycle it...even with the latest linksys firmware
07:23.13phixpeanut-: true, get a 660Watt Antec :) weighs a ton and requires the same weight in money to buy it
07:23.16peanut-i3inary: stop downloading so much porn on bittorrent..
07:23.29phixheh
07:23.44i3inaryyeah...sabayon is 3d porn...ill give you that
07:23.55i3inary3d os pr0n
07:24.08phixok so any way, any more suggestions about both of my problem? (crappy SIP quality and audio playback messages cut out)
07:24.14phixStrom_M: ?
07:24.17phixi3inary: ?
07:24.27i3inaryisolate the issue to either asterisk or bandwidth
07:24.51i3inaryim leaning towards network bandwidth unless there is a pattern to the cut offs
07:25.01*** join/#asterisk remmo (n=junk@203.32.47.250)
07:25.27i3inaryonly time i ever had jitter was due to bandwidth or serialization delay
07:26.04i3inaryif the site is that small unplug all but asterisk and make a call
07:27.22i3inaryif it is still bad i would powercycle router and reload the asterisk process
07:27.50*** part/#asterisk kiscokid (n=Ron_Laut@adsl-216-101-109-187.dsl.snfc21.pacbell.net)
07:28.10i3inarydid you say if this was a persistant issue or just came up?
07:28.50i3inarymy advice would pertain to a new issue rather than an issue there since install
07:30.24i3inaryyou said your in AU?
07:31.13*** join/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
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07:50.28i3inarywow did i loose connection or something
07:51.44J4k3nah, its just really boring
07:51.45J4k3haha
07:52.18i3inaryok well...bored me is gonna go punch the sandman in the throat
07:53.51*** join/#asterisk parag0n (n=parag0n@87-194-9-117.bethere.co.uk)
08:03.39*** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com)
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08:17.59nexiluswhat squiggly do i use in the dialplan to know "when the caller has hang up" ?
08:18.32peanut-Oct 16 22:07:59 WARNING[939]: translate.c:133 ast_translator_build_path: No translator path from gsm to unknown           <- is that likely a configuration issue or an install issue? it's brand new fbsd 6.2 and built from ports
08:18.52peanut-happens when a test call from soft phone dialed itsel
08:18.54nexiluscause i'd want an agi to run on the asterisk once a certain someone hangs up (prior,or post actually talking to someone)
08:18.57peanut-*itself
08:20.22mildknexilus: use the h priority and DeadAGI
08:20.39*** join/#asterisk bantu (n=Miranda@p54A32738.dip0.t-ipconnect.de)
08:22.40nexilusthanx mildk
08:23.35peanut-oh, my version is horribly old, I didn't upgrade ports first..
08:30.13peanut-what's the difference between the 1.2 and 1.4 lines?
08:31.35*** join/#asterisk LT (n=lt@unaffiliated/lt)
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08:46.29nexilusIs there a way to catch when a user is "out of" a que and actually talking to an agent?
08:47.03phixhey
08:47.05phixso any more ideas?
08:57.24billybongowhat's the right way to connect * to a mysql cluster for realtime etc?
08:58.16billybongois there such a thing as SRV records for dsn?
08:58.50JTdsn?
08:59.12JTphix: what codec were you trying to use?
08:59.23phixg729
08:59.46JTi meant peanut-
09:00.35billybongojt, yeah dsn
09:00.43phixJT: ok, you could ask me a question though :) regarding my two issues
09:01.02JTbillybongo: what are you talking about?
09:01.41JTphix: i have no idea what your issues are
09:01.41phixJT: you want me to re-paste?
09:02.12JTpaste?
09:02.30phixre-paste my question
09:02.38billybongodsn = database source name
09:02.54JTif it's a pretty concise question you should be able to retype :)
09:03.00billybongois there a way to refer to your mysql cluster in a redundant-type way?
09:03.12JTbillybongo: well wouldn't that depend on the sql client?
09:03.13billybongoi.e. without having to specify a single IP number
09:03.23billybongoyes, I'm talking about the one in asterisk
09:03.29JTyou can use a normal dns record definitely
09:03.41billybongocan you use a SRV record?
09:03.56JTdoubt it
09:04.08JTdepends if odbc supports it i guess
09:05.11billybongoI would have thought this would be a common question, but I can't find it anywhere
09:05.15phixJT: ok question 1) I have a good ping to SIP provider but I am gettign crackling interference on the line, when I call the asterisk box via PSTN it works fine. 2) When audio is played back in a call, busy message etc.., it cuts out before finishing the message.
09:05.32JTbillybongo: like redundant sql servers is really super common in this area
09:06.26JTphix: how many sip providers have you tried?
09:06.57billybongoJT ok, not the _most_ common configuration, but not completely off the wall either
09:07.28phixJT: I only have one to test out, although the other providers that have been used in the past didn't have this issue, they had other issues.
09:07.53JTbillybongo: it's pretty non-standard, as far as asterisk goes
09:08.03JTphix: so try another
09:08.26phixJT: :/ so you reckon it is the SIP provider? not the Internet or any thing ?
09:08.45JTwho knows, you need to cancel possibilities out one by one
09:09.27phixok well changing SIP providers isn't an option atm, unless you have a AU one I could use :)
09:09.30phixto test
09:09.36JTpennytel
09:09.45*** join/#asterisk yassaccan (n=yassacca@admin189.hgo.se)
09:10.03billybongoJT: that surprises me - plenty of people using realtime - many of those must have at least thought about redundancy at some point
09:10.17badcfeanyone could help me with a dtmf receipt problem i have here.  i have made certain observaitions of my problem and could pastebin the small snippets demonstrating the situation.   ..?
09:10.34JTbillybongo: and out of the plenty, hardly any are close to being in the provider game
09:11.00JTand you can use linux-ha for sql anyway
09:11.13JTobviously an existing connection will be terminated
09:11.16billybongoyeah, that's a bit hacky if you ask me
09:11.23JTbut when they retry they'll be ok
09:11.31JTmost sql replication schemes are hacky
09:11.41JTand fairly seemless network connectivity isn't hacky
09:11.44*** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net)
09:12.08badcfemy problem is that i see that asterisk gets the dtmf, but neither the read app nor the waitexten seems very interested in it.  any hints?
09:12.13billybongoquite often when a box goes down it can hold onto its IP number
09:12.25billybongonot much use if you want otherbox to take it over
09:13.32JTbillybongo: how would it hold on if it's down?
09:14.19billybongowell, if it can be taken down by removing the power then it clearly will relinquish the IP number
09:14.44billybongohowever if it kernel panics or similar then it can hold onto it
09:14.52billybongoand requires human intervention
09:15.18billybongowhich is fine if the admin is awake :-)
09:15.26JTa proper setup requires no human intervention
09:15.35JTonly a poorly designed ha setup requires intervention
09:16.09JTif the kernel panics and is not responding, you can presume the ip
09:16.18JTyou can also put a load balancer in front
09:16.29JTand forcefully bring down network links with managed switches
09:17.23billybongonetwork cards will often keep responding despite a kernel panic
09:17.44JTand the supervisor will realise it's broken and disconnect it
09:17.56billybongowhich supervisor?
09:18.19JTif you have heartbeat running elsewhere
09:18.26JTit can see that the machine has failed
09:18.38billybongook, so something will attempt to connect to a service on the machine, fail and then pull its plug?
09:18.38JTand take action to fence the failed node
09:18.45JTyes, fencing
09:18.59billybongothat sounds a bit scary
09:19.02JTyou have a heartbeat daemon running on the server
09:19.09JThow is it scary?
09:19.16JTit's done all the time in enterprise networks
09:19.21billybongoyes, I know
09:19.29billybongodoesn't stop it sounding scary
09:19.37JTi see
09:19.50billybongoput it this way
09:20.00*** join/#asterisk psk (n=psk@golia.caltanet.it)
09:20.23billybongoa failover built into dns sounds less radical, and more smooth, than a failover requiring some monitoring software to implement isolation via hardware
09:20.46JTdns failover is also less reliable
09:20.47JTand slow
09:20.56billybongohow so?
09:21.11JTlinux-ha failover times are measured in millisconds
09:21.42JTcomputers cache dns results
09:21.42badcfehmm i found some pretty interesting stuff here on asterisk 1.4.2
09:21.44nexilusIs there a way to catch when a user is "out of" a que and actually talking to an agent?
09:21.54billybongoSRV records can be cached
09:21.57billybongoand should be
09:22.14JTnexilus: queue?
09:22.20nexilusyeah
09:22.21billybongoI can't see how bringing up another node can be quicker than the client deciding for itself to use a secondary servery
09:22.31nexilusqueue... and a few extra ueueueueue's to go :)
09:22.32badcfeits that when the dialed number contains an # then even if you goto like hell and the # is long ago, your read and waitexten wont work.
09:22.33JTbadcfe: a super old version
09:22.36billybongoserver even :-)
09:22.57JTbillybongo: the other node is already up, traffic is simply rerouted to it
09:23.13JTif you can't see how, you must not be thinking hard ;)
09:23.21JTdns srv is good for loosly coupled servers
09:23.21badcfeJT: when i dial asterisk extention 1#2 and then goto some test,1,1 then neither 1,1,waitexten nor 1,1,read will work
09:23.29billybongobut that's precise how SRV records work
09:23.35JTlinux-ha is good for servers that are on the same lan
09:23.37billybongoboth nodes are up all the time
09:23.47Strom_Mbadcfe: what kind of idiot puts # in the middle of an extension number?
09:23.51billybongowith ha you have to (at the very least) bring up the second IP number
09:23.59Strom_M# means "I'm finished dialing so put my call through now thanks!"
09:24.01JTand we can argue this till the cows come home, but what sql supports dns src?
09:24.08badcfeJT: but when the dialed exten was e.f 12 to start with (whatever that doesnt contain an #, then all works from there)
09:24.24billybongoJT, sure - hence my intiial question
09:24.27JTbadcfe: correct operation
09:25.02JTbillybongo: bringing up second ips or repartitioning the network is much faster than a client end timeout
09:25.17badcfeStrom_M: yes.  but the # is part of the sip peer, and i dial it like that.  the read or waitexten is going on in a context with no # hanging around.
09:25.26JTif you use a load balancer, that's the fastest yet
09:25.45billybongoJT not when you factor in that some clients are already connected and will need to time out
09:25.45JTwhy would the sip peer have #?
09:25.56badcfeokay then
09:25.56Strom_Mbadcfe: that still sounds like someone clueless set things up
09:26.15JTbillybongo: umm they will be redirected immediately on the server and you can reset the tcp connection
09:26.39badcfeanother question:  the codec negotiation is done separately on each channel and then theyre pridged (possibly not matching) is this right?
09:26.43JTbillybongo: your logic seems lacking
09:27.00JTthere is NO WAY that an average client app fails over faster with DNS SRV than linux-ha
09:28.23billybongoJT isn't that down to the client?
09:28.52billybongoif I configure the client to failover to the second server within 10ms of not finding the first, then I don't see how anything can be faster
09:30.12*** join/#asterisk cypherdelic (n=cypherde@p5B27CA3F.dip.t-dialin.net)
09:30.23cypherdelicpeople claim that im too quite on the fon
09:30.23cypherdelic<PROTECTED>
09:30.25nexilushmmm, so nobody knows how to know when a call from a queue reaches the final destination (the conversation with an agent), or do i have to do this by adding an AGI to the agents extension?
09:30.42nexilus...but will that have the same uniqueid as the call in the que?
09:32.02badcfeStrom_M: is there some way of making asterisk _not_ treat # like this.  just taking it as another character 0-9 ?
09:32.15billybongoJT, if it wasn't fast then it would be useless in telephony
09:32.25billybongowhich is primarily where it is used
09:32.50Strom_Mbadcfe: # is universally the "End of input" digit
09:32.52badcfeStrom_M: cause im setting up a ivr where the user will often do the navigation with the # and continuing navigating after that point still
09:33.15Strom_Mbadcfe: don't make broken decisions and then kludge your way around them
09:34.06badcfeStrom_M: its a top-down decission for the navigation that sometinges a # should be entering and then the navigation in the contexts should continue
09:34.20badcfeStrom_M: top-down for the system im trying to make i mean
09:34.22Strom_Mbadcfe: which idiot made that decision?
09:34.41Strom_Mi would guess someone who has no clue about telephony
09:35.14badcfeStrom_M: well, its another working system that we are using as template
09:35.27badcfeStrom_M: maybe i dont understand correctly.  making shure ..
09:35.43badcfeStrom_M: the read application uses # as term right?
09:35.47Strom_Mperhaps I should make it fairly clear:
09:36.02badcfeStrom_M: what if i do more that one read in your context.  will it work or not?
09:36.03Strom_Mnever never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never use # in the middle of an input string
09:36.19Strom_Mnever never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never never
09:37.02billybongonot ever?
09:37.02badcfeStrom_M: the # wasnt in the input string, it was only in the original called number.  so all # was done when entering the first context in the dialplan
09:37.24Strom_Mbadcfe: your design is broken.  i don't think i can make you see that.
09:37.32Strom_Mhave fun with all that
09:38.18badcfeStrom_M: is this a sensitive point for you
09:38.57JTbadcfe: you just can't accept no for an answer
09:39.00Strom_Mi'm always irritable when people who have no clue what they're doing barge right into telephony without bothering to find out how things work first
09:39.36badcfeif im not trying to find out ..
09:39.55badcfebut you _can_ call the read application twice, right?
09:40.00J4k3woo, I just got a P4-2.66/533 for my asterisk box.
09:40.19phixJ4k3: nice
09:40.31J4k3(or well, I found most of the parts to a P4/2.66 in my entry hall... and then found more various pieces to put it together... and now it works)
09:40.41JTbillybongo: and what magical client is this?
09:40.43*** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62)
09:40.43J4k3it'll be much nicer than the P3-700 its barely running on now ;)
09:40.51JTbillybongo: DNS SRV is primary telephony? news to me
09:41.09billybongodo you know what an SRV record is?
09:41.23J4k3steve ray vaughn?
09:41.31billybongoyeah, that's the one
09:41.37J4k3er stevie ray vaughn
09:42.04JTbillybongo: yes
09:42.10JTbillybongo: and your point is?
09:42.14billybongosip clients are almost universally supporting it
09:42.19JTwow sip
09:42.21JTomg bbq!
09:42.32JTsip != telephony in general
09:42.34billybongosip is a form of telephony
09:42.37billybongoI didn't say it was
09:42.42JTa dodgy form, but yes
09:42.48billybongoI said that telephony is one of the primary uses of SRV records
09:42.49JTtelco grade stuff doesn't tend to use sip
09:43.04JTmaybe sip is one of the primary uses of SRV records
09:43.19billybongowhatever
09:43.27JTit's fact
09:43.30JTnot "whatever"
09:44.02billybongosince sip is a form of telephony telephony is a primary use of SRV also, but I see no reason to get into splitting hairs
09:44.36JTtelephony primarily does not use sip
09:44.49J4k3sip is pretty godawful.
09:44.59JTit is better than IAX2 however
09:45.28badcfethe read app doesnt work.  and the _only_ # present wherever is the one i type after the digits
09:45.37badcfesee this small paste:  http://pastebin.ca/739589
09:45.51billybongosip is going to kill conventional telephony, but hey, let's put our heads in the sand
09:45.59J4k3JT: I was of the understanding that IAX2 "trunking" worked fairly well if both ends had precision timing devices available
09:46.03JThaha you're insane
09:46.15JTsip is never going to kill conventional telephone networks
09:46.18badcfejust take a look at http://pastebin.ca/739589 as its pretty small and scaring.  asterisk version is 1.4.2
09:46.20JTsip lacks way too many features
09:46.23JTand is inefficient
09:46.49phixJT: is IAX better?
09:46.52billybongoonce people have sip on their mobiles and decent wifi coverage you can wave goodbye
09:46.55JTphix: no
09:47.01J4k3"decent wifi coverage"
09:47.02JThahahahaha
09:47.02phixJT: Hsomething or other?
09:47.03J4k3BAHAHAHHAHAHAHAHA
09:47.04JTha
09:47.04J4k3omfg
09:47.05JTha
09:47.07J4k3hahahaha
09:47.29J4k3dammit, I told you to move your product ONLY in #trixbox
09:47.30J4k3hehe
09:47.37JTsplittle the seams with laughter here
09:47.50phixJT: why?
09:48.04JTwifi is not even close to telco grade
09:48.11phixok
09:48.13JTphix: because it's not a standard, and it does not scale
09:48.16billybongoand who cares about that?
09:48.17JT(iax)
09:48.24phixok
09:48.29JTpeople who want their phones to actually work
09:48.29phixJT: what is the best then?
09:48.30bagpuss_thecatbillybongo: sip is crap on broadband and home networks, how on earth do you propose getting it working on random wifi networks?
09:48.40billybongoit does work
09:48.41phixor better even
09:48.47JTphix: H.323 is one of the best voip protocols from a telco standpoint
09:48.54JTbut all voip is meh compared to circuit switched
09:48.55billybongoI was using my e65 on "The Cloud" at a cafe the other day
09:48.57J4k3billybongo: not in a way people are willing to use it.
09:48.58phixok, that was the one i was thinking of :)
09:49.13phixJT: although isn't H.323 properity?
09:49.24J4k3billybongo: 802.11 can't even roam quickly enough to keep oldschool GSM users from going "holy shit this is awful"
09:49.28JTphix: no, it's an ITU-T standard that was around well before SIP
09:49.29J4k3and GSM handoffs are pretty damned awful.
09:49.35JTand H.323 is much better designed
09:49.54billybongoyes, gsm is a great example, it sucks big time, but people love it
09:50.05JTbillybongo: then you left "the cloud" cafe, hopped into your car, drove 30 miles, and the call kept working?
09:50.09JTgsm may suck
09:50.16J4k3but GSM works a LOT better than sip-over-wifi
09:50.18billybongoof course it didn't keep working
09:50.19JTbut it's 1000 times better than wifi at transporting voice calls
09:51.03billybongothe company I work for sells mobile phones - sip is now a feature that people ask for
09:51.14JTbillybongo: as an extra
09:51.23JTbillybongo: not to replace a proper mobile phone network
09:51.39J4k3and it'd appear AT&T is seriously lacking any UMTS networks outside the most overpopulated crapholes of north america.
09:51.40billybongoin the office they use their mobile as a sip phone, when out they use gsm
09:51.52JTcongrats
09:52.18billybongothankfully AT+T doesn't even make it here
09:52.19J4k3billybongo: wow...  I just use Alltel and add the PBX's # to the "free" list.
09:52.21JTand this relates back to the obliteration of telco voice networks how?
09:52.30J4k3yay for Alltel "My Circle"
09:52.41J4k3yay for Alltel letting me use Alltel, Verizon and Sprint EVDO :)
09:53.04J4k3(Verizon only lets me use verizon, sprint only lets me use sprint and alltel, alltel gives me all three...)
09:54.24billybongoJT, maybe where you are things are different but where I am voip, and SIP in particular is pushing out trad telephony. No-one knows how long it will take, but eventually there is bound to be convergence onto a single (IP) nework.
09:54.37JTone of these days i need to write a guide for nubs as to why telco networks are here to stay and why they're more reliable
09:54.53JTbillybongo: sounds like a marketing wanker's pipe dream
09:55.00JTsure a lot of people will adopt voip
09:55.08badcfei see waitexten get called.  then i see the dtmf is received, and then waitexten times out.  this is the order of the events.  i type 2 and the exten => 2,1 is there, but asterisk does not put me there but to the timeout t exten.  seems to me like an issue -- probably with myself, but anyone has a hint here?
09:55.11JThowever existing phone networks are here to stay
09:55.21J4k3VoIP will give "POTS" some competition
09:55.26J4k3the cellcos have absolutely nothing to worry about
09:55.27JTheh
09:55.50JTeveryone using VoIPoI will be chaos
09:56.03JTas if you can guarantee voip calls over the Internet
09:56.05bagpuss_thecatbut the network provider gives absolutely appaling upstream
09:56.07JTto work properly
09:56.07J4k3even the most badass 900 mhz WISP gear can't hold a candle to GSM or CDMA when moving.
09:56.07bagpuss_thecatappalling
09:56.33J4k3JT: I could bet against my POTS line working in the morning
09:56.39J4k3and run a fairly decent chance of winning that bet
09:56.42JTJ4k3: hehe
09:56.46J4k3I wouldn't call POTS 'reliable' in most places
09:56.54JTwell
09:56.59JTit generally works
09:57.00J4k3its growing less so by the day
09:57.04JTquality can suck
09:57.11billybongoit's all down to the level of support you get from your provider, whether pots or voip
09:57.17J4k3due to a total lack of any telco giving a shit anymore, and no lawmaker from giving a shit about what the telcos do
09:57.37J4k3of course, I'm an odd case
09:57.48JTcircuit switched digital calls is where it's at reliability wise
09:57.50J4k3I have a real data circuit with a real SLA delivered to my home.
09:58.09cypherdelicHow can i record calls? I have set trwW and wW as far as i enabled recording on demand for that ext. but still *1 doesnt work. please help
09:58.17J4k3if I was say, one of the poor bastards that got suckered into a 2 year contract for ADSL with the local telco here, I'd be totally shit out of luck to ever make a SIP call
09:58.24*** join/#asterisk _ys (i=ys@91.151.196.254)
09:58.25J4k3their network is too lossy and jittery
09:58.36J4k3to make webpages and ssh work nicely, much less voip
10:00.08J4k3JT: yes and no...  for example there was a fiber cut in north houston... my ADSL never wiggled yet I couldn't make any LD calls (or any inter-telco calls) for almost a whole day.
10:00.33J4k3hell, even my T1s died... the ATM-based ADSL survived, just slower and re-routed through another town
10:01.00JTsure
10:01.09J4k3but this is very atypical of a well configured network
10:01.11JTbut on laws of averages, and places not in the sticks.. :)
10:01.14J4k3I just don't run into any of those anymore.
