irclog2html for #asterisk on 20060501

00:01.11*** part/#asterisk swytch (n=ezcall@d83-179-214-255.cust.tele2.fr)
00:08.05robin_szword
00:08.13ManxPowerAMP/FreePBX/Asterisk@Home users should join #freepbx for support
00:09.09ManxPower<PROTECTED>
00:09.40drfoomod2no vin, nano
00:09.43drfoomod2vim
00:09.47Strom_CAMP/FreePBX/Asterisk@Home users should join #what-do-you-mean-I-have-to-use-the-keyboard for support
00:09.57drfoomod2Strom_C: you don;t want it hurt _that_ bad
00:10.05ManxPower~thebook
00:10.06jbot[thebook] Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
00:10.25Strom_Cdrfoomod2: ?
00:10.29orlokthe thing is
00:10.39drfoomod2[19:56] <Strom_C> just you, asterisk, and vim
00:11.03Strom_Cdrfoomod2: hey, thats how I work with asterisk
00:11.03orlokmost admins who manage any decent amount pof systems eventually see the good things about automated scripts and smart templating systems rather than making every single change by hand
00:11.25robin_szthe basic problem is that theres a difference between "I ahve aproblem with asterisk" and "I have a problem with the configuration written automagically, by a GUI I dont understand, and I dont understand the configuration either"
00:11.41robin_szwell yes
00:11.43robin_szbut ...
00:11.56Strom_Corlok: sure, but thats why you write custom scripts suited to your exact need
00:11.59robin_szwhen you need support, go see the author of the templating system
00:12.20orlokrobin_sz: well, the specific templates that manage asterisk
00:12.31orlokas the same templates also handle apache, squid, qmail, dhcp, etc etc
00:12.44robin_sznot so ...
00:12.53orlokit makes it very easy to collect and manage customisations, push them out to lots of servers, etc
00:12.56surfduei know this isnt the room but i use freepbx, does anyone know off hand why freepbx could not be writing to the configs with no errors?
00:13.03orlokheh
00:13.19*** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it)
00:15.04robin_szorlok, the problem is that 99% of the problems sre not from asterisk per-se, but from the configuration written by freepbx, a@h etc, we dont know how they manage their configs, the users dont either, the only people they can go to for support is #freepbx or #amp or whatever
00:15.34robin_szlike the guy just then
00:15.43orlokrobin_sz: yeah, it would be like trying to diagnose a bug in java from the output of a coldfusion developer
00:16.00robin_szright
00:16.01orlokfix an engine by looking at the dash lights, etc
00:16.18orlokthe bonnet must be open! :)
00:16.44robin_szworse than that, fix an engine, based on a description of the dash lights from someone who doesnt understand them
00:17.24jqlhmm... linux must be easy to install these days with the abundance of gui-addicts using it. I'm heartened
00:17.31orlok"the watering can is red!"
00:17.57orlokjql: yup
00:18.05orlokjql: www.contribs.org
00:18.08robin_szbut ... they need to go see their integrator for support
00:18.56robin_szmuch the same way as you wouldnt get the ford engine design team being hassled by old granny smithers about her non-working fiesta
00:19.21robin_szs/fiesta/some other flavour of ford/
00:19.27wunderkinorlok, your packet was out of order
00:19.35orlokheh
00:21.35orlokhmm. 41 meg of bzip2'd .sql files
00:21.35*** join/#asterisk niZon (n=ilt@S010600080db4ab60.wp.shawcable.net)
00:23.39*** join/#asterisk RES2 (n=RES@chello213047231029.tirol.surfer.at)
00:25.25tainted-<Strom_C> orlok: you should try to set up a system without any guis or prototyping tools or anything
00:25.27tainted-idiot
00:26.29Strom_Chey, look, it's the troll
00:26.56tainted-nothing wrong with guis
00:27.07tainted-depends on the needs of the end user
00:27.23RES2Hi.
00:27.23Strom_Ctainted-: he was complaining about being unfamiliar with asterisk, so I suggested that as an exercise to increase familiarity
00:27.23tainted-just empower ppl and get the fuck out of the way
00:27.26Strom_Cifiot
00:27.31Strom_Calso, typinh
00:27.34RES2Can anyone help me with spandsp please?
00:27.36Strom_Cblah
00:28.23Strom_CI cut my fingers off
00:28.27filethat's unfortunate
00:28.27Strom_Cso I can't poke you really
00:28.36robin_szend users should always be dealt with by the highest level of abstraction
00:28.38Strom_Ci can kind of jam my elbow into you, though
00:28.56websaesuch pleasant thoughts!