10:01.56J4k3JT: I dunno, I grew up in the 4th largest city in the USA...
10:02.12J4k3dialtone served off a 1A, iirc.
10:02.16cypherdelicbumstown idaho??
10:02.27J4k3cypherdelic: Houston, Texas... ever heard of it?
10:02.33J4k3its got about 8 million in the metro area now...
10:02.38JTget it right, it's "bumfuck, idaho" :)
10:02.41cypherdelicyes the state of legal murderes, right?
10:02.53J4k3cypherdelic: the whole USA is the state of legal murder, see Iraq.
10:03.46cypherdelicJ4k3: i only say: Bush@Whitehouse& sudo falseflag --target $HOME --match 911 && dd if/dev/america of/dev/world && sudo killall moslems
10:03.48*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
10:04.08J4k3cypherdelic: bush is a religious zealot, he thinks he's doing god's work.
10:04.16cypherdelicJ4k3: i only say: Bush@Whitehouse$ sudo falseflag --target $HOME --match 911 && dd if=/dev/america of=/dev/world && sudo killall moslems
10:04.36cypherdelicJ4k3: i know to much about usa as far as too much about europe
10:04.51JTcypherdelic: what are you talking about?
10:04.58JTi can't decipher your "command"
10:05.01cypherdelicpolitical matters
10:05.08JTobviously
10:05.13JTbe more specific
10:05.41cypherdelici mean highly decorated politicans will do more falseflag operations to claim on terror
10:05.57JTah ok
10:06.09J4k3shock and awe
10:06.09cypherdelicto have a nice reason controlling the world
10:06.46J4k3you're dealing with a few weirdo christian zealots being puppeteered by the richest men in the world.
10:07.07cypherdelicBillie?
10:07.09J4k3bush = religious weirdo
10:07.17J4k3cheney = all about his money
10:07.44cypherdelicyes i mean acting in name of jesus in non different from acting in name of mohammad
10:07.50J4k3exactly
10:08.05*** join/#asterisk appelza (n=d@dsl-240-153-182.telkomadsl.co.za)
10:08.09billybongowell I'm glad we all agree on politics
10:08.20appelzais there some sort of embedded version of asterisk, thats stripped and very small?
10:08.28billybongoobviuosly less contentious than telephony
10:08.45cypherdelicbe glad not to live in germany, i guess in some years linux is a hacker tool here too and going to be illegal
10:09.11cypherdelicbecause any piece of software that CAN have the purpose to illegal computer crimes
10:09.15J4k3heh, only if Microsoft ends up owning your government
10:09.15cypherdelicis illegal
10:09.25cypherdelicisnt Microsoft Windows illegal in that case too??
10:09.37J4k3wow, I can commit all sorts of crimes with XP WZC alone :)
10:09.47cypherdelicbut its the truth
10:09.52J4k3telnet.exe's gotta go too.
10:09.53Nuggettelnet is eeeeeeevil!
10:09.55cypherdelicvery lot tools are now illegal here
10:09.56J4k3!!!
10:10.00cypherdelicsniffers interceptors
10:10.05cypherdelicwtf i CANT do MY WQORK
10:10.10cypherdelici leave this country
10:10.39cypherdeliceven a portscanner can bring you to jail, not to use it but to OWN it
10:10.40J4k3I guess cisco catalyst etherent switches are illegal there too
10:10.47J4k3they all have port replication capability
10:10.58billybongocypherdelic: wow which country is that?
10:11.01JTyeah but they're illegal just because they're cisco
10:11.04cypherdelicJ4k3: you see our law is insane
10:11.20J4k3cypherdelic: web browsers must be illegal too
10:11.27J4k3one *could* view child pornography with them.
10:11.36J4k3eyeballs
10:11.39J4k3are illegal too
10:11.45billybongoenvelopes
10:11.52billybongomust be illegal
10:11.56cypherdelicJ4k3: our leaders could be say about any tool: its illegal
10:12.01J4k3they're nothing but a wideband microwave RF reciever
10:12.08badcfedoes xeyes create jitter on the net?
10:12.34cypherdelicJ4k3: and they do if they dont like you, i.e. if you like to hang around with socialist or terrorists
10:12.44J4k3cypherdelic: I'm not too worried about it here...  I can go buy a friggin assault rifle legally right now.
10:13.22J4k3hacker tools, who needs those?   how about a 7.62 in your server?
10:13.32cypherdelicIMHO im a terrorist, because if i had the chance to bomb of legislative, im GONNA do that :D
10:13.33JT.50cal, kthx
10:13.40cypherdelic7.62?
10:13.53JTmm
10:13.59J4k37.62x39 = standard AK47 ammo.
10:14.05cypherdelicok :)
10:14.15cypherdelicdoesnt shoot on bits and bytes
10:14.30J4k3sure it does...  aim for the bus or ram slots.
10:15.15J4k3woo, this pieceofjunk P4 seems to work
10:15.20cypherdelicin germany its funny: child pornography is illegal, thats ok, but IMMITATING a child with a 20year old girl, by pulling a lolly to her mouth or anything related to childs, then it IS a ILLEGAL childporn :D
10:15.43cypherdelicto 17years old kids fucking each other
10:15.46cypherdelicis illegal
10:15.54cypherdeliclol
10:16.04cypherdelicyeah i did a crime with 15 ;)
10:16.24J4k3I know a guy who did 10 years in the state pen for having sex with a 17.75 year old girl when he was 19.
10:16.25JTage of consent is 18?
10:16.28J4k3statutory rape.
10:16.33J4k3completely consentual sex.
10:16.34JTJ4k3: america is fucked
10:16.42cypherdelicJ4k3: germany is america is fucked
10:16.44J4k3of course, he was black, she was white, and her family had a few dollars to rub together.
10:17.11JTcypherdelic: is the age of consent 18 there?
10:17.14*** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com)
10:17.25J4k3cypherdelic: your country is so fucked you're trying to forget about WW2 entirely.
10:17.30cypherdelicJT: yes but politicans want to raise it
10:17.36Swat2LOL
10:17.37JTcypherdelic: raise?!
10:17.44J4k3yet you're breeding nazis at an outrageously high rate of speed due to the whole mystery factor of the whole event.
10:17.44JTcypherdelic: it's 16 here
10:17.45Swat2gotta love political banter
10:17.52cypherdelicJ4k3: i didnt fight WW2 your country entirely killed more people
10:17.54cypherdelicAND
10:18.09cypherdelicJ4k3: my country has been stopped, what about yours?
10:18.11JTand some states have a sliding scale allowing people just under 16 as long as they're within 2 years of each other
10:18.26Swat2Germans and Americans are as fucked as eachother. Aussies rule
10:18.33JTheh
10:18.41J4k3cypherdelic: the USA is wrapped up on shock-and-awe factor.  See Dresden, Hiroshima, Nagasaki, and 9/11.
10:18.48billybongoAussies only rule at Aussie Rules
10:18.53J4k3lots of other examples in there but its also 5:20am.
10:19.02JTaussies rule at a bit more than some sport
10:19.13J4k3aussies rule at burnouts, and thats about it.
10:19.15J4k3;)
10:19.17cypherdelicJ4k3: we are too, WW2, 9/11
10:19.17Swat2billybongo: not lately, ben cousins screwed up again, drug possession, thats his career gone :)
10:19.41billybongowhich other countries play Aussie rules?
10:19.42*** join/#asterisk Valery_Koply (n=amdvk@84.23.42.70)
10:19.47cypherdelicyou see our country is directly connected to yours, thats what your leaders do with any country they move in and kill
10:19.54J4k3cypherdelic: yeah but germany never caused mot of the population of an entire major city to die within a matter of seconds
10:19.55cypherdelicyour media is hour media
10:20.00Valery_Koplyhi all!
10:20.02cypherdelicpoliticans of hours taking the same shit
10:20.14cypherdeliccatching voices with fear
10:20.21cypherdelicours lol
10:20.27cypherdelictalking
10:20.37Swat2billybongo: ahh, but, if we didnt have Aussie Rules, other countries wouldnt have a chance at winning any sport!
10:20.57cypherdelicJ4k3: that what USA did, just to walk right into germany afterwards
10:21.14cypherdelicJ4k3: 2 cities ...
10:21.28cypherdelicJ4k3: next will be iran
10:21.35Swat2I watched Band Of Brothers Series, it was pretty good
10:21.47J4k3Iran can protect itself, and if we touch Iran, China will *own* us.
10:22.04J4k3China already warned the US government about further threats/encroachment on Iran.
10:22.07cypherdelici dont think so
10:22.20cypherdelicIran cant protect against US-Army
10:22.21J4k3China gets too much oil from Iran on the cheap
10:22.25J4k3but China can
10:22.29J4k3and China is like *right there*
10:22.33J4k3we're half a globe away
10:22.34Swat2yeah
10:22.42cypherdelicCHina cant protect anybody from USA
10:22.45cypherdelicnobody does
10:22.47billybongoSwat2: bad luck on the rugby BTW
10:22.52J4k3running around with 110k of america's best trailer park boys.
10:23.08Swat2billybongo: We obviously didnt want it enough, next year :)
10:23.19J4k3China has all the equipment it needs
10:23.31J4k3and more fight-ready men than the USA has population (no shit)
10:23.51billybongoJT> whereabouts are you?
10:23.57J4k3China beat the USA's ass in Vietnam
10:24.01J4k3when you want to come down to it.
10:24.02JTbillybongo: what about you?
10:24.17billybongoWiltshire/England/UK
10:24.21Swat2'Nam was a bit different J4k3...
10:24.23JTAustralia
10:24.25cypherdelicthey wont attack USA in any way, no state will ever try, you believe they will come and get you, for the things you did, but nobody will
10:24.25JTSydney
10:24.37billybongooh, also bad luck on the rugby then ;-)
10:24.40cypherdelicchina fears USA
10:24.46J4k3Swat2: not at all...  China has *everything* it needs to run a very huge scale war.
10:25.01JTbillybongo: i don't care about sports
10:25.04JTbillybongo: sports suck
10:25.05appelzawe are gonna win the rugby :D
10:25.06appelza^_^
10:25.10J4k3why would China fear the US?  They cut off the supply of products and every major US corporation is *fucked* overnight
10:25.13cypherdelici dont think they will fight for iran's oil, furthermore this conflict goes on to Africa
10:25.14billybongoJT, just as well, your country sucks at rugby ATM
10:25.20Valery_Koplyplease help
10:25.21J4k3from the auto industry to the computer industry, we're screwed.
10:25.32Valery_Koplyi have error in asterisk log [Oct 17 14:06:18] ERROR[3419]: rtp.c:2397 ast_rtcp_write_sr: RTCP SR transmission error to 62.140.244.100:19847, rtcp halted Operation not permitted
10:25.33billybongoappelza: you from RSA or England?
10:25.36JTbillybongo: yawn
10:25.57J4k3bah, rugby
10:26.03cypherdelicJ4k3: so when USA gets Iran, and take the oil for themselves, why should china atatck for that
10:26.07*** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-ee4f2ca9c72208f1)
10:26.18cypherdelicUSA cuts itselfs off from chinas products
10:26.30*** join/#asterisk StevenElvisda_ (n=Steven_E@202.47.107.60)
10:26.30J4k3realize something
10:26.35Swat2everyone relys on china
10:26.36J4k3the USA represents about 310 million people
10:26.43J4k3the world represents about 6.5 billion people
10:26.56J4k3therefore the USA is pretty much peanuts compared to the rest of the world market
10:27.08appelzabillybongo: RSA
10:27.09J4k3its just we pretended to have all the money forever...  now the rest of the world knows how to talk this money bullshit too
10:27.10Swat2that piece of plastic your typing on, probably comes from china
10:27.14J4k3yep, exactly
10:27.15cypherdelici personally think not the amount of people decides but the amount of money
10:27.21J4k3and if china doesn't sell me another piece of plastic, I'm fucked.
10:27.34J4k3I can't do *my* job, and therefore half this damned county loses internet service eventually.
10:27.48J4k3as the piece of plastic I'm typing on is also the service console.
10:28.06billybongoappelza: yeah, you might well win
10:28.09Swat2not to mention all the hardware boards and memory etc
10:28.22J4k3cypherdelic: america's worldwide money-hustles are going bad at a rapid rate.
10:28.25cypherdelicyes yes again but why should China attack USA
10:28.29JTmost dram chips are made in singapore and the usa
10:28.39cypherdelicChina attacking USA for Oil of Iran??
10:28.49Swat2US dollar is crapola atm
10:28.51J4k3cypherdelic: yes, do you think china randomly pulls oil out of its ass?
10:28.52cypherdelici guess the support iran with weapons etc
10:28.57J4k3its oil situation is no better than the USA's
10:28.58JTJ4k3: the australian dollar is expected to outstrip the US dollar in value by mid next year
10:29.03cypherdelicbut never going to attack usa on american ground
10:29.09cypherdelicthat would cause WW3
10:29.14J4k3JT: canada already hit like 1.02
10:29.34J4k3cypherdelic: duh...  china would just sit back and wait for us to implode.
10:30.01JThmm
10:30.08J4k3Iran also has enough pull to get us into a horrible pile of crap trade-wise.
10:30.11cypherdelicJ4k3: why, USA gets Irans oil and sells it expensive to china, china increases prices of products and USA fucked itself
10:30.41J4k3cypherdelic: that scenario doesn't exist...  Iran sells its oil to China now, China isn't going to be willing to buy the oil via the USA
10:30.48J4k3the USA will increase the price, china isn't stupid.
10:31.04cypherdelicok then USA and China both are fucked
10:31.10J4k3China also doesn't want more US influence in the region
10:31.14cypherdelicbut no WW3
10:31.29J4k3just like we wouldn't put up with China randomly taking over western hemisphere countries at random, either.
10:31.53*** join/#asterisk penguinFunk_ (n=penguin@unaffiliated/penguinfunk)
10:32.07J4k3(theres actually very very old US government policy concerning 'outside' influence of the western hemisphere)
10:36.40Swat2Whilst theres a tonne of banter going on....
10:36.54Swat2Can you do a call forward unconditional on a ring group?  I've got a bit of a unique situation where it's a home/office with 2 separate sets of lines going to 2 different ring-groups.. (Home (601) or Office (600)) i need to be able to forward the calls differently when im out of the office
10:38.55*** join/#asterisk sergee (n=serg@voip1.west-call.com)
10:40.19bagpuss_thecat11:28 < Swat2> US dollar is crapola atm
10:40.27bagpuss_thecatit will forever be known as the Yankee Dinar from now on
10:41.00*** join/#asterisk bintut (n=bintut@203.125.63.150)
10:41.44Swat2heh
10:43.02appelzathis dialing pattern: 0|. means any calls starting with 0 right?
10:43.05bintutis it possible to compile the latest asterisk without some features/modules?
10:43.13appelzahow do I have any calls starting with 0 or 1
10:47.39appelzaanyone :<
10:48.11*** part/#asterisk Strom_M (n=strom@208.127.172.112)
10:48.32*** join/#asterisk Strom_M (n=strom@208.127.172.112)
10:48.48*** join/#asterisk CVirus (n=GoD@82.201.222.194)
10:52.36*** join/#asterisk NoCarrier (n=John@unaffiliated/badpacket)
10:53.03*** join/#asterisk sehh (n=sehh@cust-224-67.on1.ontelecoms.gr)
10:53.23*** join/#asterisk BadPacket (n=John@unaffiliated/badpacket)
10:53.31billybongoappelza: maybe _{0|1}. ?
10:53.41sehhhey people
10:53.52appelzathanks, ill try
10:54.27sehhq: is there a tutorial on how to setup asterisk for a some home installation with 2 SIP phones? (no external phones, just internal VoIP software phones)
10:54.36sehhsome=small
10:54.41appelzamaybe X|. ?
10:54.42JT~thebook
10:54.42jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ ---  Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
10:54.44JTis pretty good
10:55.16sehhJT: am i going to get lost reading it? i need the very basics.. :P
10:55.16CVirussehh: http://www.asteriskguru.com/
10:55.23sehhCVirus, thanks i'll check it out
10:55.30CVirussehh: no problem
10:57.09sehhone more question: does asterisk require a database to run? i dont need call logging or anything like that, just need to be able to pick up one phone and dial the other extension
10:57.41JTnope
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11:00.36sehhhmm
11:00.51*** part/#asterisk CVirus (n=GoD@82.201.222.194)
11:00.51sehhwhen i start it, i get lots of errors about postgresql not loading/connecting
11:01.07sehhmaybe asterisk automaticaly loads modules for that and i need to disable them?
11:01.32roxluhi
11:01.35sehh(i'm using Fedora with RPM packages)
11:01.40sehhhi roxlu
11:01.51*** part/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl)
11:02.03*** join/#asterisk roxlu (n=Roxlu@84-107-142-180.dsl.quicknet.nl)
11:02.21J4k3anyone know of a chart or something on the 'net comparing ethernet chipset performance in linux?
11:02.38roxluI'm thinking to buy the Siemens Optipoint 150 S phone, but I'm curious if this is a good one?
11:08.25*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net)
11:09.01appelzaanyone know of a substitude for 0|. that will also match 1 instead of just 0
11:09.48*** join/#asterisk slavon_net (n=slavon@slavon.bigtelecom.ru)
11:11.01slavon_nethello all.. anyone may sey why "i" extention don't work in AEL? asterisk say "Call from '353' to extension '205' rejected because extension not found." but i have "i" extention in context
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11:30.27appelzasomeone provide me with a dialing rule that will match 10222 please
11:31.05JTsehh: eww, avoid those packages
11:31.18JTthey were obviously compiled by idiots
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11:39.52nicoxhello
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11:40.45nicoxdoes anyone knows why its possible that ztcfg freeze a complete system after start? (tried with zaptel 1.2.18 and 1.2.20 and kernel 2.6.18-5-amd64 (debian)
11:42.02J4k3am I completely insane to consider using FreeBSD for the host OS on a Asterisk 1.4 box (all IP calls, just using it for small conferences, voicemail, and basic call handling pretty much)
11:44.28puzzledhi
11:51.59*** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com)
11:52.25jm|laptophullo
11:52.50jm|laptopI am getting a lot (~20/min?) of OPTIONS sip dialogs in my log
11:53.03jm|laptopwhy would something be checking capabilities so frequently?
11:53.09JT100% normal
11:53.12JTswitch off sip debug
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11:53.21*** mode/#asterisk [+o blitzrage] by ChanServ
11:53.22jm|laptopJT: hmm.
11:53.42jm|laptopJT: I have had 'issues' whereby I get stutter? on active SIP calls when this line logs
11:53.56JTstutter?
11:54.08jm|laptopother party saying "you went and came back a bit there"
11:54.22*** join/#asterisk masus (n=tet@88.248.14.186)
11:54.30JTi don't see how monitoring sip messages could possibly help with that
11:54.43blitzrageSIP does not carry the media, FYI
11:54.46blitzrageRTP does
11:55.07jm|laptopwell I was wondering if this amount of OPTIONS traffic was affecting my bandwidth
11:55.10masushi all, cant see the callerid on the bt100 display , howto fix this ? I have set the callerid to "TEST" <115>
11:55.13JTno
11:55.17jm|laptopblitzrage: yes, indeed.
11:55.19jm|laptopok.
11:55.27jm|laptopmust be coincidence then
11:55.41JToptions is qualify
11:55.49jm|laptopoh is it?
11:55.54jm|laptopI don't really need that
11:56.01jm|laptopnow that I've got past testing
11:56.14JTwhat is the setup?
11:56.15jm|laptopJT: qualify helps with call boards and stuff, right?
11:56.25JTcall boards?
11:56.58jm|laptopJT: domestic, three sip phones, a couple of softphones here and there, one broker for outgoing calls, a separate one for PSTN-->VOIP
11:57.10jm|laptopJT: those boards where you can see whois on the phone and stuff
11:57.23jm|laptopmaybe my terminology is archaic :)
11:57.36JTno qualify isn't for that
11:57.46jm|laptopk
11:58.12JTit says a sip extension is accessible
11:59.45J4k3hehe
11:59.59JTit also helps with nat
12:01.10*** join/#asterisk lirakis (n=lirakis@65.200.191.253)
12:01.19Maliutabut is it g-nu g-nat ;)
12:01.27jm|laptop:/
12:01.33jm|laptopthanks for help JT
12:01.36J4k3gnunit.
12:02.31MaliutaI may have to gnash my teeth
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12:24.18masusI am using a grandstream bt100 (sw 1.0.8.33) with asterisk and it works but when receiving a call it does not show the calling parties number but its own number. What is wrong? Thanks...
12:24.47JTapart from it being one of the worst voip phones ever?
12:25.16jm|laptopteehee
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12:37.57snk00sji have trouble registering my sip channel (to place outgoing calls)
12:38.25snk00sjsip show peers shows 3stars host=(unspecified) Nat:N Status=UNKNOWN
12:38.38snk00sji am pretty sure it has to do with my NAT setup,  but i can't seem to get it working
12:38.49snk00sjalthough i have nat=yes in the sip.conf
12:39.31J4k3masus: sounds like your * is sending the wrong info for clid.
12:39.56J4k3my gs bt 101's display correct CLID.
12:40.27*** join/#asterisk zotz (n=zotz@24.244.163.157)
12:40.29J4k3I've discovered they seem to work a lot better when using a headset
12:40.47J4k3you just have to hack up the wiring to make it work with it, at least with any of the headsets I found in my house.
12:41.03[TK]D-Fendersnk00sj: If this is for an ITSP, then you should have a host filled in.
12:41.05*** join/#asterisk inso123 (n=nutcase@dsl-241-219-75.telkomadsl.co.za)
12:41.45inso123hey
12:41.49[TK]D-Fendersnk00sj: and for NAT you need a lot more :
12:41.51[TK]D-Fender~sipnat
12:41.52jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
12:41.57[TK]D-Fender^^^^^^^^^^^^^^
12:42.53*** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
12:44.04*** join/#asterisk ming_zym (n=ming_zym@124.254.57.242)
12:44.18inso123anyone here tried asterisk now?
12:44.35[TK]D-Fenderinso123: Quite possibly, qhy?
12:45.03inso123i wanna know if its anygood
12:45.15[TK]D-Fender*bleh*
12:45.28*** join/#asterisk Dr-Linux (n=Nothing@202.125.139.198)
12:45.39lirakisinso123: probably for a simple home/hobby install .. but a gui is a bind on flexibility
12:46.15snk00sjthanks [TK]D-Fender
12:46.50Dr-Linuxi've 2 t1 ports having 2 PRI's i restarted server and now having some problem:
12:47.08Dr-Linuxwhen i do "ztcfg -vv" it shows at the end:
12:47.09Dr-LinuxCAS signalling on span 2 conflicts with Clear channel on channel 40.
12:47.45Dr-Linuxis it a zaptel module bug or what? :S
12:47.45inso123everytime i start asterisk i get the error asterisk died with code 0
12:47.48inso123everytime i start asterisk i get the error asterisk died with code 0
12:47.50tzafrirDr-Linux, patch that warning out of ztcfg :-(
12:48.07Dr-Linuxooooo
12:48.27Dr-Linuxtzafrir: i google for this error and i read your name?
12:48.50[TK]D-Fenderinso123: the error message alone doesn't say much.  You'll have to pastebin your CLI output up to the point of failure
12:48.52[TK]D-Fender~pb
12:48.52jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
12:48.53[TK]D-Fender^^^^^^^^^^^^^^^^^^^^^^^^
12:48.54tzafrirDr-Linux, look for it in bugs.digium.com for best results
12:49.27[TK]D-FenderDr-Linux: is this a brand new card?
12:49.48Dr-Linuxtzafrir: so only solution is patch to fix it?
12:50.04Dr-Linux[TK]D-Fender: i'm using these cards from last 14 months
12:50.21inso123ive been battling with this damn prob
12:50.27[TK]D-FenderDr-Linux: ok, so this particular car was working 100% fine just before this issue?
12:50.32Dr-Linux[TK]D-Fender: but due to power outage in fremont data center last week server restarted
12:50.54Dr-Linux[TK]D-Fender: yes
12:50.55tzafriror move the analog card after the digital ones :-(
12:51.21Dr-Linuxtzafrir: what analog card? i don't have any analog cards
12:51.50tzafrirDr-Linux, hmmm, so it may be incorrect zaptel.ocnf after all
12:52.19lirakisinso123: start in live mode... 'asterisk -c'  and pastebin the output
12:52.47Dr-Linuxtzafrir: well, if it's wrong zaptel.conf then how it was working just fine from last 14 months in production
12:53.08*** join/#asterisk cypherdelic (n=cypherde@p5B27CA3F.dip.t-dialin.net)
12:53.14*** join/#asterisk mocker (n=ksexton@198.247.173.227)
12:53.28Dr-Linuxtzafrir: all i did, once i upgraded all asterisk/libpri/zaptel packages 2 months ago
12:54.06tzafrirCould you please pastebin: cat /proc/zaptel/*   and /etc/zaptel.conf ?
12:54.13Dr-Linuxso when server restarted 2 days ago, zaptel version didn't accept that setting :S
12:55.16Dr-Linuxtzafrir: sure,
12:55.44Dr-LinuxPS. i've fix it with different settings, but i want to understand what's going on
12:56.49mockerAre rxgain and txgain the main things to check for echo on a tdm400p?
13:01.48*** join/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net)
13:02.30Dr-Linuxtzafrir: please see here: http://phpfi.com/269498
13:02.50[TK]D-Fendermocker: first, yes.  Then check your EC routine
13:03.04mocker[TK]D-Fender: Cool, thanks.
13:03.17mocker[TK]D-Fender: Doing this all remotely, so I can't even hear the echo! :)
13:03.56[TK]D-FenderDr-Linux: you shouldn't have those 3 spans all listed as PRIMARY timing source...
13:04.37Dr-Linux[TK]D-Fender: this was not like this before, i had no span2
13:04.45Dr-Linuxbcoz i don't have pri plugged in span2
13:05.04Dr-Linuxbut that was nomore working they way it was wroking since last year
13:06.46*** join/#asterisk anonymouz666 (n=anonymou@201.19.182.176)
13:07.24mockerHuh, switchfox offers a free/unlimited demo?
13:07.34mockerThat doesn't really sound like a demo..
13:07.39mocker~switchvox
13:07.42De_Monswitchfox, heh
13:08.03mockerDe_Mon: Gimme a break, I'm pre-coffee.
13:08.04mocker:)
13:08.19De_MonI was just thinking, mozilla is doing a pbx?
13:08.36mockerYeah, but it'll just be a plugin for Firefox.
13:08.37De_Monthen the correct spelling dawned on me
13:09.00De_Monheck ya sip phone plugin for firefix
13:09.34tzafrirmocker, oslec looks quite promising (and lacks the licensing limitations)
13:09.54tzafrirDr-Linux, I don't see any problems. Strange
13:10.07tzafrirthis is not the issue that test should check for
13:10.27tzafrirEither report it as a bug or get rid of that test...
13:10.55mockertzafrir: Huh, hadn't seen that before.
13:10.56DRTHMdoes anyone know how to check zaptel version?
13:11.01mockerThat's pretty slick.
13:11.17tzafrirDRTHM, modinfo zaptel | grep ^version
13:11.22mocker"This code is the best since thing since, well, Asterisk !!!"
13:11.57tzafriralso, for most systems: cat /sys/module/zaptel/version
13:12.39DRTHMah thanks guys!
13:12.51Dr-Linuxtzafrir: did you understand my comment on pastebin .. sorry for bad english
13:14.41Kattymew.
13:15.02jm|laptopDe_Mon: a client or a wrapper?
13:16.12anonymouz666Katty: !!!
13:17.28Kattyanonymouz666: mew (=
13:18.00mockertzafrir: Agreed, based on this thread I'm reading OSLEC looks awesome.
13:20.50*** join/#asterisk famicon (i=scenesta@c51447ddc.cable.wanadoo.nl)
13:21.29Qwellmocker: they're that confident in their product, that they're willing to do so...
13:21.33Qwellswitchvox, that is
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13:25.24mockerQwell: You need to join some non-asterisk channels. :P
13:25.25*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
13:31.46anonymouz666hmmmm :)
13:32.01*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
13:32.01*** mode/#asterisk [+o anthm] by ChanServ
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13:38.50Kattyso quiet this morning.
13:39.02waKKucoffeeeeeeeeeeeeeee :X
13:39.11Kattyi see.
13:39.40anonymouz666waKKu: eu aceito. :D
13:39.59waKKuanonymouz666 hehehe :P .. me too
13:42.30Kattyman, everything is fighting me this week.
13:42.38Kattysupplier won't return my call about an rma.
13:42.44Katty3 people in the office won't return my emails.
13:44.07[TK]D-FenderKatty: s'ok.... its not paranoia... people really ARE out to get you ;)
13:44.43Kattyi KNEW it
13:45.23*** join/#asterisk Ubirajara (i=c8a0f38a@gateway/web/cgi-irc/ircatwork.com/x-82caf50647fc48f0)
13:46.35Kattymaybe i should just marry a rich guy and stop working, like all the other girls in this area.
13:47.07anonymouz666loooooool
13:47.09waKKudamn... i'm poor ;(
13:47.14anonymouz666waKKu haha
13:47.20waKKu:P
13:47.49waKKu¬¬
13:47.52Kattyhi Nugget!
13:47.57brad_msswKatty: so you're the one that posted that ad http://howardlindzon.com/?p=2725
13:48.08Kattyyeah i'm not even going to answer that.
13:48.35UbirajaraHi all, I have bug in Asterisk that had been solved in the trunk version since 01-17-07, but it has not applied to the branch version 1.4.
13:49.30waKKulol..
13:49.34UbirajaraThe bug is the 0008834
13:49.55Kattywyoming just reminds me of cows.
13:49.58*** join/#asterisk felix_da_catz (n=felix@PixNationalGroup.phonoscope.com)
13:50.53putnopvutUbirajara: I believe the bug was only fixed in trunk because you didn't mention that it was a problem in 1.4 when you opened the bug.
13:51.07lirakisKatty: remindes me of mountains... indiana reminds me of cows
13:51.15lirakisKatty: .. i dont think cows like mountains
13:51.29Kattylirakis: moo.
13:52.31Nugget]:8)
13:53.37Ubirajaraputnopvut: I didnt opened the bug, I just had it now and I found it in the bug tracker, and after analising the svn I found that it was solved at the trunk and not in the branck
13:54.19codefreezeand the wolves.
13:54.28UbirajaraHow do I do? I need to open this bug to the branch version?
13:54.48codefreezeM8834
13:55.02putnopvutcodefreeze: MuffinMan isn't in this chanel.
13:55.11putnopvutUbirajara: Sorry, I thought you had opened the bug.
13:55.28putnopvutUbirajara: you can reopen the bug and mention that it's still an issue in 1.4
13:55.32*** join/#asterisk martin_lundstrom (n=martin_l@ip-20.net-81-220-171.nice.rev.numericable.fr)
13:55.44putnopvutOr you can create a new issue and mention that it is fixed in trunk already.
13:55.50codefreezeputnopvut: apparently not!
13:55.52martin_lundstromHello folks
13:56.31Ubirajaraputnopvut: Thanks, I will do that, I will open a new issue.
13:57.59KattyNugget: there was a movie on last night, about the 10 plagues...
13:58.15KattyNugget: some scifi redo thing... mayhaps on the discovery channel, i forget ^_-
13:58.20martin_lundstromAnyone got troubles with read outs from dtmf, I get doubles that is not supposed to be there ( I used skype out to sennd the dtmf:s) Anyone have a clue what I can improve?)
13:58.28KattyNugget: but anyway, there was the bit about Cows.
13:58.35KattyNugget: or the livestock getting sick plague.
13:59.01KattyNugget: i will never be able to think about cows without thinking about the movie :<
13:59.56*** join/#asterisk corruptor (n=corrupto@styx.mcn.ru)
14:00.29martin_lundstrom(there does not seem to be any echoes on the line)
14:01.03martin_lundstromand my echo cancelation seemes to be up
14:03.18snk00sjis there an easy way to play gsm files on an ubuntu machine ?
14:03.39martin_lundstromsnk00sj: yes
14:03.58martin_lundstromMaybe this can solve my problem http://lists.digium.com/pipermail/asterisk-biz/2006-July/016423.html
14:06.06martin_lundstromsnk00sj: do you have asterisk installed?
14:06.35snk00sjon the server
14:06.43snk00sjno soundcard/vidcard there
14:06.59phearlesshi guys
14:07.16phearlessis it possible to add in the extensions.conf some "time condition",
14:07.22[TK]D-Fendersnk00sj: What are you expecting to HEAR it on then?
14:07.25phearlesslike , "if it's after 7pm"
14:07.29phearless?
14:07.34[TK]D-Fenderphearless: "show application gotoiftime"
14:07.34martin_lundstromsnk00sj: then make a call to the server and let it play the gsm for you
14:07.48phearlessthanks [TK]D-Fender !
14:08.23*** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com)
14:09.06snk00sjmargin, i am trying to listen to 500 gsm files
14:09.13snk00sjmartin, i mean
14:09.31[TK]D-Fendersnk00sj: Again, listen using WHAT?
14:10.09snk00sjusing vlcplayer, mediaplayer on my ubuntu machine
14:11.14[TK]D-Fendersnk00sj: But what device are you going to HEAR it through?  You say "play on SERVER" and it doesn't have a soundcard.
14:11.17*** join/#asterisk michael-i (n=michael-@141.41.40.55)
14:11.41[TK]D-Fendersnk00sj: You are chicken&egg-ing yourself...
14:11.48snk00sj:)
14:11.58snk00sjoké, let me explain
14:12.11snk00sji have a server with no soundcard/videocard with an asterisk setup
14:12.25snk00sjas i am setting up the interactive menu's, i downloaded some gsm files from the internet
14:12.35snk00sjand i want to listen to those on my workstation (a normal pc)
14:12.49snk00sjso i want to open those files using a multimedia player (i don't care which one)
14:12.54[TK]D-Fendersnk00sj: Does THAT system have a sound card?
14:13.01snk00sjofcourse
14:13.04snk00sj:)
14:13.19[TK]D-Fendersnk00sj: What OS?
14:13.42snk00sjubuntu gutsy using 2.6.20kernel
14:14.11*** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net)
14:14.32[TK]D-Fendersnk00sj: I would think VLC should be able to play them.  if not, worst case, use them in your dialplan and listen through a softphone.
14:15.08snk00sjit doesn't
14:17.40*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
14:18.26[TK]D-Fendersnk00sj: Then you're on to Plan-B
14:18.38[TK]D-Fendersnk00sj: Go set up your soft phone and you dialplan to play them.
14:19.06snk00sji tried that
14:19.13snk00sjafter 15 different files i got tired
14:19.28snk00sjso i need another way to just play em
14:19.42jm|laptopIAX > SIP
14:21.10[TK]D-Fendersnk00sj: Think smarter, not harder.
14:21.35[TK]D-Fendersnk00sj: number the files then make a loop to play then.
14:21.41jm|laptopalthough the concept of an IAX softphone is mashing my little mind a little
14:22.34jm|laptops/little/tiny/1
14:23.39snk00sjyou have got to be kidding me
14:23.55[TK]D-Fendersnk00sj: Why would you say that?
14:24.13*** join/#asterisk djMax (n=chatzill@artsalliancelabs.com)
14:24.15[TK]D-Fendersnk00sj: easy enough...
14:24.17snk00sjso, these files all have nice names
14:24.21snk00sjlike welcome.gsm
14:24.28snk00sji am renaming those to 1.gsm -> 99.gsm
14:24.33snk00sjand then pick the ones i like
14:24.45[TK]D-Fendersnk00sj: You aren't THINKING here....
14:24.56djMaxI'm updating * from "svn-trunk-r46489", any hugemungous changes I need to know about? Or a pointer for how to turn that into a version?
14:25.10snk00sjand in 4 weeks, when i want to change the current menu, the filename doesn't have an indication what the voice sais
14:25.22*** join/#asterisk cypherdelic (n=cypherde@p5B27D57E.dip.t-dialin.net)
14:25.22snk00sjwell, i have been playing with this thing for the past 4hours without a break
14:25.28[TK]D-Fendersnk00sj: You don't have to lose the NAME, just make a DUPLICATE numbered set, and add the number as a PREFIX to the old one for mapping.
14:25.49snk00sjand the coffee machine is begging to get used
14:26.02snk00sjwhy can't i just play those files ? :)
14:26.08*** join/#asterisk Dan0maN_Work (n=dschuh@64.149.174.136)
14:26.17snk00sjis there a codec i need, or a seperate lib i need to get
14:26.18[TK]D-Fendersnk00sj: so "welcome.gsm" gets renamed to "001-welcome.gsm), and copied as "001.gsm"
14:26.31[TK]D-Fendersnk00sj: Big bloody deal...
14:28.05[TK]D-Fendersnk00sj: Little IVR loop with *=prev, #=next, 1=skip to number, etc
14:29.22snk00sjthanks for the advice
14:29.36snk00sji do know howto loop, i just want to play those files, excuse me for being stubborn
14:31.38mockersnk00sj: xmms or winamp. ;)
14:31.46[TK]D-Fendersnk00sj: Would be easy on Windows.... Winamp does this wonderfully.
14:31.51michael-ii'm trying to get e-mail notifications running for when a call goes unanswered or is routed to voicemail because the called party is busy. here (http://pastebin.ca/739817) is the really basic implementation i have so far but it is not working for when someone simply hangs up before the dial() times out. How can I catch these cases?
14:32.08snk00sjhmm oké,  then i could launch my VM and play em in there
14:32.15snk00sjso winamp plays those by default ?
14:33.28[TK]D-Fendermichael-i:
14:33.34[TK]D-Fendermichael-i: "h" <-------
14:33.46michael-iarr!
14:33.59[TK]D-Fendermichael-i: And why do you have  "\"'s all over the place in there?
14:34.00martin_lundstromI changed my RFC2833 in sip;config and now it works some times with no double dtmfs
14:34.08michael-i[TK]D-Fender: thanks :)
14:34.16[TK]D-Fendersnk00sj: Yes
14:34.27michael-i[TK]D-Fender: it's a snip from inside a php script which generates extensions.conf
14:34.56[TK]D-Fendermichael-i: Well they had better not be in the final generation...
14:35.16*** join/#asterisk AnDY414 (n=AnDY414@81.92.157.101)
14:35.28AnDY414hello anyone here to help?
14:35.35michael-i[TK]D-Fender: trust me, they're not.
14:35.41AnDY414I have problem to compile mISDN
14:36.01AnDY414error is /usr/src/install-misdn-mqueue/mISDN-1_1_6/drivers/isdn/hardware/mISDN/capi.c:261: error: too many arguments to function âkmem_cache_createâ
14:36.18AnDY414my kernel is 2.6.23.1
14:37.51*** join/#asterisk HCEvan (i=edude@pool-72-77-226-91.tampfl.fios.verizon.net)
14:38.54nicoxhi, anyone there who is using asterisk with "a lot of traffic"?
14:39.47*** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com)
14:41.17*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
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14:50.47lirakisanyone send sms messages from asterisk?    i tried smsq from the command line .. but i dont think it went
14:52.52*** join/#asterisk wunderkin (n=wunderki@ip72-223-86-126.ph.ph.cox.net)
14:53.08lirakisnicox: whats "a lot of traffic" to you?
14:53.14shido6how do you decipher origtime in voicemail ?
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14:54.45*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
14:54.56jm|laptopI still haven't worked out if/howto set ring patterns :/
14:55.24lirakisnicox: 100,000 minutes a day?
14:55.37lirakisnicox: more, less?
14:56.16orakleman, that's a lot of minutes
14:57.14lirakisorakle: not in the carrier world!
14:57.40*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
14:58.18*** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk)
14:59.15oraklewell no
14:59.25oraklebut for a little guy running asterisk it sounds like a lot
15:02.24[TK]D-Fenderorakle: So a "little lot", not a "lot lot"? :p
15:02.53[TK]D-Fenderorakle: This is why God invented MATH.  So you could tell us your TARGET so we can say "yes feasble, or no".
15:03.46jm|laptopI need uk_ring_ring.mp3
15:03.51[TK]D-Fenderjm|laptop: Patterns to be heard on what?
15:04.06[TK]D-Fenderjm|laptop: Should be in indications.conf....
15:04.14jm|laptop[TK]D-Fender: I think I have answered my own questions: the "ring style" is client phone dependent
15:04.28[TK]D-Fenderjm|laptop: Depends.
15:04.49jm|laptopvoip:/etc/asterisk# grep country indications.conf
15:04.49jm|laptop;    order according to the 2-character country codes!
15:04.49jm|laptopcountry=uk              ; default location
15:04.54[TK]D-Fenderjm|laptop: If you are generating audio ring internally for a caller going through your system then you can set the indication tones
15:05.09jm|laptopI thought I was .....
15:05.29jm|laptopwhen I call another extension I currently get riiiiiiiiiiiiiiing in my ear and riiiiiiiiiiiiing from the phone
15:05.45jm|laptopI would prefer "ring ring" in my ear at least; and ring ring on the phone would be great, too
15:05.50jm|laptopstops the luddites freaking out
15:07.37*** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
15:08.25[TK]D-Fenderjm|laptop: Well the PHONE ringing is up to the phone itself.  What the CALLER hears is another matter.
15:08.41jm|laptop[TK]D-Fender: indeed.
15:08.52sehhq: if i've got an alarm system that must "dial out" and i've got an ISDN line, how do i make it work with Asterisk? Do i need a special PCI card to support the analog line of the alarm and redirect it over ISDN?
15:08.56jm|laptop[TK]D-Fender: I have managed to make it play moh as it's dialling
15:08.58*** join/#asterisk ming_zym (n=ming_zym@124.254.52.241)
15:09.17jm|laptopI was going to cheat and just use m(ringing_uk) or something
15:09.39[TK]D-Fendersehh: Yes
15:09.54[TK]D-Fenderjm|laptop: Perfectly feasable.....
15:09.56sehhah, thanks
15:10.02jm|laptop[TK]D-Fender: is it the right way, though?
15:10.03sehhis such a card expensive?
15:10.22jm|laptopoh wait; I might get it
15:10.42[TK]D-Fenderjm|laptop: I suppose technically not... but it ENFORCES it.... meaning that it'll take over in case * wants to pass ringing as an indication VS an audio stream.
15:10.53[TK]D-Fenderjm|laptop: In most cases I'd agree to your approach
15:11.03jm|laptop[TK]D-Fender: okies, thanks.
15:18.07*** join/#asterisk javb (n=javb@190.80.200.173)
15:19.23javbI dont have echo cancel in hardware (TDM400P), i`m getting ECHO 2 days ago dont know why, how can verify or check thath i have everything available to supress echo?
15:19.28javbzapata?...
15:20.17mockerjavb: zapata.conf rxgain/txgain can help.
15:20.31javbmocker, how?
15:20.33Maliutathe 400p can do echo cancelation
15:20.38Maliutamine does
15:20.42Maliutafxotune
15:20.50mocker~fxotune
15:21.07Qwelljavb: How new is your card?
15:21.09mockerjavb: voip-info has tons of good info on echo can..
15:21.22MaliutaI get echo at the start of some pstn call, but it get sorted fairly quickly
15:21.23mockerYou can also potentially get HPEC for free from Digium (it's their software echo can)
15:21.30javbMaliuta: fxotune? how?