00:29.20RES2"loader.c: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: t30_completion_code_to_str"
00:29.22robin_sznot as good as the book
00:33.04*** join/#asterisk mr-russ (n=admin@CPE-60-224-135-193.vic.bigpond.net.au)
00:33.15tainted-robin_sz it was a book?
00:34.15*** join/#asterisk Johnnie (n=jdlewis@dynamic-acs-24-154-91-195.zoominternet.net)
00:34.17robin_sztainted-, yes, a very famous book
00:34.32robin_sztainted-, won the boardman/tasker prize when it cam out
00:35.24robin_sztainted-, yes, a very famous book
00:35.26robin_sztainted-, won the boardman/tasker prize when it cam out
00:38.35orlokShuld the asterisk registrar also be a peer?
00:44.10*** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com)
00:44.55*** join/#asterisk TUplink (n=Tommy@68-232-82-147.chvlva.adelphia.net)
00:45.25*** join/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com)
00:45.30TUplinki have 2 SIP lines to the PSTN... in my dialplan how can i make it use one or the other... if one is bussy
00:48.51orlokMay  1 10:48:25 NOTICE[3945]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
00:49.49tainted-orlok is the SIP client registered?
00:50.19tainted-robin_sz the movie's ending was too rushed
00:51.18robin_sztainted-, read the book, its totally gripping, the movie left me cold, the people who made it were non climbers really, it meant little to me
00:52.16robin_szits one of those books that you end up reading cover to cover in one sitting
00:52.58orloktainted-: sip show registry seems correct
00:53.00drfoomod2is there any xml/soap shim for the manager api?
00:53.25demigod2kwhich book?
00:53.25orloktainted-: should the provider i am registered with show as a sip peer as well?
00:55.31tainted-robin_sz i'll have to check it out
00:55.43tainted-orlok are they your origination/termination provider?
00:55.58tainted-orlok can u pastebin your sip.conf w/o the username/password
00:56.11orloktainted-: yup they are
00:56.39orlokwtf, i'm getting a 400: Bad Request in response to a NOTIFY sip:
00:56.59*** join/#asterisk techman97_andy (n=me@70-98-31-249.dsl1.rsm.mn.frontiernet.net)
00:57.23*** join/#asterisk PMantis (n=pmantis@cpe-66-66-115-197.rochester.res.rr.com)
00:57.46techman97_andyhey all - I need to get off of my current VoIP provider, VoiceEclipse, like yesterday.  Their reliability is shoddy at best.  Any good suggestions for a SIP provider?
00:58.04PMantistechman97_andy, Location ?
00:58.13techman97_andy(there should be no reason that I have to have people calling my number wait 15 seconds before it is delivered to my * box)
00:58.18techman97_andyMN - USA
00:58.57orloktainted-: http://pastebin.ca/52463
00:59.58PMantistechman97_andy, For DIDs, you can look at: connect.voicepulse.com, BroadVoice, ViaTalk
01:00.11tainted-drfoomod2 what platform are u using?
01:00.13PMantistechman97_andy, BV and VT also offer good OB dialing plans
01:00.33tainted-broadvoice eats babies
01:00.37techman97_andywow
01:00.38techman97_andyhehehe
01:00.38tainted-i have a support ticket for that too
01:00.45PMantislol
01:01.02PMantisI've had BV problems, but attibuted that to latency.
01:01.05xachenViaTalk is too buddy
01:01.06tainted-<Broadvoice> we have been known to eat the occassionaly baby
01:01.14xachen:P
01:01.28xachener, buggy
01:01.39tainted-orlok what does your Dial string look like
01:02.41PMantistainted-, xachen, ok... then from my list, that leaves voicepulse ... which charges by the minute for all OB dialing.
01:02.52tekatitainted: Tell me another VOIP provider besides Broadvoice that gives you unlimited inbound and outbound calling to the US even for a flat rate of 20.00 or close to it.
01:03.17tekatiI agree their service leaves a bit to be desired at times.
01:03.52tainted-1) 78.02 to set up an acct
01:03.52PMantistekati, and 4 simultaneous calls
01:04.03tainted-2) unlimited not really unlimited - softcap at 1000
01:04.44tainted-3) little asterisk support, weird downtime, random config changes that break your box, no support for anything but inband ulaw
01:05.02tainted-4) customer service by special olympians
01:05.14PMantisheh
01:06.03tekatiWho is that?  Broadvoice with Asterisk did not charge me that to start up.  They do not softcap me at 1000 I had over 3250 minutes last month.  Very weird asterisk support with weird downtimes and random configuration changes I will give you that.  LOL on the Customer Service I agree there.