15:21.46Maliutastart by reading the docs
15:21.58*** join/#asterisk jsaunders (n=super@66.119.165.91)
15:21.59MaliutaI'm going to bed since it's 1:20am
15:22.03mockerBut even not free it's like $10, and there's an open source option called..... OSLEC
15:22.08javbOk. Thanks.
15:22.20Qwelljavb: Is your card still under warranty?
15:22.46javbQwell, i have to check.. why?
15:22.50Qwellhow new is it?
15:22.57*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
15:23.06[TK]D-Fendereek
15:23.08javb8 months.
15:23.21Qwelljavb: Get the serial number off of the card, call Digium, and ask for some free HPEC licenses.
15:23.30Qwellecho-be-gone
15:23.43*** join/#asterisk NixerX (n=NixerX@rrcs-72-43-56-143.nys.biz.rr.com)
15:23.49javbQwell, where can find that serial number?
15:23.52[TK]D-Fenderjavb: PRAY my child.....
15:23.54Qwellit's on the card
15:24.11javb[TK]D-Fender: why ? :p
15:24.48[TK]D-Fenderjavb: HPEC works great for most, and worse for others.
15:25.01[TK]D-Fender(comparatively)
15:25.05Qwell[TK]D-Fender: haven't heard any complaints in a while
15:25.10NixerXI have a Somewahat off topic question for the VOIP gods here...
15:25.37[TK]D-FenderQwell: I set it up for a customer once and he said he was better off before...
15:25.44Qwelladmin error
15:26.32[TK]D-Fender:p
15:26.35NixerXWhat are the Router requirements for VOIP? Dose it differ from vendor to vendor?
15:26.41*** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com)
15:26.43Uatechey
15:26.54Uatecwhat is this "Zap/pseudo-399914746" channel?
15:27.06Uateci don't know where it came from, but it's up...
15:27.07[TK]D-FenderNivex: tahts a very gray statement.
15:27.13[TK]D-FenderUatec: ZTDUMMY
15:27.45UatecWhat's that then?
15:27.48UatecWhat's it do?
15:27.49UatecWhat's it for?
15:28.01NixerX[TK]D-Fender, true... I can be totally open ...but its not about askterisk....its VOIP in general.
15:28.46[TK]D-Fendernicox: Yes, and you haven't mentioned protocols, what kind of solution is running or ANYTHING.
15:29.01[TK]D-Fendernicox: I could ansewr that 10 different ways and none of them matching your intent
15:29.26nicoxmost traffic with E1's
15:29.51[TK]D-Fendernicox: I could say "You have a router and want your softphone to connect to a public IP SIP server" , to which I could answer "you don't need to do ANYTHING at all, and that Linksys POS is fine".
15:30.01[TK]D-Fendernicox: E1 has nothing to do with VoIP.
15:30.11nicox3 E1 with full traffic
15:30.17[TK]D-Fendernicox: MEANINGLESS.
15:30.24NixerX[TK]D-Fender, Ok...Say I have a Cisco VOIP Server.... Do I need to deploy CISCO Routers? Can I use HP aslong as it supports Qos VOIP?
15:30.27nicoxasterisk convert to VoIP
15:30.45nicoxand terminating it
15:30.55nicoxand other ones do it the other way
15:31.36[TK]D-FenderBoth of you.... just
15:31.38[TK]D-Fender~hafc
15:31.38jbotfrom memory, hafc is hire a freaking consultant.  Look, if you're having difficulty understanding what you're doing and need a solution soon, you will be far better off hiring a competent consultant than continuing to pull your hair out.
15:31.38*** join/#asterisk Katty (n=Katty@64.82.232.30)
15:31.41Kattywocka.
15:31.44Kattyjbot: hi!
15:31.45jbothi
15:31.49Kattyjbot: wocka?
15:31.51[TK]D-FenderKatty: All fear the Jabberwock!
15:32.02Kattyjbot doesn't know wocka :<
15:32.12Kattyjbot: learn wocka
15:32.19Kattyboo.
15:32.35Kattyfile: teach jbot wocka
15:32.50[TK]D-FenderKatty: What is "wocka" in your context?
15:33.09Dan0maN_Workmuppets
15:33.14Kattyyes. Fozzie Bear
15:33.23[TK]D-FenderAh
15:33.35creativxtoy.. muppets? or real ones
15:34.08[TK]D-Fender~wocka
15:34.09jbotFozzie Bear: Wocka! Wocka! Wocka! Wocka!
15:34.12KattyWocka Wocka Wocka, followed by rotten tomatoes from Statler and Waldorf.
15:34.28Katty[TK]D-Fender: there's only 3 wockas :<
15:34.40[TK]D-FenderKatty: He wasn't echo-cancelled ;)
15:34.51Kattygosh darn analog!!
15:35.40[TK]D-Fender~wocka
15:35.41jbotFozzie Bear: Wocka Wocka Wocka! (cue:  thrown rotten tomatoes from Statler and Waldorf)
15:35.43sehhq: if i set "autoload=yes" in my modules.conf, then asterisk will try to load every single compiled module?
15:35.54Kattymuch better
15:36.03[TK]D-Fendersehh: Unless specifically told to exlude some, yes
15:36.48*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
15:37.14sehhaah i see
15:37.34sehhno wonder it loads so many stuff when it starts..
15:39.02lirakis.. must reconfigure xorg for new monitor .. brb
15:39.04*** part/#asterisk lirakis (n=lirakis@65.200.191.253)
15:42.48*** join/#asterisk CVirus (n=GoD@82.201.222.194)
15:45.33*** join/#asterisk CunningPike (n=arodgers@204.239.12.183)
15:47.09disa-helpmornin!
15:47.37*** join/#asterisk lirakis (n=lirakis@65.200.191.253)
15:47.54*** join/#asterisk penguinFunk_ (n=penguin@unaffiliated/penguinfunk)
15:50.49*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-75-212-181.bflony.east.verizon.net)
15:51.08SuPrSluGhello
15:51.40SuPrSluGis it possible to use a fax line for outbound calls?
15:52.37[TK]D-FenderSuPrSluG: sURE
15:53.49*** join/#asterisk trippss (n=ss@72-20-150-196.dhcp.cmts1.phonoscope.net)
15:54.20SuPrSluGi've been trying and the call appears to go through, but nothing happens. I dial out from the cli to my number through the fax line and it never reaches my line.
15:54.59[TK]D-FenderSuPrSluG: Well I guess if you're looking for help you'd better pastebin something meaningful for us to look at...
15:58.14jarrodhow come i upgrade my polycom ip500 to bootrom 3.2.2 and sip 2.0.1 and it still doesnt support HTTP boot server in the settings?
15:59.19*** join/#asterisk syneus (n=syneus@syneus.aemcom.net)
16:01.36jm|laptopare there pitfalls to using .wav for moh files?
16:01.44[TK]D-Fenderjarrod: it does.
16:02.03jarrodi can view the versions through the status on the phones
16:02.07[TK]D-Fenderjm|laptop: No moreso that any other non-native codec format.
16:02.20jarrodbut still not HTTP option in the Server Menu
16:02.34[TK]D-Fenderjarrod: loot at it fromt he boot-rom
16:02.35jarrodis it because its an IP500, and not an IP501 ?
16:02.38jm|laptop[TK]D-Fender: but it will be transcoded to whatever codec the channel is using, right?
16:02.43[TK]D-Fenderjm|laptop: Correct
16:02.54jm|laptopk cool
16:03.03[TK]D-Fenderjm|laptop: So wav is as good a foramt as any.  Lower load than MP3 I'm sure assuming that even mattered
16:03.07[TK]D-Fenderformat*
16:03.24jm|laptopwell mp3 is giving me issues sorta
16:03.27jm|laptopso good.
16:03.53jarrodtk: i am in the initial boot setup, and the only options are TFTP/FTP
16:05.37*** join/#asterisk ZaVoid (n=zavoid@host-24-225-239-34.patmedia.net)
16:05.41ZaVoidmorning guys
16:07.22*** part/#asterisk lirakis (n=lirakis@65.200.191.253)
16:08.55ZaVoid[TK]D-Fender: hey fender you around?
16:09.59twistedI wish level 3 would calm the fsck down
16:10.12twistedi don't need a notification every 2 fscking minutes that the fscking b2b portal is going to be down
16:10.14*** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose)
16:10.41twisted(i'm serious about the 2m thing too..  10:00, 10:02, 10:04, 10:06, 10:08(2), 10:10, etc.)
16:11.08De_Monwhats the correct way to write this line: Set(CALLERID(all)="Elephant Outlook" <+15558774177>)
16:11.32De_Moncallerid displays as Elephant <+1555....>
16:12.02QwellDe_Mon: drop the quotes
16:12.26twisteddon't drop the quotes baby
16:12.28twisteddon't flip the quotes over
16:13.25*** join/#asterisk lirakis (n=lirakis@65.200.191.253)
16:13.31lirakisnice... .1680x1050
16:13.42filetwisted is a little... twisted
16:13.43Qwelllirakis: you know what's nicer than 1680x1050?
16:13.51twistedfile: how would you know?
16:13.51Qwell2 monitors at 1680x1050
16:13.55Corydon76-digfile: a LITTLE?
16:14.08filetwisted: spy satellite
16:14.10lirakisQwell: yeah.. i have too many phones on my desk... :
16:14.14twistedew.
16:14.21lirakisQwell: i could only take 1 of 2 monitors
16:14.30De_MonIiinteresting, thanks qweel
16:14.30Qwellmove the phones
16:14.33De_Monqwell
16:14.34Qwellmonitors > phones
16:14.43twistedyou know what's better than 2 monitors at 1680x1050?
16:14.52outtolunc2 computers <G>
16:14.54twisted4 monitors at 1280x1024
16:15.01Qwelltwisted: no, you fail
16:15.03QwellGTFO
16:15.07twistedattached to 3 computers
16:15.08sehhq: looking at the config files of asterisk, it seems to support ALSA. Can someone please tell me what asterisk can do with ALSA? (or OSS)
16:15.16twistedand 6 phoens
16:15.18Qwell4 monitors at 1680x1050, attached to 1 computer
16:15.27twistedi do not fail
16:15.35twistedyou are drooling over my setup
16:15.35ZaVoidmac os X in 9 days
16:15.41Qwelltwisted: no I'm not
16:15.42lirakisQwell: its a laptop.. i have it setup to dual head now .. i dont think i could do 2 external monitors though
16:15.48Qwell1280x1024 is for chumps
16:15.49twistedQwell: i can see it through the copper
16:16.34lirakisQwell; i mean.. i only have one port on the back for an ext. monitor.. so one screen has to be the laptop im pretty sure
16:16.35twistedQwell: if you'd see it, you'd shit bricks.
16:16.54Qwelllirakis: some docking bays have pcie...
16:17.07Qwelllirakis: get a quad dvi pcie video card...problem solved ;)
16:17.22Netgeeksbah, two monitors at 2560x1900 turned 90 degrees, connected to one mac  ;)
16:17.27lirakisQwell: i doubt that a quad dvi pcie video card has good linux support
16:17.29lirakislol
16:17.34Qwelllirakis: why not?
16:17.37*** join/#asterisk NOT_guru (n=chatzill@24-241-103-142.static.stls.mo.charter.com)
16:17.47Qwellit's all the same drivers as a dual...
16:17.50Qwellor even a single
16:17.51lirakisQwell: b/c many video cards have sub par support
16:17.58Qwelldoesnnvidia
16:17.59Qwellerm
16:18.03Qwelldoesn't nvidia make a quad dvi card?
16:18.12lirakisQwell: .. and .. its a laptop.. so i dont have pcie slot
16:18.27lirakisQwell: an old laptop at that
16:18.29Qwelllike I said - some laptop docking bays do
16:18.38Qwellyou clearly just need a new laptop :p
16:18.42lirakisQwell: yeah .. thats a possibility .. but .. for now .. im happy
16:18.53Qwellbuy you could be happIER
16:18.54Qwellbut*
16:18.57NOT_gurujust hoping someone might know,  can anyone point me in the right direction for reprogramming the softbuttons on a cisco 79X0 phone?
16:19.02sehhq: looking at the config files of asterisk, it seems to support ALSA. Can someone please tell me what asterisk can do with ALSA? (or OSS for that matter)
16:19.13lirakisQwell: i had a fortune cookie yesterday that said "greed leads to poverty" ...
16:19.27lirakisQwell: maybe its had an effect on me .. lol
16:19.36QwellI had a fortune cookie yesterday...
16:19.39Qwellit had no fortune
16:19.40twistednice.
16:19.47twistedyou should sue
16:19.51Qwell3 days ago actually, but whatever
16:19.52*** join/#asterisk _charly_ (i=kroseneg@sunrise.schmidham.net)
16:19.54lirakisQwell: maybe youll get hit by a bus soon
16:19.54twistedso you can get a monitor bigger than your car
16:19.59twistedand run it at 1680x1050
16:20.23NetgeeksQwell must drive a small car
16:20.31[TK]D-FenderZaVoid: Here
16:20.54anonymouz666Qwell drives a nine eleven porsche 4S
16:21.02[TK]D-Fendertwisted: I somehow doubt anyone here has a bigger monitor on their server than I do :)
16:21.10ZaVoidhey man got time for a PM.. not asterisk related
16:21.26nestArZaVoid: :P
16:21.31twisted[TK]D-Fender: big monitors on servers is a waste
16:21.35twistedservers don't need monitors
16:21.38[TK]D-Fendertwisted: Are not!
16:21.58[TK]D-Fendertwisted: My server's are multi-purpose!
16:22.02twistedahhh
16:22.03Qwellquake?
16:22.04[TK]D-Fendertwisted: http://gallery.aocomputing.net/index.php?album=2007-07-02+New+Home+Theater+%26+Table+I+was+planning+to+buy&image=02-07-07_1628.jpg
16:22.19nestArit has a flight sim file system browser, he can fly around his stuff.. like Jurrasic Park or Hackers!
16:22.20twistedthat's not a monitor
16:22.21Corydon76-digtwisted: http://web.archive.org/web/20041225135029/phreaknic.org/pix98/antimony/group_1.jpg
16:22.22twistedthat's a screen
16:22.35[TK]D-Fendertwisted: Same difference :p
16:22.47twistedCorydon76-dig: yeah, i remember that
16:22.57twistedi was pwning kryptic's machine
16:23.06nestArthat pic looks like some buildup to some kinky d&d sex.
16:23.22jarrodi guess ip500 doesnt support HTTP boot server even with bootrom 3.2.2 and sip 2
16:23.36Corydon76-dighttp://web.archive.org/web/20041225000912/phreaknic.org/pix98/ataraxia/twisted_and_mixer.jpg
16:23.37*** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net)
16:23.44twistedright when I got done with the pwnership, we called the main conf rm and had people calling up for root :)
16:23.47outtolunceveryone ** LOOK at the FLASH ** <G>
16:23.48Corydon76-digtwisted: it's this weekend
16:24.10twistedoh yeah, that's right.  I knew there was a reason I didn't want to come to nashville
16:24.15nestArif there are any girls there, i want to do them!
16:24.55Corydon76-digtwisted: you don't want to come see Decius on his drunken rant?
16:25.07hmmhesaysBah telephony depoot won't send me a replacement a200 till the bad one is back
16:25.18*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
16:25.34ZaVoidi'm depressed
16:25.41twistedi really don't have the time this year
16:25.41ZaVoidi gotta test asterisk 1.4.13 for an rtp fix
16:25.53ZaVoid1.4.9 is the perfect stable asterisk build :( everything else crashes
16:26.22Corydon76-digtwisted: ah, too bad
16:27.15*** join/#asterisk jgoddess (n=womkim@g-cipher.net)
16:27.17jgoddesshehe
16:27.17twistedyeah, i hate working over weekends, but i have shit i have to get done before my vacation next week
16:27.19jgoddessboo
16:27.21jgoddess:)
16:27.22*** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net)
16:27.29fileZaVoid: have you reported the crashes with the needed information?
16:28.07*** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk)
16:29.30twistedyay
16:29.36twistednow i get to turn this box into a fax server.
16:30.11twistedcould anything else that fun happen today? </sarcasm>
16:30.36ZaVoida ew times
16:30.40ZaVoida few times with support cases
16:31.42fileyou called Digium?
16:32.23jarrodif its outside the box
16:32.29jarrodyou have a better change troubleshooting yourself
16:33.37*** join/#asterisk TrentCreek (n=tjones14@cpe-70-117-207-168.rgv.res.rr.com)
16:36.36*** join/#asterisk sriramnrn (n=chatzill@122.167.83.11)
16:37.35Alan_Hickstwisted: At least you're getting a vacation.  :^P
16:40.18hmmhesaysoh [TK]D-Fender I could use some guidance on the 501 directory.xml
16:42.17*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
16:42.41[TK]D-Fenderhmmhesays: shoot
16:43.43Uatechey
16:43.54zerohalo[TK]D-Fender: That's my solution. Shoot the 501 and worry about the directory.xml some other time.
16:45.06*** join/#asterisk ManxPower (n=manxpowe@115.sub-70-220-244.myvzw.com)
16:45.16Uatecis there a system in asterisk where by a user can turn up to ANY phone connected to my asterisk box, and dial a number to login and recive all calls to their extension from that phone
16:45.28hmmhesaysis there any way in the global directory file to make a phone ignore itself as an entry?
16:45.34Uatecthen maybe move to a different phone, login from there and have all their calls transfered to that one instead?
16:45.42hmmhesaysI'm trying to figure out a decent way to have one directory file instead of having to create one for each phone
16:45.43Uateci'm thinking of a kind of hotdesking thing
16:45.47QwellUatec: if the phones register, sure
16:46.12BBHossi think the snom phones do this with ease
16:46.15UatecQwell, i don't mean the user carrying a phone around with him. I mean the user turning up a different phone each time
16:46.24QwellUatec: yeah, should be trivial
16:46.31[TK]D-Fenderhmmhesays: No.
16:46.35UatecQwell, how do you propose?
16:46.41zerohaloUatec: hotelling?
16:46.50hmmhesays[TK]D-Fender that sucks
16:47.07themayorif a call comes in over sip, can it not be routed to an s extension?
16:47.18[TK]D-FenderUatec: Its all dialplan, you can do whatever you want.
16:47.23Dr-Linux[TK]D-Fender: issue is not resolved yet, asterisk is being crashed again and again, i guess D channel is being dropped
16:47.35[TK]D-FenderDr-Linux: What issue?
16:48.10Dr-Linux[TK]D-Fender: the same i was having: http://phpfi.com/269498
16:48.24Uatec[TK]D-Fender, i want the user to to be able to reroute the phone calls to the new location
16:48.26[TK]D-Fenderthemayor: Typically when you register you tell the ITSP what exten to send calls to.  Either that or they arbitrarily send you the exten based on a DID you have with them, etc..
16:48.27Uatecnot to rely on me to do it
16:48.37hmmhesaysI wish you could transfer a call by selecting a directory entry also
16:49.06[TK]D-FenderUatec: Yes, it is on you to do this.
16:49.16Uatec[TK]D-Fender, that's my point though
16:49.28[TK]D-FenderUatec: (to provide the mechanism, not "run-time" programming")
16:49.42Uatecok
16:49.51Uatecnothing currently exists?
16:49.58Uatecdoes anybody even know how i would go about doing it?
16:50.03filethe tools to do it are there
16:50.09BBHosswhat kind of phones do you have?
16:50.11themayor[TK]D-Fender: yeah, i know its odd, because i have a call coming into this context, the context starts with an s extension and its not going there for someone reason, it hangs up right away, when i send it to _X it works
16:50.21Uatecmaybe when they login i could find their channel and store it in the database...
16:50.27BBHossno
16:50.30BBHossmuch easier
16:50.41*** part/#asterisk Dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
16:51.19*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
16:51.22BBHossuatec: what kind of phones do you have
16:51.24dandreHello,
16:51.31dandreI have this error:
16:51.32dandrecheck_auth: username mismatch, have <kwtk-100000>, digest has <pbxiris>
16:51.47Qwelldandre: users.conf?
16:52.12UatecBBHoss, at the moment SPA 922s
16:52.17Uatecand a snom 190
16:52.17dandreI have put fromuser = kwtk-100000 in my sip.conf
16:52.18Dr-Linux[TK]D-Fender: something looks wrong with zaptel module, not sure if i should upgrade it or downgrade it .. currently i'm using zaptel-1.2.20.1
16:52.18[TK]D-FenderUatec: make you dialplan so they can call a "log me in here"
16:52.26*** join/#asterisk ReDNeQ (n=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com)
16:52.28dandreQwell ye on one side
16:52.36BBHosshmm
16:52.44[TK]D-FenderUatec: And no, extensions.conf determines what devices get called.  Its all in there.
16:52.44Uatecand some aastras
16:52.45BBHossuatec: not familiar with either of those
16:52.48Uatecand some softphones
16:52.52Uatecbut soon we're going to choose some
16:52.57Uatecto go with properly
16:53.00Uatecbut we'v enot decided yet
16:53.24[TK]D-FenderDr-Linux: Ok, I stepped away fromt his a long while back.  Its outside of my experience.
16:53.35Uatec[TK]D-Fender, how do you store realtime information, such as which phone they called from, in extensions.conf?
16:53.40sehhq: which driver is required to run the Fritz PCI card?
16:53.50BBHossuatec: you don't want to store info like that
16:54.02Dr-Linux[TK]D-Fender: ok thanks
16:54.08BBHossuatec: you just need to register the phone to a different extension
16:54.14[TK]D-FenderUatec: store the "extension XXX is useing phone YYY" stuff in a database somewhere and look it up with an exten is called
16:54.20BBHossuatec: it will require a bit of user training though
16:54.20[TK]D-FenderBBHoss: NO
16:54.42BBHoss?
16:54.43[TK]D-FenderBBHoss: That implies users are going to reconfigure other peoples PHONES.  Thats ludicrous.
16:55.58*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
16:56.06BBHossthen why dont you chime in on how it should be done
16:56.10Uatec[TK]D-Fender, that's what I thought.  but someone said no
16:56.13[TK]D-FenderBBHoss: I just did
16:56.19UatecBBHoss, he did. as did I, but you said no
16:57.13[TK]D-FenderBBHoss: Make a "logn" exten that will map the users exten to the phone they are calling from.  When their exten gets called, yuo look up what devices they are "logged" to and ring it.
16:57.30dandreQwell: is it a known bug?
16:57.30BBHossok
16:57.46BBHosskind of like what freepbx does with the agent logon logoff deal
16:58.01BBHossbut with this dial a *code
16:58.06BBHossthen enter your agent id
16:58.07[TK]D-FenderBBHoss: Only in the loosest sense.  Thats for Queues.
16:58.19BBHossthen it maps those calls to your exten
16:58.20Kattymhmm.
16:58.23hmmhesaysgrrrr, I can't get one touch parking to work
16:58.33Kattyhmmhesays: how about two touch?
16:58.39hmmhesaysyes that works fine
16:58.40Kattyhmmhesays: kick in the tail touch?
16:58.40hmmhesays;)
16:58.48Kattyoh i see.
16:58.49Kattytricksy!
16:58.52[TK]D-Fenderhmmhesays: 3 minimum <---
16:59.03hmmhesayshaha
16:59.28hmmhesayswhat should I look for to see why its not working I have parkcall => #7 defined in features.conf
16:59.36hmmhesaysif I transfer the call to my parking extension it works fine
16:59.38[TK]D-Fenderhmmhesays: [transfer] [speed-dial-to-parking] ... listen ... [transfer]
16:59.48hmmhesaysyeah, one touch should work
16:59.56*** join/#asterisk [T]ank (n=ckwall@206.71.78.158)
17:00.00[TK]D-Fenderhmmhesays: No, it shouldn't.
17:00.13hmmhesaysIt is defined, it should
17:00.14[TK]D-Fenderhmmhesays: Thats 2 touches.  "#" + "7"
17:00.22hmmhesayshaha shut up
17:00.42[TK]D-Fenderhmmhesays: only 3 kinds of people in this world.... those that know math, and those that don't....
17:00.46dandreQwell: ?
17:00.57hmmhesaysshowing your age there
17:00.59hmmhesays:D
17:01.53[T]ankI changed a sip extension from 1008 to 1001 and now I get the error: [Oct 17 11:00:35] WARNING[11611]: chan_sip.c:8126 check_auth: username mismatch, have <1008>, digest has <1001>
17:01.53[T]ankhow can I clear that out.
17:01.53*** join/#asterisk CBU[^_^]M`` (n=love@210.213.138.6)
17:01.53mcabjarrod: that's correct - the IP 500 and IP 300 just plain don't support HTTP
17:01.53[T]ankI have factory reset the phone and made sure that it is all set for just ext 1001 and I have restarted asterisk.
17:01.57hmmhesaysok is there somewhere I should look to enable the parkcall feature
17:01.57[T]ankno matter what I do it gives me that error
17:02.05[T]ankif i set it back to 1008 it works again.
17:02.12[T]anki can dial 1001, but cannot dial out from it.
17:02.27[TK]D-Fenderhmmhesays: If you want that features.conf thing to work you have to set the DYNAMICFEATURES var before calling dial...
17:03.18Katty[TK]D-Fender: i take comfort in knowing you will always be older than me.
17:03.57hmmhesayshrm why don't I see this on the wiki
17:03.58[TK]D-FenderKatty: I take comfort in knowing I'll always be smarter than just about everybody else :)
17:04.32*** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
17:04.54hmmhesaysI don't see a reference to that anywhere
17:05.04Kattyneither do i.
17:05.08Katty[TK]D-Fender: where's your support reference?
17:05.12Katty</outofcontext>
17:06.11[TK]D-Fenderhmmhesays: http://www.voip-info.org/wiki-Asterisk+config+features.conf
17:06.20[TK]D-Fenderhmmhesays: Set(DYNAMIC_FEATURES=hangup#play#testfeature)
17:06.38[TK]D-Fenderexten => 123,1,Set(DYNAMIC_FEATURES=automon) ; enable One-touch
17:06.52[TK]D-Fenderexten => 123,2,Dial(SIP/phone100,,wW) ; wW allow one-touch recording
17:06.57hmmhesaysparkcall is not part of the applicationmap though
17:07.02[TK]D-Fender[globals]  DYNAMIC_FEATURES=>automon
17:07.24[TK]D-Fenderhmmhesays: Neither is AUTOMON, but there you have it
17:07.40[TK]D-Fenderhmmhesays: its all in there.
17:07.41*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:07.58Katty[TK]D-Fender: do i pick on you too much? or not enough?
17:08.20[TK]D-FenderKatty: Then she tried Momma-bear's and it was juuuusssttt right!
17:08.26BBHosslol
17:08.34Kattyteehee.
17:08.37Kattyk'then
17:08.53hmmhesays[TK]D-Fender: that would suggest that I can only have one feature enabled in the featuremap
17:09.11hmmhesaysand why do blindxfer and atxfer work without that , they are also under the featuremap
17:09.47Kattyanthm: ping?
17:11.11[TK]D-Fenderhmmhesays: Looks like it might be "#" delimited...
17:11.36[TK]D-Fenderhmmhesays: Set(DYNAMIC_FEATURES=hangup#play#testfeature)
17:11.56*** join/#asterisk [hC] (n=hardcore@66.119.167.162)
17:12.29hmmhesaysthat makes no sense though, why are atxfer and blindxfer working without that?
17:12.56[TK]D-Fenderhmmhesays: Corroborated by : http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
17:13.58[TK]D-Fenderhmmhesays: I presume because "transfers" are something of a given, and other functionaliy can more globally deactivated...
17:14.01hmmhesaysthat mentions nothing about parkcall in featuremap, and alot about the applicationmap section
17:14.45[TK]D-Fenderhmmhesays: What it does is show how multiple features can be enabled using that var.  Syntax and what the specific function does are 2 different things
17:14.57[TK]D-Fenderhmmhesays: At least it would appear in this case.
17:15.04[TK]D-Fenderhmmhesays: Gotta run with what you see a bit...
17:15.54hmmhesaysok
17:16.07*** join/#asterisk gardo (n=gardo@125.212.13.141)
17:18.15*** join/#asterisk Overshard (n=isaac@nc-205-240-45-138.sta.embarqhsd.net)
17:18.45OvershardHello, I'm having trouble getting asterisk to start on this system. http://pnpaste.com/show/b7cad741
17:20.17*** join/#asterisk BadPacket (n=John@unaffiliated/badpacket)
17:21.23De_MonOvershard it says here that chan_oss was loaded but no config was found, you could start by noloading chan_oss.so
17:21.32*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
17:21.40OvershardHow does one do that?
17:21.57De_Monyou also appear to be running asterisk as root, which is.. whats the term, a "bad idea"
17:22.02De_Mon~book
17:22.03jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ ---  Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
17:22.17*** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
17:22.27OvershardIt isn't a bad idea on this system it is a router ;)
17:22.33OvershardI just wanna test it out on here
17:22.46[TK]D-FenderOvershard: you are missing a config file, thats all... copy over oss.conf from the samples folder
17:22.47OvershardMy main asterisk server is running fine
17:22.47*** join/#asterisk Ritzerisk (n=Ritztech@24.120.190.66)
17:22.54OvershardOk thanks
17:24.18Ritzeriskhmmm is three a way to add elastix fax functionality to asterisk
17:25.47Katty[TK]D-Fender: you think i should setup that t1 server yet?
17:25.53Katty[TK]D-Fender: i've got the card, but still no t1
17:26.12[TK]D-FenderKatty: T1 server?
17:26.40[TK]D-FenderKatty: You mean ditching your hybrid T1>CB>TDM combo for direct CAS?
17:26.52Kattyuhh.
17:26.55Kattyi don't know what cas means.
17:27.01*** join/#asterisk grandpapadot (n=null@adsl-070-148-053-190.sip.bhm.bellsouth.net)
17:27.01Kattybut yes, no more channel bank maddness
17:27.38[TK]D-FenderKatty: Previously you had 8 channels coming in over a CAS T1 to a CB, then out to 8 analog lines and into 2x TDM400P's
17:27.57[TK]D-FenderKatty: So you want to ditcht he "middlemen, right?
17:28.45Katty[TK]D-Fender: yeah.
17:28.51Katty[TK]D-Fender: well, have more lines heh
17:28.57Katty[TK]D-Fender: i always want to ditch that telco.
17:29.00Katty[TK]D-Fender: they're a bunch of morons
17:29.35*** join/#asterisk MindTheGap (n=MindTheG@201.80.207.58)
17:29.43[TK]D-FenderKatty: And you haven't gone PRI from what I recall... VERY sad...
17:29.53*** part/#asterisk [T]ank (n=ckwall@206.71.78.158)
17:30.27*** join/#asterisk bkruse (i=bkruse@nat/digium/x-5c36adc9ad1e401b)
17:30.36bkruseanyone have any experience with the b410p?
17:31.07MindTheGaphello all, anyone experiencing fax reception problems w the new zaptel 1.4.5.1 drivers? fax here worked fine till we upgraded 1.4.4 w 1.4.5.1
17:31.51Katty[TK]D-Fender: yeah :<
17:31.56Katty[TK]D-Fender: and that telco is why.
17:32.03Katty[TK]D-Fender: they're simply mad.
17:32.10hmmhesaysdoes the polycom ip 320 use the same directory file as a 501?
17:32.14hmmhesayssame structure
17:32.23zerohalohmmhesays: yes
17:32.43*** join/#asterisk Agnt_0rnge (n=chatzill@mail.enplan.com)
17:32.43hmmhesaysbah why are my 320's not populating the directory then
17:34.39[TK]D-Fenderhmmhesays: well tell us exactly what you're trying...
17:35.10hmmhesaysi'm trying to use the same directory file to populate the contacts list on a polycom ip 501 and a 320
17:35.14hmmhesaysthe 501 is working
17:38.00[TK]D-Fenderhmmhesays: pastebin everything...
17:38.06bkruseman no one rocks the b410p
17:38.15bkruseis there like a #asterisk-euro channel? :P
17:38.18[TK]D-Fenderhmmhesays: including the FILENAMES.
17:38.41[TK]D-Fenderbkruse>man no one rocks the b410p <-- self-fulfilling prophecy
17:39.21bkruse[TK]D-Fender: haha, that true huh?
17:39.45[TK]D-Fenderbkruse: I don't make news, I just report it.
17:40.09hmmhesaysonly using one filename 000000000000-directory.xml
17:40.42hmmhesaysi just segfaulted 1.4 with a bad if statement, woohoo
17:40.43bkruse[TK]D-Fender: but of course
17:40.49blitzragebkruse: oh snap
17:41.14bkruseblitzrage: :X
17:41.16[TK]D-Fenderhmmhesays: that won't do you much good.  A phone only imports that file ONCE.
17:41.19bkruseblitzrage: you in the hsv?!
17:41.25blitzragebkruse: hehe... heck no :)
17:41.34[TK]D-Fenderhmmhesays: It then saves it to <mac>-address.xml
17:41.42bkruseblitzrage: of course not, im moving to toronto?, because you say its so awesome
17:41.53[TK]D-Fenderhmmhesays: Polycom's don't HAVE a "corporate" directory.
17:42.14BBHossyou could probably do corporate easier with XML web browser
17:42.24[TK]D-Fenderhmmhesays: thats just a "default load" file.  Once it inititalizes its Directory It'll never look again.
17:42.37Kattyhi blitzrage!
17:42.45[TK]D-Fenderhmmhesays: Indeed The MicroBrowser is the best way to implement that.
17:43.22hmmhesays[TK]D-Fender: it is reading that on every reboot
17:43.30[TK]D-Fenderhmmhesays: Shouldn't be.
17:43.36hmmhesaysit does
17:44.06[TK]D-Fenderhmmhesays: Never uses it in my experience
17:44.34hmmhesaysit says right in the doc's that is a directory file that all phones will request
17:44.45hmmhesaysall these phones are most definately requesting it
17:50.29*** join/#asterisk southtel (n=southtel@68-114-23-151.dhcp.gwnt.ga.charter.com)
17:50.39blitzrageKatty: hi!
17:50.48Netgeekshrm, in 1.4 is the CDR(userfield) function defaulted to append to the field?
17:50.55blitzragebkruse: yes, you should move to Toronto
17:52.51codefreezeNetgeeks: might be.
17:53.07Netgeekshrm, know if there is an option to replace and not append?
17:53.09[TK]D-Fenderhmmhesays: Should be according to 4-10 in the admin guide
17:53.35*** join/#asterisk marc\cba (n=l@cpc2-whit2-0-0-cust886.cdif.cable.ntl.com)
17:54.18hmmhesaysif I remove the <sd></sd> field with the poly automatically order them?
17:56.54[TK]D-Fenderhmmhesays: nope.  I believe it auto-orders NEW entires however.
17:56.57[TK]D-Fenderentries
17:58.55disa-helphmmhesays: oh joy. the love of polycom + directories
17:59.01disa-helpgood luck with that. let me know you got it to work
17:59.01disa-helpheh
18:00.21sevardhahaha
18:00.31disa-helpspeaking of polycom. much h8 to the 650's
18:02.27themayorhey can some help me out with doing something like an input loop?
18:03.17themayori want to wait for input, and if the wrong thing is entered ask again twice then hangup
18:04.03[TK]D-Fenderthemayor: "show application read" , "show application gotoif" , "show application set"
18:04.20bkruseblitzrage: that would be awesome
18:04.23bkrusehow cold is it now?
18:04.25bkrusei want a skyline :D
18:04.55[TK]D-Fenderdisa-help: What's your beef with the 650?
18:05.34blitzragebkruse: how's this?   http://www.facebook.com/photo.php?pid=24782&l=cb2f5&id=512680761
18:05.47blitzragebkruse: it's around 18-20C here now
18:05.53disa-help[TK]D-Fender: the firmware.
18:05.58blitzragebasically pants, tshirt and jacket weather
18:05.59disa-helpfirst off, going to polycom to get the bootROM is pointless
18:06.00disa-helpthey hang up on you
18:06.02disa-helpso i find the right one
18:06.12disa-helpnow it boots, grabs new bootrom, updates it, formats the filesystem, reboots
18:06.21disa-helpthen it grabs the same bootrom, updates it, formats the filesystem, reboots
18:06.24disa-helpreturn 0;
18:06.25disa-help...
18:07.40*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)
18:07.45*** join/#asterisk VJFROMGT (n=vjfromgt@68.161.227.229)
18:07.49[TK]D-Fenderdisa-help: must have messed something up in your provisioning files.  As for Polycom not handing your the firmware its known that you have to get it from your reseller.
18:07.59VJFROMGTinbound call keeps getting a busy signal http://pastebin.ca/740078
18:08.05disa-helpright, that's where i got it from
18:08.17disa-helpand the provisioning file, from what ican tell (included in the firmware docs)
18:08.20disa-helpis 100% correct
18:08.22bkruseblitzrage: you have a skyline?!!?!??
18:08.33blitzragebkruse: that's the view from my balcony / living room
18:08.36bkruseno way
18:08.39bkruseI just saw it
18:08.47bkruseWow, thats incredible.
18:08.52[TK]D-FenderVJFROMGT: Looking for user_ip in from-internal (domain 192.168.20.4) SIP/2.0 404 Not Found <-- stop trying to dial an IP as if it were an EXTENSION
18:08.56bkruseIm seriously going to have to check that out :D
18:08.58blitzragebkruse: ya... I like it a lot :)
18:08.59*** join/#asterisk michael-i (n=michael-@141.41.40.55)
18:09.14blitzragebkruse: I'm about a 20-30 min walk to the CN Tower :)
18:09.18blitzragei.e. downtown
18:09.21*** join/#asterisk stimpie (n=michiel@84-104-5-115.cable.quicknet.nl)
18:09.23bkruseman, thats insane
18:09.27bkrusewhat all is downtown for fun?
18:09.30disa-helphttp://shell.intarwebnetorg.com/outmywindow.jpg
18:09.33VJFROMGThmm.
18:09.35disa-helpheh, shitty pic, but yay 2 skylines
18:09.35[TK]D-Fenderdisa-help: Of course for YOU its 100% correct, but that doesn't make it right, so if you'd like a hand, show us what you're doing and we'll see what we can suggest.
18:09.56themayor[TK]D-Fender: thanks man!
18:10.04disa-help[TK]D-Fender: hrm, it's not mission critical, but i'll take your advice when i pull that out again
18:10.17[TK]D-Fenderdisa-help: You've closeted your phone?
18:10.26disa-help*THE* phone, yeah
18:10.43[TK]D-Fenderdisa-help: How sad.  Its the Mercedes of SIP phones.....
18:11.11disa-help[TK]D-Fender: meh, mine works fine...HRM. maybe i should look at that! :)
18:11.40J4k3http://www.intrastar.net/~jsuter/stuff/3-31-05/ = a view of my skyline.
18:11.52michael-ican someone point out my stupidity in this dialplan snip (http://pastebin.ca/740081) a simple variable check in a gosubif has a parse error
18:13.28blitzragebkruse: ummm... lots of stuff :)
18:13.43bkruseblitzrage: youll have to show me and the girl around one time :D
18:13.46blitzragebkruse: basically anything you'd want to do is there... and I live in the entertainment district, so there are concerts and things all the time here
18:13.50bkruseJ4k3: as in skyline the car?
18:13.59J4k3bkruse: no... as in a view.
18:14.00[hC]J4k3: what sort of antenna were you mounting?
18:14.05bkruseJ4k3: ahh, cool
18:14.14J4k3skylines are just overhyped maximas.
18:14.14bkruseblitzrage: thats so awesome!
18:14.19bkruseJ4k3: sure...
18:14.19*** part/#asterisk kclaussen (n=kclausse@204.13.224.242)
18:14.28bkruser34-tt? lol, ya, maximas
18:14.39bkruse:]
18:14.43J4k3hey, the only car I've ever tried to make love in was a maxima.
18:14.58bkruseJ4k3: was it difficult?
18:14.58J4k3that was like 5 years ago and I think I still have bruises.
18:15.02bkruseouch.
18:16.12J4k3[hC]: thats a 2.4 ghz omni...  it ended up with 3-bands of WISP gear on it, sectored.
18:16.29bkruseJ4k3: nice
18:16.56[hC]J4k3: i work with a WISP in vancouver, edmonton, winnipeg, and phoenix.. im familiar with your view :)
18:18.53ajohnsonso I'm trying to allow calls from any SIP user without actually creating a sip peer
18:19.00stimpiecan I log the final sip response into a cdr?
18:19.15ajohnsonI have allowguest=yes and context=from-unpriv, but I still get Failed to authenticate user
18:19.55*** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net)
18:19.58michael-ibah, i give up for tonight...8pm here, time for some FOOD!
18:20.09michael-ibye everyone
18:20.46*** join/#asterisk hi365_m (n=hi365@213.151.59.7)
18:21.04*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)
18:21.40hi365_mim having dificulty getting x-lite to work behind a dd-wrt
18:21.46*** join/#asterisk twisted (n=twisted@pdpc/supporter/active/twisted)
18:21.46*** mode/#asterisk [+o twisted] by ChanServ
18:22.13hi365_mit registeres, but there is no audio
18:22.21hi365_m(dd-wrt v23 sp2)
18:24.20[hC]with the asterisk appliance, are the fxo/fxs modular? could i open it up and rip out an fxs and put in an fxo instead?
18:26.23*** join/#asterisk vargran (n=naquad@78.26.128.253)
18:27.41*** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)
18:28.20vargranhi everyone
18:28.44deeperrorIf i'm using ATT for lines and Cavalier for LD carrier would an 800 or toll free number route thru the LD carrier or be handled only by att?
18:31.05*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)
18:31.43vargranI've installed asterisk and configured simple sip. the problem: I can't make it recieve calls. My config: http://pastebin.ca/740115 I got only sip.conf. nothing else. Client X-Lite says request timed out if I'm trying to enable "recieve calls" option
18:32.01vargranare there any ready configurations?
18:33.15*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)
18:33.48[TK]D-Fendervargran: You'd have to show us sip debug info from a call attempt to/from your SIP device
18:33.55[TK]D-Fendervargran: From * CLI
18:33.57[TK]D-Fender~pb
18:33.58jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
18:33.59[TK]D-Fender^^^^^^^^^^^^^^^^^^^^
18:34.24vargran[TK]D-Fender: how can I get that Cli? 0_o
18:34.40[TK]D-Fendervargran: "asterisk -r
18:34.51vargranby the way I've pasted config to the pastebin.ca
18:34.52[TK]D-Fendervargran: "sip debug"
18:35.11*** join/#asterisk kclaussen (n=kclausse@204.13.224.242)
18:35.35[TK]D-Fendervargran: the pastebin will not tell us the problem unless its blatant.  One this I do see is you have not defined the CODECS used for your device
18:35.38vargranI don't have it :(
18:35.47vargransip
18:35.52vargranexecutable I mean
18:36.21[TK]D-Fender?
18:36.57vargranI've installed the default configuration. sip debug doesn't work. at all: bash: sip: command not found
18:37.55vargranand where do I get it?
18:38.01[TK]D-Fendervargran: You do "sip debug" from ASTERISK CLI, no BASH CLI
18:38.12vargranWHERE DO I GET ASTERISK CLI????
18:38.14[TK]D-Fendervargran: "asterisk -r" <----------
18:38.22[TK]D-Fendervargran: pay attention
18:38.45vargranNo such command 'sip debug' (type 'help' for help)
18:38.46vargran:(
18:39.12[TK]D-Fendervargran: Where is your softphone?