01:06.43tainted-did u sign up for byod?
01:06.52tekatiIf they could get their stuff together they would have a good service.
01:06.58tekatiYes I did sign up for the BYOD.
01:07.03tainted-try consistently pushing 3250 every month
01:07.24tekatiHmm interesting let me see what my average a month is.
01:07.39PMantisI'm doing between 2 and 2.5k /month with BV
01:07.58tainted-wow must be a new policy then
01:08.07tainted-all US?
01:08.18tekatiYes. for me anyway.
01:08.34PMantisYes, mostly within the same areacode
01:09.35PMantisBut my BIGGEST complaint with them is the fact that the closest DID is located 20 Miles away, and long distance for my neighbors... second biggest probelm is the 10+ seconds is sometimes takes to connect.
01:09.54tekatiI did 3100 last month.  3250 the month before that. 3225 the month before that. 1339 the month before that. 1439 the month before that. 2245 the month before that.  So consistantly over 1000 for sure.
01:10.20tekatiI guess I am lucky they have local numbers for me here.
01:10.31tekatiActually they were the only ones I could find that actually do.
01:10.41tainted-tekati where?
01:10.48*** join/#asterisk Ridgeback (n=jircii@180.246.8.67.cfl.res.rr.com)
01:11.03tainted-what are u doing to push 3250/min/month
01:11.05tekatiBakersfield, CA
01:11.23tekatiConference calls unfortunatly.  I work from home.
01:11.34tainted-20.00 is their residential plan
01:11.40tainted-but you're doing business on it
01:11.42tekatiYep this is my house.
01:11.47*** join/#asterisk anthm (n=anthm@CPE-69-76-83-52.wi.res.rr.com)
01:11.47*** mode/#asterisk [+o anthm] by ChanServ
01:11.47tainted-so technically you're cheating them
01:12.00tekatiIts a very gray area.
01:12.01tekati:)
01:12.15tainted-well i would stay with them
01:12.31tainted-b/c the more customer like u that go with BV, the more likely they will go bankrupt
01:12.32tekatiIn my defense I have 2 numbers with them.  one is just a additional number to the main account.
01:13.04tekatiI would not mind paying them 45 if their service got a little better.
01:13.17tekatiIt has got better lately.
01:13.23tekatiBut still not quite there.
01:13.43tekatiI just do not like that I can not send them CID.  I have to use someone else for that.
01:13.46tainted-yea when i had them they'd suspend if i pushed too many mins through
01:14.22tekatiReally.  That is interesting.  The first time they do that we are going to have to exchange some words.  I have been there since 7/2004 though.
01:14.27tekatiSo I hope it does not come to that.
01:14.55tainted-one of the olympians told me that there's an algorithm to detect abuses
01:15.05tainted-predictive dialers and etc
01:15.29tekatiThe reason they probably leave me alone is that I am very rarely on more then 1 call at a time unless the kids are home calling a friend or something.
01:15.46tekatiYea I don't do any of that kind of stuff.
01:15.48tainted-yea don't they offer two channels?
01:15.48tekatiReal usage.
01:16.04tekatiUp to 4 from what PMantis was saying.
01:16.13tainted-i thought that was VP
01:16.20tekatiAh maybe it is.
01:16.22tekatiCould be.
01:16.37tainted-who do u conference call with
01:16.37*** join/#asterisk nain (n=nain@202.59.90.180)
01:16.41nainHi Every body
01:16.45tainted-headquarters?
01:16.57tainted-same folks each time or different
01:17.04PMantistainted-, Perhaps that is the case... but I *know* we've been on at least 2 before, and I *think* we've gone up to 3 simultaneous calls with them.
01:17.14tekatiDifferent people but it is a headquarters conference bridge I do use.
01:17.20nainI would like to patch this file to asterisk for packetization purpose of frame: http://bugs.digium.com/file_download.php?file_id=9746&type=bug
01:17.32tekatiSometimes the same people.
01:17.42tekatiSame number to dial in anyway.
01:17.45nainWhat is the syntex or where to place a file for patch to be work correctly
01:17.55tainted-ahh
01:18.07tainted-why not set up a conference bridge and then everyone can talk for free
01:18.17nainlike can any one guide me how to patch any file ? like "patch -p1 < ...... ?"