18:39.18vargranon a localhost
18:39.20vargran:)
18:39.26vargranit's not on the server
18:39.28[TK]D-Fender(assuming thats what it is, and I'm sure I know the answer as well)
18:39.46[TK]D-Fendervargran: not a good answer...
18:39.54[TK]D-Fendervargran: Try being a bit clearer
18:40.43vargranI'm trying to set up asterisk on a production server, the client is box from which I'm working atm. client is X-Lite
18:41.29[TK]D-Fendervargran: try "sip show peers"
18:42.02vargranI don't have any commands prefixed with sip at all (tried 'help')
18:43.29[TK]D-Fendervargran: "load chan_sip.so"
18:43.38[TK]D-Fendervargran: then repeat the others
18:44.15vargrangot it :) commands work. posting to pastebin...
18:45.14*** join/#asterisk JimVanM (n=jimvanm@bas1-toronto63-1088799816.dsl.bell.ca)
18:45.31vargranhttp://pastebin.ca/740130
18:46.54[TK]D-Fendervargran: ok, bad user/pass
18:47.06*** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net)
18:47.51VJFROMGTdo ivr have an extension number?
18:48.04VJFROMGTor how do i point to an ivr?
18:48.07[TK]D-FenderVJFROMGT: only if you invent one that leads to it.
18:48.41vargran[TK]D-Fender: the problem is that when I'm not trying to recieve calls everything is fine: I'm allowed to call somewhere and etc :(
18:48.43[TK]D-FenderVJFROMGT: Exten => 123,1,Goto(ivrcontextthatrunsoffs,s,1)
18:48.50*** join/#asterisk GreggB (n=GreggB@66.206.86.107)
18:49.27[TK]D-Fendervargran: fix your auth setup on your x-lite, and your codecs in your SIP entry
18:49.37marc\cbaso in the dialplan how could i write something like exten => 101,1,Dial(Sip/1,,r)
18:49.49marc\cbabut specify the outgoing caller id
18:50.08[TK]D-Fendermarc\cba: Set the callerid before you dial
18:50.10vargran[TK]D-Fender: how do I do that? 0_o
18:50.16[TK]D-Fendervargran:
18:50.18[TK]D-Fender~book
18:50.19jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ ---  Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
18:50.32marc\cbaooh, i've got that book in front of me.
18:50.39marc\cbanot sure it tells me how to set caller id mind
18:50.56VJFROMGTin what file do i define ivr
18:50.57[TK]D-Fendermarc\cba: "show function CALLERID"
18:51.02marc\cbaty
18:51.04[TK]D-FenderVJFROMGT: extensions.conf
18:51.49vargranoh hfuck.... I already have it. why is open source which screams that's it's the best which can happen to ya is hard to configure? I saw some sip server which was running under windows - 3 clicks - all done. after that linux users are very surprised that linux is not a desktop system :(:(:( life is cruel :(
18:52.29*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
18:53.19[TK]D-Fendervargran: You have a completely free book that tells you how to configure everything..... how long have you been working at this?
18:53.34denonhehe
18:54.16vargranI'm a programmer and a very bad admin :( with asterisk I began today (from morning)
18:54.26denonvargran: there are like 1-click solutions for asterisk too -- boot a cd and its done
18:54.33denonbut it's not as flexible as doing it yourself
18:55.16marc\cbaso setting caller id is easy, thanks - how about if a call comes in from a peer with Remote-Party-ID: "123" <123> set for example
18:55.28[TK]D-Fendervargran: You can stop crying then.
18:55.40marc\cbaand i want to use either the From or Remote-Party-ID that came in with that original invite, as the outgoing CLI
18:55.47*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
18:55.56vargranI know about it AsteriskNOW if I don't mind, but think about it: for my purposes I need something simple, I don't have a complex infrastracture or lots of peers only 3 people and now I need: (according to the book): dhcp, lots of configuration, reading or replace a whole server with only one asterisk :(
18:56.06[TK]D-Fendermarc\cba: "show function SIP_HEADER"
18:56.13marc\cbaloving it :)
18:56.27*** join/#asterisk MACscr (n=MACscr@adsl-75-23-96-108.dsl.peoril.sbcglobal.net)
18:56.37[TK]D-Fendervargran: you are a handful of lines off from functional I'm sure.
18:57.09[TK]D-Fendervargran: But you should go fix your user/pass on X-Lite now.
18:57.26MACscrThink there is a headset i can get that will enable to use it for my pc and also my polycom? With just a button/switch to go in between the two? =P
18:57.47vargran[TK]D-Fender: second time: it works without trying to recieve calls!
18:58.03[TK]D-Fendervargran: Show me.
18:58.15vargranwant a screenshot or something?
18:58.36[TK]D-FenderMACscr: Maybe Plantronics / GN Netcom makes one.....
18:58.45[TK]D-Fendervargran: No, * CLI pastebinned...
18:58.46vargranx-lite says: ready. your username is: 1000
18:59.08vargranone sec
18:59.23[TK]D-Fendervargran: I want to see a call in or out with SIP debug
18:59.34vargranthere is nowhere to call yet
18:59.44[TK]D-Fendervargran: Well then what is "working"?
19:00.09vargranand where do I call if I can't configure asterisk to accept calls?
19:00.20[TK]D-Fendervargran: a CONSULTANT
19:00.28*** join/#asterisk Alowishus (n=jpenix@fwsdo.projectdesign.com)
19:00.48marc\cbahmm
19:00.48marc\cbaso something like
19:00.48marc\cbaexten => 101,1,CALLERID( SIP_HEADER( From ) )
19:00.48marc\cbaexten => 101,2,Dial( SIP_HEADER ( To ) )
19:00.49[TK]D-Fendervargran: go set up your dialplan...
19:00.49marc\cba?
19:01.05vargran[TK]D-Fender: I wish I know whats that...
19:01.10[TK]D-Fendermarc\cba: Go look at what it DOES and you tell me :)
19:01.19marc\cbait fails ;p
19:01.41[TK]D-Fendervargran: Welcome the CHAPTER 5 : Dialplan Basics
19:01.50vargranlooking at it
19:01.55[TK]D-Fendermarc\cba: pastebin is your friend....
19:01.59hmmhesaysok directory shiat figured out
19:02.05hmmhesaysnow I'm getting some really strange dial behavior
19:02.11hmmhesaysit isn't timing out right away when it can't find a host
19:02.20[TK]D-FenderHOST?
19:02.38vargran[TK]D-Fender: exten=>name,priority,application() and other bad syntax hash records? if yes then I found it :)
19:03.24hmmhesays[TK]D-Fender: say if extension 500 is not registered  and I try and dial 500 it is taking like 15 seconds to time out and continue with dialplan
19:03.33[TK]D-Fendervargran: Ok, stop now.  Go sit down and READ.  You are flying at this completely blind and haven't gone through the book and learned even the most basic critical stuff.
19:04.04[TK]D-Fenderhmmhesays: thats your dialplan / * DNS resolution / something else...
19:04.12*** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)
19:04.19hmmhesaysit bet it is trying to resolve SIP/500
19:04.26hmmhesaysbut this is on a local network so there is no dns
19:04.31[TK]D-Fenderhmmhesays: pastebin is your friend...
19:04.39TrentCreekno
19:04.42TrentCreekgoogle is my friend
19:04.51[TK]D-FenderTrentCreek: and I wasn't talking to YOU :p
19:05.13hmmhesaysmy dialplan is exten => 500,1,Dial(SIP/500); exten => 500,2,Voicemail(500)
19:05.13*** join/#asterisk Strom_M (n=strom@208.127.172.112)
19:05.30[TK]D-Fenderhmmhesays: Show me it executing.  and show me your peer entry
19:05.30TrentCreek:-D
19:05.36peanut-how long does it take voicepulse to activate after you send them their stupid fax?
19:06.02Kwakwadon't answer that, its rhetoric!
19:06.07hmmhesayswhat a pain, I have to connect this box to the intarwebs now
19:06.35hmmhesaystk it is dns
19:06.38*** join/#asterisk |dennis| (n=dennis@200.32.236.18)
19:06.54hmmhesayscause when I connect the box to the internet and dial it fails right away
19:06.54[TK]D-Fenderhmmhesays: host is a domain name?
19:07.14hmmhesaysits not
19:07.20hmmhesaysbut asterisk seems to be treating it as such
19:07.24[TK]D-Fenderhmmhesays: pastebin a call with SIP debug then....
19:07.51hmmhesaysasterisk should see that 500 is not registered and fail the call immediately
19:09.37hmmhesaysyou know what i'm saying?
19:09.43Corydon76-dighmmhesays: does the name contain a period?
19:09.56hmmhesaysit does not
19:10.03hmmhesaysSIP/500
19:10.10hmmhesays500 is the sip.conf entry
19:10.18Corydon76-digIs qualify=yes ?
19:10.30hmmhesaysI have no qualify
19:10.54hmmhesayscurious why that should make a difference, asterisk should not be trying to resolve 500
19:11.01Corydon76-digIs it host=dynamic?
19:11.04hmmhesaysyes
19:11.10Corydon76-digIs the host registered?
19:11.16hmmhesaysthe host is not registered
19:11.19[TK]D-Fenderhmmhesays: PASTEBIN THE CALL
19:11.20hmmhesaysthats my problem
19:12.02Corydon76-digRight, so register the phone into the server
19:12.18hmmhesaysCorydon76-dig: you missed my problem
19:12.41hmmhesayswhen the phone is not registered Dial takes like 15 seconds to time out
19:13.15Corydon76-digSo DNS is not responding in that time?
19:13.26*** join/#asterisk gandhijee (n=user@mail.win-ent.com)
19:13.36hmmhesaysmy question is why is asterisk trying to resolve SIP/500
19:13.39[TK]D-Fenderhmmhesays: PASTEBIN THE CALL <----------
19:13.42gandhijeehey, does anyone know how to recover/reset the password on a WIP300
19:13.43hmmhesayshold on
19:13.44gandhijee?
19:13.57Corydon76-dighmmhesays: it has a game plan for resolving what it is
19:14.08Corydon76-dighmmhesays: when one fails, it proceeds to the next
19:14.15hmmhesayscan I turn that off?
19:14.25Corydon76-digDon't think so
19:14.31hmmhesaysbecause what if I had some crazy peer name that matches some crazy host somewhere
19:14.38[TK]D-Fendergandhijee: http://www.google.ca/search?hl=en&q=WIP300+factory+reset&btnG=Google+Search&meta=
19:14.51[TK]D-Fendergandhijee: FIRST LINK.  Yuo may want to try a little harder....
19:14.51Corydon76-dighmmhesays: it won't, as a FQDN
19:15.07Corydon76-digunless the peer name contains a period
19:15.11hmmhesaysOk, what if I have some remote extensions that aren't registered and DNS goes down
19:15.18mockerWoo, the person who installed my A200 didn't plug the molex connector in.
19:15.29hmmhesaysdoesn't really matter if they are fxo ports
19:15.29Corydon76-digGuess you'll have to make sure that DNS doesn't go down, then
19:15.35mockerAnd... it's in another country.
19:15.35Corydon76-digLocal DNS proxy?
19:16.10marc\cbagreat so i've got SIP_HEADER and CALLERID, now how can i check if caller id has already been set?
19:16.11hmmhesaysi'm saying it makes no sense to try and resolve a known user
19:16.23[TK]D-Fender.................
19:16.41[TK]D-Fendermarc\cba: You should know this already....
19:16.56marc\cbai should? :(
19:17.07trippssDRTHM: hey ever figure out what the situation with the voice mails hanging up after 10 seconds is we discussed the other day?
19:18.28[TK]D-Fendermarc\cba: YOU KNOW HOW TO SET cid, YOU SHOULD KNOW reading IT IS THE SAME...
19:19.15hi365_mim having dificulty getting x-lite to work behind a dd-wrt
19:19.15crudpuppygot my x10001p in...now time to try it out
19:19.17hi365_mit registeres, but there is no audio
19:19.38[TK]D-Fenderhi365_m: ...
19:19.41[TK]D-Fenderhmmhesays: ~sipnat
19:19.45[TK]D-Fender~sipnat
19:19.45jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
19:19.51[TK]D-Fender&=^^^^^^^^^^^^^^^^^^
19:19.53[TK]D-Fenderaklskjsdhkjs
19:20.09marc\cbai mean - if it's already been defined in the dialplan
19:20.14marc\cba.. or not ;o
19:20.25marc\cbasort of IsDefined()
19:20.45marc\cbaor do i just need to check it has a value?
19:21.10hmmhesaysbtw [TK]D-Fender my one step parking problem was because of the Kk options not being present
19:22.26[TK]D-Fendermarc\cba: What do you want to check?
19:22.36hi365_m[TK]D-Fender: do you recomend using stun?
19:22.46[TK]D-Fenderhi365_m: No need.  Read the guide.
19:23.00trippssseen DRTHM
19:23.20trippss!seen DRTHM
19:23.30[TK]D-Fender~seen DRTHM
19:23.55jbotdrthm is currently on #asterisk (2d 3h 56m 22s). Has said a total of 31 messages. Is idling for 6h 11m 16s, last said: 'ah thanks guys!'.
19:23.55[TK]D-Fenderhrm
19:23.55trippssmmm
19:23.59trippssahy
19:24.00trippss:
19:24.05trippss:)
19:24.16trippssthx
19:30.59gandhijee[TK]D-Fender: thanks, i didn't think too look for factory defaults, just skimmed the guide for password reset
19:32.33mocker[TK]D-Fender: Can anyone add things to jbot?
19:32.50[TK]D-Fendermocker: Most can, what would you add?
19:33.01mockerOh, just wondering.
19:33.14mockerThere've been times in the past I wanted to add something.
19:33.24mocker~echo
19:33.25jbotecho is probably an issue which can be best fixed using this link: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html#AEN1718, or fixed with fxotune: http://www.voip-info.org/wiki/view/Asterisk+fxotune, or best fixed by troubleshooting your pci bus: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting, or of ...
19:33.35[TK]D-Fendermocker: I have done a LOT of jbot training, and the majority of the popular stuff was my handywork.
19:35.05sevard[TK]D-Fender: you're just a god amongst men.
19:35.26mocker[TK]D-Fender: make sure you have a backup.
19:35.39sevardoh man! that reminds me of my dream
19:35.42sevardthat i lost everything
19:35.59mockersevard: Don't forget off-site!
19:36.31sevardoff-site? I have off-shore international
19:37.13sevarddata safe - pending war.
19:37.47mockerI backup to clay tablets to avoid data rot.
19:37.52sehhq: which ports need to be open on the server running asterisk (using SIP)?
19:38.18sevard5060 and your rtp ports
19:38.53sevardwhere is that now? I haven't set up an asterisk box in months.  asterisk.conf?
19:39.39hmmhesays[TK]D-Fender this h261 rocks man
19:40.22[TK]D-Fenderhmmhesays: Yeah, the noise-cancelling mic is 100x better than that voice-tube psycho-strong one.
19:40.38sevardwtf
19:40.47sevardh261?
19:41.23[TK]D-Fendersevard: Plantronics headset
19:41.53*** join/#asterisk |Rain| (i=rain@2001:440:eeee:fffb:42:0:0:2)
19:42.18*** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com)
19:42.20hmmhesaysok is there a way to allow a caller to park a call again with one step parking after they have picked up a parked call?
19:42.21sevardthey actually put out a good product?
19:42.34|Rain|has anyone else had trouble with hold music not playing and app_queue not responding to DTMF while an Agent/ channel is running Dial()?
19:44.04putnopvut|Rain|: that sounds very familiar. Could you pastebin your dialplan?
19:44.14[TK]D-Fender|Rain|: Show us your call and we'll be able to comment.
19:44.16[TK]D-Fender~pb
19:44.17jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
19:44.38[TK]D-Fendersevard: Plantronics has all sorts of good products
19:44.59|Rain|heh.  my dialplan is 27kb all told
19:45.09|Rain|will the execution trace from verbose 3(ish) do?
19:45.38|Rain|alternately, I can try to cut out all the relevant pieces
19:46.35putnopvut|Rain|: Try pasting just the relevant pieces.
19:47.31|Rain|alrighty, give me a few
19:48.24hmmhesaysno?
19:48.41themayorhow do you match everything except a certain set of digits?
19:49.01[TK]D-Fenderthemayor: give a specific example
19:49.03themayorfor example, i want to send everything to another context unless the user input is 1
19:49.21*** join/#asterisk hi365_m (n=hi365@213.151.59.7)
19:49.37themayorif its 1, i have goto(context,s,1), how do i send it somewhere if its anything other than 1
19:49.41[TK]D-Fenderthemayor: Then yuo check if it IS "1" and continue, otherwise jump out
19:49.59themayorhow do you check the user input
19:50.28themayorgotoif?
19:50.29[TK]D-Fenderthemayor: GotoIf($["${myvar}" != "1"]?context,exten,priority)
19:50.36*** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy)
19:51.08themayorthanks
19:53.27themayor[TK]D-Fender: you're gonna think im an idiot, what is the user input called to assign it to a variable? is the user input a pre-defined variable?
19:53.55[TK]D-Fenderthemayor: .....huh?!
19:55.23*** join/#asterisk rantsh (n=rantsh@190.36.185.139)
19:55.38themayorokay, if i want to set the user input as the value for the variable, how do i do that? do i need to grab the info from the SIP_HEADER(TO)
19:57.01*** join/#asterisk darkskiez (i=mhb@cpc1-broo3-0-0-cust659.renf.cable.ntl.com)
19:57.32[TK]D-Fenderthemayor: where is this "input" coming from?
19:57.36themayorthe keypad
19:57.53[TK]D-Fenderthemayor: what what is telling * to LISTEN for it?
19:58.47*** part/#asterisk anonymouz666 (n=anonymou@201.19.182.176)
19:59.16themayorthe user calls in, they are greeted with a playback
19:59.33themayorand then there is a waitexten
19:59.41hmmhesaysyeah this sucks you can't park a call twice with one step parking
19:59.44[TK]D-Fenderthemayor: So what is TAKING their input?
20:00.06[TK]D-Fenderhmmhesays: how do you park a call twice?
20:00.13themayorthe dialplan?
20:00.14[TK]D-Fenderhmmhesays: Or why would you?
20:00.31[TK]D-Fenderthemayor: You need to be clear about where you inputis coming from...
20:00.33hmmhesays[TK]D-Fender: you can't park a call, retrieve it and park again
20:00.39[TK]D-Fenderhmmhesays: Sure you can...
20:00.47hmmhesaysnot with one step parking
20:00.52hmmhesaysit doesn't allow you to
20:01.00[TK]D-FenderO RLY?  Show me the money :)
20:01.03themayor[TK]D-Fender: what do you mean, like which channel type?
20:01.11themayoror do you want to see what im trying to do?
20:01.21[TK]D-Fenderthemayor: No, Your "input" is ALREADY in a variable....
20:01.29hmmhesaysI don't know what you want to see, after you pick up the parked call asterisk doesn't respond to the one step parking digits anymore
20:01.38themayor[TK]D-Fender: yeah, i know, im asking what its called
20:01.44[TK]D-Fenderhmmhesays: Show me where you pick it up :)
20:02.25[TK]D-Fenderthemayor: Depends how you are getting this input.  Is it via READ, or through a pattern-match in an IVR?
20:02.31sehhq: what is a low cost solution in order to connect the house alarm system to asterisk? (so that the alarm can make outbound calls to the station) PS: i believe all alarms i've seen are analog
20:02.34|Rain|okay, I think I got all of the relevant pieces: http://themuffin.net/tmp/app_queue-no-moh-no-dtmf
20:02.54[TK]D-FenderOh shit... AEL
20:03.03themayor[TK]D-Fender: neither, should i be doing a read?
20:03.16[TK]D-Fenderthemayor: Then where is your input coming from?
20:03.35hi365_m[TK]D-Fender: i read the guide. other than using stun, basicly they want you to make sure your ports are forwrded properly. mine are (other clients are connecting). my softphone can connect but i dont get audio
20:03.41hi365_mdid i miss something?
20:04.02[TK]D-Fenderhi365_m: there is a HELL OF A LOT more in that guide than just port forwarding and you have not shown us your configs.
20:04.10hmmhesaysI pick it up by dialing the extension where the call is parked
20:04.19[TK]D-Fenderhmmhesays: SHOW ME you doing this.
20:04.45hmmhesayswhat you want a webcam shot or what?
20:04.45themayor[TK]D-Fender: i dont understand what youre asking
20:05.26themayor[TK]D-Fender: a call comes in over the phone, it gets dropped to the context, where there is a WaitExten to read the input
20:05.33[TK]D-Fenderhmmhesays: No, I want to see dialplan exectution of you PICKING UP the parked call.
20:05.47themayorthe input is ${EXTEN}, thats what it is
20:06.01themayordid you understand how im getting the input?
20:06.09hmmhesaysi see
20:06.10hi365_m[TK]D-Fender:i seem to be in catagory 9 (or is it 4?). is it just hopeless?
20:06.15[TK]D-Fenderthemayor: The when you dial a valid pattern it goes to that EXTEN... I'll give you a chance to GUESS what var holds what they DIALED....
20:06.35[TK]D-Fender~sipnat
20:06.36jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
20:06.39hi365_m[TK]D-Fender: no. im 4. ill read somemore
20:06.41[TK]D-Fenderhi365_m: the FIRST guide
20:06.54[TK]D-Fenderhi365_m: Screw the 2nd
20:07.14hi365_mhope my wife doesnt see
20:07.50[TK]D-Fender|Rain|: Dial(${destext_channel},,m()M(setup_chaninfo^${destext_extension}));
20:08.23[TK]D-Fender|Rain|: You seem to be overriding your Queue's MoH by injecting it in the DIAL, and also calling a MACRO which can't be a good thing...