01:18.25Strom_Cnain: man patch
01:18.39Ridgebacknain -- copy that web page to a text file and use patch -p.....
01:18.44Strom_CPMantis: I just tested - I can only do two concurrent calls on my BV account
01:18.45Cybertoytainted, try ext 514 in fwd...
01:18.53nainStrom_C: Ok where to place the file and how to execute the patch
01:19.12tekatiStrom_C does that include if you do three way calling etc?
01:19.20Ridgebacknain in the same directory as the original source file
01:19.31tekatinain: patch -p1 < patchfile
01:19.44Strom_Ctekati: "three-way calling" is just setting up another SIP call and letting the phone switch back and forth
01:20.02naintekati: I did the same but patch required different file to patch in different directory of asterisk and it ask me for path
01:20.02Strom_Ci'm using this with asterisk; dont know about using an ATA
01:20.28tekatinain: patch -p1 filetopatch < /somedir/somepatch
01:20.33nainHowever i have provided path but some not all of files or Hunked are successfully patched
01:20.42PMantisDoes anyone know if a SIP phone in a call center can allow an agent to be set "Not Ready" ?
01:20.59tekatinain: sounds like the patch is out of date.
01:21.26RidgebackPMantis. perhaps by using the "presence" functionality?
01:21.33nainrtp.packetization-2006-03-30.patch [^] (13,344 bytes) 03-29-06 14:40
01:21.57nainTekati: isnn't it, could u plz check it if will it work for current release or not ?
01:22.08anthmcute that was my patch =D
01:22.28tekatinain: sounds like you go the author of the patch to talk to :)
01:22.41PMantisRidgeback, Hmmm, suggest a SIP phone that can do that?
01:23.00PMantisRidgeback, Any know of any way to report on how long an agent was "present" ?
01:23.13tekatiStrom_C: You appear to be correct.  2 calls maximum at a time.  Did not even know that.
01:23.19tekatiLearn something new everyday.
01:23.27RidgebackPMantis, yep! Ploycom IP600. I have one of them. Within a local Asterisk network, the phones can be set for anynumber of "statuses
01:23.29Ridgeback"
01:23.57Qwellanthm: You can help him fix it then. ;)
01:23.59RidgebackPMantis, the only problem is, it cannot send SIP prescense data across IAX or SIP to other asterisk switch, that I know of...
01:24.23tekatiI know the answer to this already but no one by chance has the ADMIN GUIDE for a PAP2 do they?  I have the PAP2-NA and need to know what some of the configuration options are.
01:24.54anthmi wrote it in september
01:25.02anthmso now it's kinda out of my hands
01:25.23tekatinain: that is going to be the problem then the patch is way out of date.
01:26.08anthmhttp://bugs.digium.com/view.php?id=5162
01:26.10tekatiThat patch date is 3/30/06 though so it should not be that out of date.
01:26.16Ridgebacktime for bed.... later guys.
01:26.30nainaha
01:26.38*** part/#asterisk Ridgeback (n=jircii@180.246.8.67.cfl.res.rr.com)
01:27.01naintekati: it's not too old patch it should work
01:29.00tekatinain:  Any changes that have been commited since 3/30 can cause that patch to fail.  anthm would be your best bet if he does not want to do anything you will have to go through it yourself and try to correct the imbalance.
01:29.39tekatiPayPal him a few bucks.  That might get him to work on it for ya.
01:31.20tekatiAnyone want to trade a TDM40B for a decent PolyCom or Cisco 79XX phone?
01:31.31tekatiWithin reason that is.
01:31.43QwellI'll give you a grandstream
01:31.58Strom_Ccan I give you the 7960 that has a broken base?
01:32.03tekatiHad many offers for Grandstreams.
01:32.21Strom_Cwait, I already have two TDM400 cards, what the hell do I need a third for?
01:32.27tekatiLOL
01:32.37fileyou should send me one
01:32.42file:D
01:32.51Strom_Cfile: well one is in my stable PBX and one is in my dev box
01:33.12Strom_Coh Qwell, can you link me to your patch again?
01:33.23Qwellteam/north/chan_skinny-fixup
01:33.43nainwell patch is another case: Actually i am getting problem of overhead of IP, with g729 or any other codec that is 4 time to codec bandwidth required
01:34.04nainlike 8kbps required by g729 but it is consuming upto 30kbps
01:34.11*** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net)
01:34.21nainCan any one have idea how to resolve or reduce overheads ?
01:35.17Qwelliax2 trunking
01:35.23Strom_Cor just go to ulaw so you have proportionally less overhead ;)
01:36.01nainStrom_C: I can't afford ulaw ....
01:36.07Strom_Cit was a joke
01:36.28nain:)
01:37.01nainQwell: iax2 trunking will only reduce few kbits like maximum 3 to 4 and it's not enough compare to 8kbps
01:37.43PMantis90k vs 13k packet size helps some...
01:38.05*** join/#asterisk ramo (n=ramo@59.92.137.242)
01:38.07nainlike 8kbps for g729 and it's going to almost 28kbps up and 28kbps down so it's too much for bandwidth
01:38.29Strom_Cnain: what kind of connection are you on where 28kbps is too much?
01:38.54PMantis19.2kbps modem ;)
01:40.01PMantisOr he's running a torrent on his OC-48. LOL
01:40.10orlokcool, hello-world works
01:40.11*** join/#asterisk techie (n=gus@antibala.com)
01:40.12orloktheres a step
01:40.35nainStrom_C: I have 512Kbps connection for 10 channels and it seems not to be good enough.
01:40.49tainted-nain what codec
01:40.55Strom_Cnain: whats your latency like to your host?
01:40.57naintainted: g79
01:40.59naing729
01:41.08tainted-yea ping your provider
01:41.21tainted-could be a crappy router too
01:41.28Strom_Cnain: I routinely run six or seven ulaw calls concurrently on a 608kbps uplink using ulaw with no quality problems
01:41.34nainlatency is not more then 230ms
01:41.47tainted-nain that's high
01:41.48Strom_C230ms is almost pushing it
01:42.02Strom_Ci think my voip proxies are like 60ms away
01:42.02nainThis is the maximum
01:42.12tainted-who cares about maximum
01:42.14tainted-what is avg
01:42.34anthmyah he's probably in pk where that is normal
01:42.51nainanthm: u right
01:42.52anthmwhich is why he knows he needs to maximize the ratio
01:42.55anthm=D
01:42.58tainted-ah
01:43.16tainted-how far is your provider in relation to u
01:43.21PMantisAnyone here build an inbound call support center with multiple queues that can give a couple pointers?
01:43.28nainProvider is in US
01:43.32anthm20ms g729 is only 3 bytes bigger that 1 rtp packet header
01:44.02tainted-nain put a box at a good colo and point your users to the colo
01:44.16tainted-then trunk to the us provider from the colo
01:44.16anthmso you can see why one would want to put more data per packet
01:44.37*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
01:45.29orlokhey
01:45.36orlokthe url for the book is missing from the topic
01:45.42Strom_C~thebook
01:45.46jbotfrom memory, thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
01:45.46naintainted: well, i have my server colocated in well known place and terminated to US, The only thing is that call is orignating from pk and that's why delay is must
01:46.43anthmPMantis run while you still can
01:47.28nainStrom_C: bandwidth issue is on orignation side, Here in Pk bandwidth is too expensive and for 10 channels 512kbps is too much and even with 512kbps i am not getting good quality
01:47.45Strom_Cnain: are you getting good quality with just one channel?
01:48.00nainSrom_C: right,
01:48.20nainStrom_C: voice is fine for even 5 or 6 channels
01:48.46Strom_Cusing IAX or SIP?
01:49.53nainI tried both SIP and IAX, IAX take about 3.2KB up and 3.2KB down per channel, While SIP take almost 3.8KB up and down
01:50.08Strom_Cnain: what packet size?
01:50.15nainso 1 channel take almost 8KB overall
01:50.43nainStrom_C: 2 Frame per packet
01:50.47tainted-nain are u transcoding?
01:50.57Strom_Chow many ms per packet?
01:51.27tainted-could be cpu
01:51.39nainSTrom_C: 20ms default
01:53.02nainStrom_C: Call is orignating from SIP Dialer like Eyebeam and it won't allow to make changes in packet per frame etc.... neither asterisk
01:53.27*** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
01:54.10infinity1i'm trying to get a backup service working for outbound calls (voxee) and i'm getting this error which i can't resolve. SIP/2.0 404 Not Found
01:54.29infinity1anyone have a clue for me?
01:54.45Strom_Cinfinity1: you have a typo somewhere, methinks
01:54.56infinity1err
01:54.58orlokinfinity1: when i had that issue, i tcpdumped the connection from * to the voip provider, checked that, turns out it was a password error
01:55.25orlokinfinity1: are you getting that from the phone -> asterisk, or asterisk -> sip provider?