20:08.32|Rain|[TK]D-Fender: I shoved the m() in there as an experiment, but it makes no difference whether I have it or not
20:08.41|Rain|the macro also isn't called until the channels are about to be bridged
20:09.07lirakis[TK]D-Fender: .. do you get paid to be on IRC all day? lol ... seriously i cant get much time in here unless.. its just really really slow
20:09.10[TK]D-Fender|Rain|: pastebin CLI output.....
20:09.30[TK]D-Fenderlirakis: I multi-task well
20:12.04[TK]D-Fendersehh: TDM01B
20:12.22[TK]D-Fendersehh: or was that TDM10B.... the one with 1 FXS
20:12.32[TK]D-Fender+/- 125$USD IIRC
20:12.32|Rain|[TK]D-Fender: http://themuffin.net/tmp/no-moh-logs
20:12.33hi365_m[TK]D-Fender: if you dont mind, lets do this together: im using the following setting: http://pastebin.ca/740246 . i belive that i followed the guide properly
20:13.48crudpuppyok, who was it that hack'd the ats x10001p?
20:14.00sehherm
20:14.19[TK]D-Fenderhi365_m: externip=62.xxx.xxx.xxx <- assuming this is right, it looks OK.  your phone's router should NOT have any ports forwarded.
20:14.25sehhTDM10B i believe
20:14.28sehh1 FXS port
20:14.37hi365_mit doesnt (besides bittorrent)
20:14.45sehhis that card supported under Linux for use with asterisk?
20:14.52[TK]D-Fenderhi365_m: then it should be fine...
20:15.00[TK]D-Fendersehh: Yes, thats the idea
20:15.07sehhgreat :)
20:15.25sehhany ideas if the drivers support x86_64?
20:15.37MACscrMan, i really wish voip phones didnt have to reboot for every single little config change
20:15.51*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
20:15.51hi365_m[TK]D-Fender: except that... it isnt! (which is why im wondering if the dd-wrt does bad things to the packets)
20:16.23[TK]D-Fenderhi365_m: Possible.... test with another outside sourec
20:16.37hi365_mother clients connect fine
20:16.46hi365_m(aside for the usual pap2 flakiness)
20:16.51[TK]D-Fenderhi365_m: I guess you've pinned it then...
20:17.17hi365_mit just seems woerd that i should be the only one woth this problem
20:17.38sehhq: for a home setup, do you think its better to get an extra FXS port for an analog fax machine (which i already have)? or should i try to setup a digital software fax on the asterisk server?
20:17.45hi365_mconsider the popularity of dd-wrt and all - google doent show much (if anything) regarding such an issue
20:17.53J4k3what do you have after you get good and pissed off at a pap2 and stomp on it?  a pap smear.
20:19.18[TK]D-FenderJ4k3: har...har.. *cough*
20:21.43MACscrOmg, does the Polycom IP330 not have a back lit display?
20:22.00lirakisMACscr: OMG ... no it doesnt
20:22.09|Rain|OMG, it's madness!
20:22.10lirakisMACscr: oh wait.. sorry thought you said 301
20:22.17lirakisMACscr: i dont kniow ... OMG!!
20:22.37MACscrLol, sry, i just think its retarded for any phone made in the last 3 years or so not to have one
20:23.05lirakisMACscr: omg... lol
20:23.08lirakisokay .. ill stop
20:23.12*** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com)
20:23.24*** join/#asterisk Strom_C (n=strom@208.127.172.112)
20:23.39MACscrThanks, i appreciate it =P
20:25.12|Rain|I have even more bizarre queue problems in 1.4.13, but I'd like to get these ones fixed first (it worked in 1.4.2 and doesn't work in 1.4.11, and that is an unfortunately large jump)
20:25.15EchinosEveryone see the story about the guy that got the SWAT team to go to someone's home in the middle of the night?
20:25.44hmmhesays10.170.172.4 falls withing the assigned private network range right?
20:25.55Echinossome are wondering if he managed to spoof ANI
20:26.03Echinoshmmhesays: yes
20:26.09Echinos10.x.x.x is private
20:27.23*** join/#asterisk dexpdx (n=dex@66-162-134-242.static.twtelecom.net)
20:27.40dexpdxAnyone seen this error before: "wan_add_timer:993 Warning: WAN Timer add error: pending or func=f8c47fc6"
20:27.50dexpdxI find any reference to it
20:27.54dexpdxcan't I mean
20:28.07sehhyes but 192.x.x.x is not all private, private is only the 192.168.x.x range i believe
20:28.48sevardYeah, look at where I'm proxying from
20:28.52sevard13:26 -!- sevard [n=sev@192.235.0.85]
20:29.02|Rain|<PROTECTED>
20:29.03|Rain|<PROTECTED>
20:29.03|Rain|<PROTECTED>
20:29.03|Rain|<PROTECTED>
20:29.12|Rain|RFC 1918        Address Allocation for Private Internets   February 1996
20:29.20sevardit's always been like that.
20:29.33MACscrDoes anyone know where i add the tftp server settings in the polycom web gui?
20:29.49Echinossehh: yeah, 10 is class A, 192.168 is class B
20:29.49sevardI wonder how many people screw up and route public blocks on the internets
20:29.56sevarderm, private blocks*
20:30.30|Rain|most providers filter advertisements they receive from their downstreams to prevent private blocks and other peoples' blocks from appearing in the global routing table
20:31.11sevardinteresting
20:31.14|Rain|which isn't to say that it hasn't happened before and won't happen again
20:32.57dexpdxMACscr: as far as I know you can't through the webui
20:33.09dexpdxyou have to do it on the boot up setup menu
20:33.21MACscrLol, thats crazy, but ok, thanks
20:33.35dexpdxMACscr: yeah I know it pisses me off too
20:34.09hmmhesaysis it possible to send double quotes inside a callerid name?
20:35.59hmmhesaysseems like asterisk strips the quotes off
20:37.53MACscrDexpdx : any idea how to do a period when doing the server address?
20:38.03dexpdxhit the * key
20:38.11dexpdxat least with polycom 501 and 301
20:38.41MACscrDexpdx : yeah, i thought i tried that and just got the asterisk, guess i just had to hit it twice or something
20:39.43lirakislater all
20:39.48*** part/#asterisk lirakis (n=lirakis@65.200.191.253)
20:42.20*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
20:42.51*** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch)
20:44.45*** join/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue)
20:48.22dexpdxMACscr: one of the soft button should switch the "entry mode"
20:49.22*** part/#asterisk neverblue2 (n=neverblu@unaffiliated/neverblue)
20:49.56|Rain|bleat.  back to the drawing board, I guess
20:50.08*** join/#asterisk Schumie (n=Steve@cmarfw01.marlow.spinvox.com)
20:52.54*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
20:55.28MACscrDexpdx : any idea how to turn on the back light?
20:55.31*** join/#asterisk ct2clay (n=ct2clay@65-60-106-98.static-ip.telepacific.net)
20:56.34[TK]D-FenderMACscr, What phone?
20:56.40*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
20:56.42MACscrip330
20:57.36[TK]D-FenderMACscr, Doesn't HAVE a backlight
20:57.52MACscrWow, thats rediculous
20:58.05MACscrMy crappy grandstream even had one
20:58.21MACscrThats what i get for not doing enough research
20:58.36MACscrWow, thats so stinkin dumb
20:59.54*** join/#asterisk Twister (n=twister@mail.positech.com)
20:59.54TrentCreekThat's why you get a cheap ATA device, and you can't go wrong
20:59.54[TK]D-FenderMACscr, No, thats merely 1 feature that model doesn't have
21:00.02[TK]D-FenderMACscr, is a backlight the make-or-break feature for you?
21:00.23MACscr[TK]D-Fender : i understand it doesnt have it, but its a basic feature IMHO. It would have been a make or break feature if i knew before i bought it
21:00.50[TK]D-FenderMACscr, and now it isn't?
21:01.14MACscrI never have my over head light on in my office unless im looking for something. My two monitors usually provide enough light, but not enough to read the caller id, etc when someone calls
21:01.17*** join/#asterisk celord (n=cesar@201.195.35.62)
21:01.45MACscr[TK]D-Fender : i dont want to go through the trouble of sending it back and i dont want to spend another $150 on a phone
21:02.03MACscrPlus i would have to pay 15% restocking fee
21:02.05TrentCreekUse desk lamps and you cn't go wrong
21:02.18[TK]D-FenderMACscr, then I guess you should have done your research.
21:02.55MACscr[TK]D-Fender : your absolutely right. I should have, but i can also still bitch that they didnt include such a basic feature.
21:03.30[hC]MACscr: the only people i know of that do backlighting are polycom's 650, aastras line does, and some high end ciscos... oh and grandstream, but they really dont even count.
21:03.30[TK]D-FenderMACscr, why aren't you screaming about Cisco?  Wasn't a priority for them either....
21:03.47[TK]D-Fender650/550 have backlight
21:04.12[hC]aastra really has a solid offering if they squish the rest of their tiny bugs that are left.
21:04.34ct2claywe have aastra phones here at my place
21:04.42[hC]mainly due to their extensive flexibility in the firmware to customize softkeys and tie in XML driven apps
21:04.42[TK]D-Fender[hC], and the shitty LCD usage, and all the physical flaws
21:04.48gandhijeeSnom's are backlit
21:05.01[TK]D-Fendergandhijee, Yes... being shit is a BONUS :p
21:05.01MACscr[TK]D-Fender : I dont use ciscos or astras, so i have nothing to bitch about with them. I just made a dumb assumption because all the phones i have bought in the past had backlights
21:05.13[hC][TK]D-Fender: the lcd could be nicer, and the physical flaws are only on some models and that is preference, imo.
21:05.26gandhijee[TK]D-Fender: i've never had any problems with the 360's i have
21:05.30[hC][TK]D-Fender: whats wrong with snom?
21:05.38gandhijee[TK]D-Fender: had them for like 4 years now
21:06.48[TK]D-FenderMACscr, http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight&quote=assume.txt&file=assume.wav
21:07.16ct2clayhaha
21:07.18[TK]D-Fender[hC], I'm talking about the 57i CT here.... top of the line...
21:08.45*** join/#asterisk celord (n=cesar@201.195.35.62)
21:10.48[hC][TK]D-Fender: with regards to physical flaws? what are they? is this all bitching about how the ct pairs and is not a separate registration?
21:12.08[TK]D-Fender[hC], SHIITY rubber buttons, handset with NO weight, tinny speakerphone and handset, awkward button placement and labelling.
21:12.47ct2clayhaha... sounds like you want the beamer of phones
21:13.50[TK]D-FenderBMW = Bimbette Motor Weapon
21:13.53watchy2hey tk
21:13.59watchy2you gonna vote for colbert
21:14.34peanut-anyone know what ANI comes up when you forward a call from your cell to another number? does your cell ANI come up with the caller's CPN? or caller's CPN and ANI both forwarded?
21:15.02[hC][TK]D-Fender: i agree, i voiced those same arguments to aastras product manager directly for the 57i.. They have already changed from icons on the softkeys to text labels so its clearer, (although  locking them to english only which is strange). I do agree about the handset and rubber buttons though.. the handset problem i have mostly is hanging up the phone likes to slide left/right too easily.. and the phone currently cannot be tilted upright enough t
21:15.02[hC]o view the display that well
21:15.12ct2clayhaha
21:15.18ct2claygood def for BMW!
21:15.36[TK]D-Fender[hC], Oh yeah, forget that part :)
21:16.11ct2clay:-)
21:16.16[TK]D-FenderPolycom IP 650 = Mercedes S600 :)
21:16.22ct2clayhahaha
21:16.43watchy2whats a budgetone?
21:16.49[hC]Hey now... polycom has their share of shitty problems.
21:16.57*** join/#asterisk syneus (n=syneus@host10-39-dynamic.2-87-r.retail.telecomitalia.it)
21:17.03rantshhas anyone implemented a queue in asterisk behind a SER server?
21:17.16ct2claymy aastra makes phone calls!!! ITS GREAT
21:17.16ct2clayhaha
21:18.11TrentCreekthat;s what it is suppose to do
21:18.18gandhijeebudgetones are like geo metro's
21:18.39watchy2haha
21:19.08[TK]D-Fendergandhijee, More like Ford Fiestas
21:19.31ct2clayhaha
21:19.31ct2claywow
21:19.37ct2claycar thread
21:19.40ct2clayLOVING IT
21:19.49gandhijeei give metro's lower rankings than the fiestas
21:19.58[TK]D-Fender[hC], What few "problems" Polycom has are of far smaller significance to that of the "competition".  I can live with that.
21:19.58gandhijeefiestas are a little heaver
21:20.11ct2clayhahaha
21:20.12ct2claywow
21:20.18[TK]D-Fendergandhijee, Yeah, but Fiestas die in OTHER interesting ways :)
21:20.18[hC]I would actually say that there is no Mercedes of IP phones.  The best we've got so far is a 3 series bmw.
21:20.19[hC]:)
21:21.05[TK]D-Fender[hC], I haven't mentioned Rolls Royce, Bugatti, Ferrari, etc yet :)
21:21.25themayorany ideas why NoOp(${DATETIME}) wouldn't work, its just spitting out the NoOp on the console without the actual value of DATETIME
21:21.29[hC][TK]D-Fender: I guess it depends how you rank the problems.  Polycom for a long time has had massive issues with dealing with unknowns by crashing/rebooting. Especially with BLF.  Aside from how much i like the phone otherwise, thats a total show stopper in certain situations.  There are some things that polycom do as well that drive me to look at other phones and weigh pros/cons
21:21.55*** join/#asterisk STeven_elvisda (n=Steven_E@202.47.107.60)
21:22.03*** join/#asterisk ManxPower (n=manxpowe@115.sub-70-220-244.myvzw.com)
21:22.13[hC][TK]D-Fender: for example not being able to properly provision a polycom onto a vlan without CDP (at least not that I can find, yet) really bites.  Having soft keys as unprogrammable as they are on a polycom kinda sucks at times too.
21:22.33[hC]and by soft keys i mean builtins like Xfer, Conf, "MyBuddies" etc.
21:22.55[TK]D-Fender[hC], I'd state that as "yes Aastra's softkeys F'N ROCK...."
21:22.56gandhijeethe 601 i have has a problem with POST and GET
21:23.14[TK]D-Fender[hC], and NOBODY touches Aastra in that regard
21:23.19[hC][TK]D-Fender: their softkeys and xml programmability are what sell me.
21:23.22[hC][TK]D-Fender: I do. a Lot.
21:23.35[TK]D-Fender[hC], Placing and managing CALLS is what sells for me :)
21:23.52[TK]D-Fender[hC], Go buy your users COMPUTERS :p
21:24.31[hC][TK]D-Fender: I guess it might be easier for me, im one of the few aastra resellers in my area, and ive luckily got access to their tier 4 engineers, so if i find something that doesnt work the way i like it to, i ask, and they can give me new firmware. ive done it 3 or 4 times already.
21:25.15[hC][TK]D-Fender: not apps that are designed for a computer... thats just silly.   One killer thing i do with them is on bootup running an XML script that asks for your ext/pass and will provision that extension to what you authenticated as, with a 'logoff' softkey for roaming users..
21:25.28[TK]D-Fender[hC], Tell them to rewrite their firmware.  They are treating their new pixel displays as if they were char-matrix.  F'. annoying....
21:25.37[hC][TK]D-Fender: or being able to disable the builtin directory and substitute it with an ldap query engine
21:25.54[TK]D-Fender[hC], Yeah, but as a PHONE, Polycom kills them :)
21:26.30[hC][TK]D-Fender: the only area i would agree with you on is physical characteristics (buttons, handset weight, etc) and speakerphone audio quality
21:26.49[TK]D-Fender[hC], I'd add handset quality as well....
21:26.51[hC]and at that, to most people, its not that big of a difference..... then again, to some people it is.
21:26.55[TK]D-Fender[hC], having used both.
21:27.09[hC]I have had people try both polycom and aastra and refuse to use polycom, and love aastra. I've also had the reverse.
21:27.29*** join/#asterisk yannj_fr (n=yannj_fr@vpn.intelunix.fr)
21:27.35[hC][TK]D-Fender: personally if i could squish some of aastras programmability into polycom, and maybe give polycom a body design refresh, we would have something really great.
21:28.05[hC]the polycom [65]50/3[23]0 bodies are getting nicer.
21:28.14[hC]not so 'nintendo' looking to people.
21:28.17[TK]D-Fender[hC], Yeah, I don't want to rant all over Aastra, but I was really expecting more, and got my hopes up.  I suggest the 480i over th 5i series because of those flaws
21:29.13[hC][TK]D-Fender: yep.. I'm working really close with aastra to try to MAKE it work, but of course that only goes so far... the physical changes are the hardest, cause it costs them the most money
21:29.53[TK]D-Fender[hC], then have them made the handset pairing INDEPENDANT for me :)
21:29.57[TK]D-Fendermake*
21:30.21[hC][TK]D-Fender: what do you mean, independent pairing? so you can use any handset you want?
21:30.47[TK]D-Fender[hC], I mean as in "don't ring on the base"
21:30.48*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
21:31.03[hC][TK]D-Fender: where would you like it to ring, instead?
21:31.21[TK]D-Fender[hC], I was referring to the DECT handset, sorry for lack of clarity there.
21:31.40[hC][TK]D-Fender: oh... are you sure you dont mean the CT? cause the DECT handsets dont have bases.
21:31.43[TK]D-Fender[hC], treat the damned thing as independant if I want to..
21:31.53[TK]D-Fender[hC], Yes, the CT
21:32.16[hC][TK]D-Fender: ah. yes, i would love for it to be able to have a separate identity as well. ive mentioned that to them already.
21:32.50[hC][TK]D-Fender: theyve designed it of course with the intention that if you have a ct, you're going to be the one at the desk, and the one carrying the ct around.. which.. kinda makes sense.
21:33.13[TK]D-Fender[hC], for the handset weith, they could just bolt in a metal bar in the handset.... then again the cradle makes it too easy to "skew" off-hook.
21:33.14[hC][TK]D-Fender: what would just make it more useful is if you could have the CT have a separate registration, or even tie it to another registration on the base itself.
21:33.53[TK]D-Fender[hC], Yes, so that each handset could be a totlally different "device" to the world
21:33.54[hC][TK]D-Fender: why phone manufacturers dont include a 'cradle' for the microphone end of a handset like cisco does baffles me. it makes the 'fall off' thing too easy, and makes it harder for shit like handset lifters to work.
21:34.15[TK]D-Fender[hC], the 5i has one.. but its moe like a dent.
21:34.17[TK]D-Fendermore*
21:34.19[hC]the aastras and the polycoms both let the mic end of the handset dangle in air.
21:34.49[hC][TK]D-Fender: yeah, its a dent. polycom's is far worse... but the dent still doesnt help enough, it should actually support the entire end of the handset, just like cisco does.
21:34.52[TK]D-Fender[hC], the polycom far less so....
21:35.24[TK]D-Fender[hC], the TOP is a bit looser, but the bottm is night & day from Aastra
21:35.27[hC][TK]D-Fender: the Plantronics HL10 at least works on the aastra, where the polycom needs to be rigged badly, and is really not a solution
21:35.58[TK]D-Fender[hC], No argument there....
21:36.10[hC][TK]D-Fender: it was pretty fun at astricon this year, i got to rip grandstream's phones apart, just like we're doing now, except to the grandstream ceo and president who were attending. :P
21:36.30[hC][TK]D-Fender: not that i think it will make enough of a difference, but at least i got to tell them how i really saw it....
21:36.40[TK]D-Fender[hC], I had lifters for my CSRs.... FUGLY.  we put a screw in the lifter crossbar to depress the hookswitch and left the handset off the phone entirely :)
21:37.14[TK]D-Fender[hC], lol
21:45.24*** join/#asterisk crudpuppy (n=someone@75-138-61-254.dhcp.gnvl.sc.charter.com)
21:45.40*** join/#asterisk zentek (n=zen@modemcable009.72-58-74.mc.videotron.ca)
21:48.22sehhq: how do i setup the Fritz!Card PCI on a Fedora 7 system? what drivers are available?
21:51.48*** join/#asterisk mitcheloc (n=mitchel@adsl-67-125-156-53.dsl.irvnca.pacbell.net)
21:54.24zentekI fell bad bugging the gurus but i need some help! I am exploring the possibility of using * for an inbound call center. I am planning to need 96 concurent calls at peek. My concern is to put all my eggs in the same basket. Any pointers for an HA scenario? I have started to look at DUNDi and it looks pretty interesting but i would apreciate input from other peoples.
21:55.25[TK]D-Fenderzentek, * + SER + AudioCodes 4-port PRI gateway + reinvites
21:56.11zentekcool! but i would like to go pure voip
21:56.42[TK]D-Fenderzentek, You ask for RELIABILITY and you're thinking VoIP over the INTERNET?
21:56.43[TK]D-Fenderlol
21:56.49zenteklol
21:56.53zenteki know i know
21:57.07zentekjus trying to make voip better :-P
21:57.27rpmI'll take a tdm circuit anyday over sip/h323 gateway/provider.
21:57.39rpmunfortunately, its not my decision to make :()
21:57.43[TK]D-Fenderzentek, Then * + SER * reinvites.
21:58.51*** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk)
21:59.27[hC]rpm: hey dude.. get your t38 stuff going?
21:59.54felix_da_catzWe are running a inbound call center on * and VICIdial right now.  We are having major call quality issues.
22:00.11rpm[hC], very close.
22:00.20[hC]rpm: with linksys ATAs?
22:00.22rpmyep
22:00.22felix_da_catzOf course we have a third party dialing our leads and doing a transfer to us.  They are running a custom VoIP solution though.
22:00.32zentekhumm
22:00.41[hC]felix_da_catz: uhhh... sooo what do you expect? that sounds like a disaster waiting to happen :)
22:01.21zentekthe shop i work for has a couple of client usin * and Genesys but they are all using TDM. Only voip on a private network
22:01.39felix_da_catz[hC] It is.  We were transfering through a normal phone number first and that was bad as well.  So we though getting rid of the extra conversion and going straight SIP would help, but it didn't.
22:02.00zenteki was exploring pure voip a personal project
22:02.50felix_da_catzWhat does SER do and would it be of any benefit to us in our situation?
22:03.03[hC]felix_da_catz: its likely your end of the SIP session, in relation to your internet quality itself
22:03.32[hC]felix_da_catz: unless theres something else completely retarded going on, the likely candidate is the quality of the internet path between the two sip endpoints
22:03.41felix_da_catzWell, we are on a fiber optic connection with 100mb of bandwidth to the office.
22:03.42[hC]SER is just a sip router
22:03.56[hC]felix_da_catz: and calls are coming to you from where?
22:04.03felix_da_catzI get 50mb test downloads pretty easily, so I know that it is not our problem.
22:04.14felix_da_catzThey are coming from California in a center there.
22:04.20felix_da_catzWe are in Texas.
22:04.31[hC]felix_da_catz: its not a matter of what the capability of your pipe is, its the quality of the link between california->texas
22:04.32zentekyou'r at the mercy of the internet...
22:04.40felix_da_catzThat I can agree with.  :-)
22:04.43[hC]felix_da_catz: in particular, the quality of a lot of small UDP packets.
22:04.49crudpuppydoes anyone know if you can have two dect6 phones inthe same area?
22:05.12rpmwhenever i think of texas, i think of the show "Blue Collar Comedy Tour".. "All other states are trying to abolish the death penalty, Texas in putting in an express lane."
22:05.38zenteklol
22:05.42felix_da_catz:-)  Well the sensible people are trying to abolish it, its the other 99.99% that don't care.
22:05.42peanut-texas > *
22:06.01peanut-sensible? how is abolishing it sensible?
22:06.18J4k3the death penalty sucks
22:06.28peanut-it's cost effective
22:06.33felix_da_catzhow does it help anything?
22:06.40J4k3texas has executed like 3 or 4 now-provable-innocent.
22:06.44J4k3in the last 50 years
22:06.45[hC]youd think so until some guy kidnapps your daughter, and rapes her as he cuts her into little bits and feeds her to pigs.
22:06.50J4k3thats a pretty shitty ratio.
22:06.56felix_da_catzNot really.  It costs more to kill someone than to keep them in prison for life.  Besides, why let them die, that is the easy way out.
22:06.59[hC]then your opinion may change. (not saying i agree with it at all)
22:06.59rpm[hC], like robert pickton :)
22:07.12J4k3[hC]: then I'm gonna kill the fucker, the state won't get a chance :P
22:07.13[hC]exactly
22:07.17[hC]J4k3: :)
22:07.23peanut-why have some mass murderer sitting around in prison when you can just get rid of him?
22:07.32J4k3peanut-: why not put him to work?
22:07.43J4k3peanut-: we've got an army full of mass murderers in iraq!
22:07.47peanut-because it'll still cost more to keep him alive
22:07.51putnopvut~deathpenalty
22:07.57felix_da_catzWell, first of all they are all on 23 1/2 hour lock down with a bible to read.  In a cell the size of a ping pong table.  How is that called living?
22:08.01peanut-J4k3: oh, I see, you're one of Those..
22:08.04putnopvutshit
22:08.23[hC]cya guys later.
22:08.25J4k3peanut-: a realist?
22:08.25tzafrir~distrofight
22:08.34k31thhi guys
22:08.35peanut-a dilusional hippie.
22:08.39felix_da_catzThanks for the input guys.
22:08.47felix_da_catztime to get to the house.
22:08.48J4k3peanut-: and you're a nazi faggot...
22:08.49k31thanyone recommend a good book for asterisk ?
22:08.54Qwell~book
22:08.54jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ ---  Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
22:09.13k31thQwell: yeah iv read most of that
22:09.13sehhq: how do i setup the Fritz!Card PCI on a Fedora 7 system? what drivers are available?
22:09.19rpmi upgraded my linksys ata to firmware 5.1.12, for some reason everytime i recieve an inbound call it resets.. anyone got the same problem :)
22:09.20J4k3peanut-: now did this conversation get us anywhere?
22:09.21peanut-J4k3: that's not very nice at all.
22:09.35k31thmight buy it and read the rest on the train.
22:09.41J4k3peanut-: I'll happily beat the shit out of you, if you'd like.
22:09.56peanut-J4k3: if you can come to texas you're welcome to try
22:10.00[TK]D-FenderJERRY! JERRY! JERRY! JERRY! JERRY!
22:10.01J4k3I'm in texas
22:10.06peanut-austin?
22:10.14J4k3no, only faggots live in austin
22:10.28[TK]D-FenderJ4k3, And steers ;)
22:10.31peanut-ohwell.. well if you happen to come by austin, let me know
22:10.33*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
22:10.39J4k3austin... a town best known for its cross dressing homeless mayor candidate...
22:11.02peanut-I thought it was best known for its influx of californians
22:11.04celordhelp unload
22:11.04peanut-dirty hippies..
22:11.12[TK]D-FenderJ4k3, peanut- : So have you decided who's playing "dumb" and who's playing "dumber" in your little tirade?
22:11.14rpmthis reminds of me where ddos attacks used to start between two 14 year olds who hacked a university with loads of bandwidth.
22:11.17J4k3peanut-: the dirty hippies were there long before you were born.
22:11.38peanut-J4k3: so you won't be comming by for your attempted beatdown?
22:12.17[TK]D-Fenderah the sounds of 2 unichs arguing about who's got a bigger dick......
22:12.26peanut-[TK]D-Fender: which one was played by jim carrey?
22:12.28[TK]D-Fender*sigh*
22:12.56*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
22:13.20peanut-he's the one that said he wanted to beat me up over the internet...
22:13.34[TK]D-Fenderpeanut-, lol, and you just egg him on.
22:13.54peanut-well I really have nothing else to do at the moment
22:14.04rpmwhy was sip never engineered in the first place to use tcp as its default transport? wouldn't that have been a wiser choice?
22:14.14[TK]D-Fenderpeanut-, way to highlight your communication skill
22:14.18J4k3eh, hey...  if you were old enough to drive I'd say meet me in say, college station.
22:14.30J4k3for you I'd say you'd be better off there, more hospitals and such.
22:14.36[TK]D-Fenderrpm, tcp interruption = dead call, UPD = delay
22:14.36*** join/#asterisk adeel (n=adeeln@c-24-7-132-155.hsd1.ca.comcast.net)
22:14.57[TK]D-Fenderrpm, jitter of TCP = BLEH!!!, jitter of UDP = livable and averageable.
22:15.03rpmahh
22:15.15peanut-college station is a schlep
22:15.33J4k3no shit
22:15.52adeelis there a way to force asterisk to locally generate the rings while a call is being dialed? my provider doesn't seem to support passing back ring progress
22:15.53J4k3its the overpriced town wrapped around the most overrated college in the entire region.
22:16.08peanut-hey, you're the one that lives there, not me..
22:16.28J4k3me?  lives there?  god no.
22:16.40J4k3if I lived anywhere in the brazos valley I'd move.
22:16.42J4k3;)
22:16.51peanut-where is it you live
22:16.58J4k3I dunno, where do I live...
22:17.05peanut-I just told you I'm in austin
22:17.12J4k3mmhmm
22:17.53J4k3dammit, I was hoping you'd at least be entertaining... :P
22:21.19*** join/#asterisk CBU[^_^]M`` (n=love@210.213.140.209)
22:21.34k31thQwell: that book is free?
22:21.59*** join/#asterisk joako (n=joako@64-238-175-230.cab.apt.gru.net)
22:22.18twistedthis is why i'm hot
22:22.35joakoAnyone have a working config for T.38 + Asterisk 1.4 + Linksks or Sipira?
22:22.39mcabrpm: re SIP over UDP vs SIP over TCP - my understanding is that there's way less overhead in doing SIP over UDP, and the retry mechanisms built into SIP are more efficient for the traffic patterns of SIP than those built into TCP
22:22.56*** join/#asterisk kraptv (n=ryan@magic.skylab.org)
22:22.58*** join/#asterisk Arc^^ (n=Arc_@a82-95-179-89.adsl.xs4all.nl)
22:23.26mcabrpm: I think the main driver for SIP over TCP right now is to use TLS. I can't say I've seen anyone using TCP for the sake of using TCP
22:23.35Arc^^Hi, anybody know if i can get a digium B410p to work on asterisk 1.4 and mISDN 1.2?
22:24.04Arc^^I'm in a world of pain to get it to work
22:24.58Arc^^I got mISDN 1.1.5 to work only asterisk did not enable chan_misdn in the buildprocess
22:25.24Arc^^Now i have misdn 1.2 and asterisk does enable chan_misdn, however now the misdn 1.2 modules won't insert into the kernel with missing symbols
22:25.40*** join/#asterisk ct2clay (n=claygorm@65-60-106-98.static-ip.telepacific.net)
22:30.20kraptvAnyone here using the Polycom Soundpoint sets with Asterisk?
22:30.39kraptvI'm trying to figure out how to setup in the phone XML to emit a tone when a SIP transfer is invoked.
22:30.50kraptvright now, no sound is made which kind of confuses the users.
22:32.05*** join/#asterisk remmo (n=junk@203.32.47.250)
22:33.05*** join/#asterisk RageMax (n=RageMax@pool-72-77-120-39.pitbpa.east.verizon.net)
22:40.04[TK]D-Fenderkraptv, I don't know of any phone that "blip"s when you want to start a transfer
22:41.53dexpdxAnyone seen this error before: "wan_add_timer:993 Warning: WAN Timer add error: pending or func=f8c47fc6"
22:42.09dexpdxcan't seem to find reference in source
22:42.58*** join/#asterisk fujin (n=aj@unaffiliated/fujin)
22:43.26kraptvnot when you start a transfer, but after you complete it.  i.e. the person on the receiving end of the transfer gets a "hi, I'm transferring dude to you, BOOP" and then the person says "Hi Dude."
22:43.43*** join/#asterisk mirco (n=mirco@p54B272A2.dip.t-dialin.net)
22:43.43crudpuppywhat is this?
22:43.43crudpuppy<PROTECTED>
22:43.44crudpuppy<PROTECTED>
22:43.44crudpuppygethome*CLI>
22:43.50crudpuppyover and over
22:43.54fujinstop and start asterisk
22:44.03*** join/#asterisk lukus (n=luke@202.172.122.210)
22:44.05JTit means something is connecting to asterisk
22:44.06fujinit's a hung asterisk -r
22:44.08lukusHey Guys
22:44.17JTif you want it to stop doing that, stop stuff from accessing it
22:44.51lukusI need to put in a prefix for a number in one of my rules in my dial plan for Asterisk. I did some Google, and I found Prefix() to be *exactly* what I wanted, but Asterisk says it can't find Prefix().
22:45.08JTnever heard of Prefix
22:45.09crudpuppyfujin,  I'm rebooting as a service asterisk restart was when it starterd
22:45.09crudpuppyhehe
22:45.12JTsounds made up
22:45.22JTjust add the prefix to the dial string.
22:45.36lukusJT: How does one do such silly things?
22:45.43JT~thebook
22:45.44jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ ---  Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
22:46.46lukusha, thanks JT. Gotta love "the book" :)
22:46.54fujinlukus: what do you mean by prefix?
22:47.05fujinare you trying to append a string of numbers to an outgoing dial?
22:47.20lukusfujin: Well, I want to be able to call say 48784574, but my trunk wants me to call 0348784574
22:47.21JTa prefix is by definition prepending a string ;)
22:47.32fujinis Dial(SIP/123123${03}@outbound) not sufficient?
22:47.35JTyou must be in Australia
22:47.39fujinis Dial(SIP/123123${outbound}@outbound) not sufficient?
22:47.44fujinblahblah? ;]
22:47.45fujinmacro
22:47.46fujineasy
22:47.49lukusNo, because it destroys my billing software :P
22:47.55JTwhy do people dial like that?
22:48.13JTDial(SIP/outbound/1234${EXTEN})
22:48.32fujinlike what?
22:48.54lukusJT: That makes my billing software (Mor) think that you answered the call and then hung up :P
22:49.13JTfujin: using @ symbols in the dial string
22:49.20JTlukus: your billing software is broken
22:49.26fujinoh;
22:49.34fujinI use @symbols in a dial string to specify which peer to go over
22:49.39fujinare there multiples ways of doing this?
22:49.51JTyes, i use the same method as for zaptel :}
22:50.18lukusJT: That may be the case, but I like the billing software I have at the moment :)
22:50.26i3inaryspeaking of billing software...im about to embark on an a2billing install anyone have any advice or other opensource billing software to try if it doesnt work out
22:50.29JTDial(Technology/Resource/Number)
22:50.37fujinheh, didn't know that
22:50.45JTlukus: seriously, that is the correct way to add a prefix
22:50.45fujinbut Dial(tech/number@resource) works
22:50.54lukusi3inary: Well, Mor is alright :)
22:53.48kraptvI gotta go...
22:53.50*** part/#asterisk kraptv (n=ryan@magic.skylab.org)
22:55.04lukusJT: Can I just re-set $EXTEN?
22:55.14i3inarylukus: thanks checking it out.  seems easy.
22:55.50JTlukus: don't know if that's reliable
22:55.56JTlukus: fix your billing software.
22:59.52fujinlukus: why does Dial(tech/resource/prefix${number}) not work for you?
22:59.59joakoAnyone ever get T.38 passtru to work on 1.4?
23:00.14_ShrikEjoako: yes
23:00.29joakoHave a sample config?
23:00.57joakoTrying Grandstream and Linksys ATA and can't get a T38 invite/reinvite to be sent
23:05.52lukusfujin: No
23:08.11lukus(wish it was though)
23:10.09*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
23:12.41crudpuppyok,  I've got my softphone ups and I got a small dialplan laid out...but how do I tell it to be able to dial 2XXX for a extension?
23:12.55crudpuppyits dialing out and coming in to main extension fine
23:13.03crudpuppybut I want internal calls
23:15.38[TK]D-Fendercrudpuppy, ....
23:15.40[TK]D-Fender~book
23:15.40jbotAsterisk: The Future of Telephony 2nd Edition (ISBN  0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ ---  Downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf
23:17.33joakoDoes it have a working example of T.38?
23:21.02javbI need to run fxotune -s as a cmd before asterisk comes up, when booting, how can do this ? (CentOS)
23:21.37jer<PROTECTED>
23:22.08*** join/#asterisk TimGroe (n=TimGroe@202.172.122.211)
23:22.13joakoJav: add it to your /etc/init.d/asterisk file
23:22.19TimGroei3inary: Hi :)
23:23.09i3inaryhi
23:23.39javbjoako, where exactly.. dont understand pretty well the syntaxt of that file
23:24.27joakoHere's my file
23:24.27joakohttp://pastebin.com/m22f0b97a
23:25.02*** join/#asterisk lukus (n=luke@202.172.122.210)
23:25.09javbok, but i dont see where is fxotune there
23:25.16joakoIn that file you would add it before the line "      /usr/sbin/safe_asterisk"
23:25.19joakoYou need to add it
23:25.41joakoRead this, it should help: http://www.dartmouth.edu/~rc/classes/ksh/print_pages.shtml
23:26.06joako&& http://en.wikibooks.org/wiki/Bourne_Shell_Scripting
23:27.56*** join/#asterisk pkwong (n=chatzill@68.195.200.20)
23:28.21*** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt)
23:28.23*** join/#asterisk moprilo (n=nada@190.10.0.64)
23:28.33pkwongYay!  Transfer problems.. Anyone else having transfer problems on 1.4.x?  Seems like when I go to transfer a call, asterisk just core dumps..
23:28.40pkwongi wonder if it's a feature.
23:28.54javblook at my file, http://dpaste.com/22745/
23:29.00javbkind of different.
23:29.03javbjoako
23:29.08mopriloi have a problem, that when i connect a call to the PSTN, the voice comming from asterisk takes about 3-4 sec to appear.  What can I do to improve this?
23:29.55moprilocould it be something with the loadzone?
23:31.14joakoJav: add it after your line 72
23:31.35mopriloprogzone..
23:31.39ReDNeQpkwong, im having random drops on transfers
23:31.45ReDNeQbut not core dumps
23:32.11sehhanyone using a Fritz!Card PCI?
23:32.31pkwongit's weird..  the calling party gets moh (normal), the transferring party hears, "transfer" then it core dumps after i enter an extension..
23:32.40javbPERFECT, thanks joako, i ll read more about scripting (if thats what its name is)
23:32.49joakoYea.. shell scripts
23:33.02pkwongi rebuilt my machine thinking it was the install getting hosed.. no dice.
23:33.37pkwongslimmed down the config alot too..
23:34.03pkwongi do get "mpg123: no process killed" on the console though.
23:35.20joakopkwong: have you tried the latest asterisk, zaptel, libpri with no patches? What Version, OS, Kernel?
23:35.53RypPnpkwong: you aren't alone
23:35.59pkwongcentos 5, zaptel 1.4.5.1, etc..
23:36.01pkwongall new.
23:36.34pkwongi'm chairman of the centpbx project.. (we haven't released yet).. so we definitely have tight control over the iso and versions.
23:36.42pkwongeverything is up to date..
23:36.50pkwong1.2 doesn't have these problems.. ugh.
23:36.56pkwongwell, i'm glad i'm not alone..
23:36.59pkwong;P
23:37.27RypPnhad to roll back to 1.4.21.1 and addons-1.4.3 to recover the situation, awaiting a test box to retest soon
23:37.36joakoyou shouldn't need mpg123
23:37.37RypPn1.4.12.1 obviously
23:38.08pkwongso 1.4.12.1 and addons 1.4.3 doesn't have the new feature, huh?
23:38.43joakompg123 is depreciated for years already. MP3 is added to asterisk since 1.2 for a while and MOH is native now
23:38.43RypPnpkwong: transfers working again, if thats what you mean, and parking works again
23:38.54pkwongheh.. that's all i need..
23:39.11pkwongnext time i decide to upgrade to the latest and greatest, I'm gonna wait a little..
23:39.16joakoWhy don't you use transfer in your SIP client
23:39.24joakoI still run 1.2 for production....
23:39.37pkwongyeah.. i'm running 7970 phones..
23:39.38joako1.2.22 for main machine
23:39.41pkwongthat's the problem.
23:39.56pkwongi get 5551212@asterisk as my caller id..
23:40.00joakoSIP SCCP or skinny?
23:40.06pkwongit was quite annoying..  Running sip.
23:40.31pkwongbetween that and the MWI not working.. drove me nuts..
23:40.37pkwongand i do love my 7970
23:40.42pkwongsoooo much.
23:40.57pkwongalso have an investment of 8 of those phones.. ugh.
23:41.05joakoThe phone transfer keys should work.. never used a Cisco with SIP because their tech support can't tell me how to upgrade to SIP F/w
23:41.19pkwongahh.. i can walk you thru it..
23:41.33pkwongit's not hard.. just long and sordid.
23:41.38joakoThis was ages ago.. we returned the phones and used Polycoms instead
23:41.48pkwongi do love my polycoms as well..
23:41.49joakoBut my point is Cisco couldn't walk me throught it
23:42.01joakoBarely could tell us WHERE to get the softare... took like a week to figure that out
23:42.08pkwongthere's just something uber elegant about the 7970s on the desk.
23:42.43*** join/#asterisk ectospasm (n=ectospas@c-68-62-219-11.hsd1.al.comcast.net)
23:42.46pkwongyeah.. cisco's no help.. i bricked my first 7970.. talk about the line of crap i gave them (smartnet) to get a new one.
23:42.56pkwongthey did however replace my 7970 with a 7971-G-GE
23:43.22pkwongthen i found a guy that would sell me his whole lot of 7971s for $180 each.
23:43.27pkwongso it was a no brainer..
23:43.31joakoThe only gear I've had to send back is Granstream... they're craptastic
23:43.51pkwongahhh.. yes.. grandstream.. i'll NEVER buy one again.
23:44.07[TK]D-Fender~gs
23:44.07jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
23:44.16joakoThe GXP-2000's are OK
23:44.18joakothe HT-series
23:44.24*** join/#asterisk Teln1100A (i=hello123@bas2-toronto12-1128738711.dsl.bell.ca)
23:44.30pkwonggrandstream strikes the fear of god into me.
23:44.45JTjoako: ok, as in less crap than other grandstreams, but still crap
23:45.00joakoWe would provision them, they would download the configfile and not apply it. We'd go into the config and settings wouldn't apply and with update from keypad disabled the things wouldnt do a master reset.. basically bricked itself
23:46.42joakoI disovered a BAD bug in the GXP 2000 fw however.. one time Iw as making some prank calls (ah... the joys of VoiceChangeDial) and pressing mute while the phone rings it shows "MUTE" On the display but it really isn't muted!
23:48.39pkwongzaptel 1.4.5.1 is ok with 1.4.3 addons and 1.4.12.1?
23:49.01*** join/#asterisk remmo (n=junk@203.32.47.250)
23:53.54joakoAnyone have t.38 working with Asterisk 1.4?
23:53.59|Rain|pkwong: should be
23:54.24|Rain|so, I don't suppose anyone came up with a solution for my app_queue problems I asked about a couple of hours ago
23:55.20pkwongcool :) thanks rain.
23:55.27pkwongwhat's the app_queue issue?
23:57.14|Rain|I'm calling app_queue, which is calling out to Agent/... extensions, and hold music stops while trying each agent and DTMF detection (to jump out of the queue) isn't working at all
23:57.59joako1.2 or 1.4
23:59.11pkwongwhat version of *?

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