01:55.36infinity1asterisk-> sip provider
01:55.44orloktypo/password error i'd say
01:55.51orlokresource not found
01:56.02nainStrom_C: any clue
01:56.41Strom_Cnain: try using IAX trunking, if possible...if not, use a bigger pipe or another box on another pipe
01:57.03infinity1looking...
01:57.33PMantisCan someone recommend a phone that has: one-button status changes, voicemail lights, speed-dial buttons, LCD for CallerID, etc.
01:57.45orlokPMantis: most of them?
01:57.50Strom_Chaha
01:57.53*** part/#asterisk Isaiah (n=Isaiah@208-187-93-4.br1.hnv.mi.frontiernet.net)
01:57.55orlokthough notsure about the one button status changes
01:58.10PMantisorlok, I have little to no experience with SIP hardphones
01:58.12orlokPMantis: sipura does all that i think,
01:58.13orlokahh
01:58.34orloksipura/linksys/cisco would be a good bet then
01:58.35infinity1sheesh . can't find a type
01:58.52orlok<PROTECTED>
01:58.52infinity1er typo. i set it up with eyebeam and it worked first trt
01:58.54infinity1er rty
01:59.02infinity1try
01:59.34infinity1sorry. just broke my thumb. i'm still getting use to typing with a cast
02:00.23PMantisorlok, What SIP phones do you have experience with?
02:02.51*** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net)
02:03.31infinity1i'm out of ideas. what else is out there besides voipjet and voxee?
02:03.45Strom_Cthere's asterlink ;)
02:03.50Strom_Cnufone
02:03.54Strom_Cvoicepulse connect
02:03.59Strom_Cwhat else
02:06.02*** join/#asterisk ReD-MaN (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net)
02:09.46nainStrom_C: Well i have tried another box as well and even i tested it between very low latency server but problem is same
02:12.24orlokPMantis: Cisco, Grandstream, Linksys
02:12.45orlokPMantis: Grandstream cheap+nasty feeling, Cisco's are $$$, Linksys are a nice balane between them
02:13.02orlokonly thing cisco has that linksys doesnt is a minibrowser
02:13.18orlokgot one of each on my desk currently
02:13.59PMantisorlok, I was just looking at some Linksys phones, D-Link phones.
02:15.22PMantisorlok, What do you think of the Cisco CP-7912G
02:16.33PMantisOr snom phones ?
02:18.42[TK]D-FenderSnom = Flakey, Linksys=overpriced, and no speed-dial/presence support, 7912 = way overpriced and sup-functional
02:18.44orlokhavent used snom or dlink
02:18.59orlokyeah, i'd avoid the cisco's :)
02:19.23PMantisok, I'm hearing a Linksys recommendation
02:19.50VoicePulsePMantis: The Linksys phones also have Sipura firmware with is a huge plus in terms of provisioning/features.
02:20.21VoicePulsePMantis: Although the importance of that depends on your deployment size.
02:20.42PMantisVoicePulse, Going to be about 20 Agents to start...
02:20.55PMantisI'm looking at http://www.voiplink.com/Linksys_Voice_Over_IP_Phones_s/35.htm trying to decide on a recommendation.
02:21.54VoicePulsePMantis: Don't use the 841s though.  I believe they are based on older hardware that wasn't as refined as the new 941/942s.
02:21.59PMantisDon't care to find the cheapest vendor (yet), just looking for a side-by-side comparison to find the cheapest phone that supports what's needed.
02:24.08PMantis...and since I've never setup a SIP phone, let alone in a call center, I'm not sure what's needed yet.
02:24.42PMantisI've only used a carded system so far... so dual RJ45's are a must for QOS.
02:28.06*** join/#asterisk DoktorGreg (n=Greg@70.91.121.89)
02:28.11PMantisCool. Linksys SPA 922 supports syslogging
02:29.45DoktorGregfunny sip nat problem
02:30.04DoktorGregcalling in always works fine
02:30.09DoktorGreghowever
02:30.28DoktorGregon the first phonecall out, cant hear on sip phone in question
02:30.47DoktorGregafter that all is fine for about 15 minutes
02:30.56DoktorGregthen first phone call out cant hear
02:31.03DoktorGregthen all is fine again
02:31.22DoktorGregis that my nat device?
02:33.01DoktorGregok question
02:33.05DoktorGregrather reality check
02:33.19DoktorGregI just replaced my ups with a bigger ups
02:33.24PMantisSniff the packets to find out for sure... its the SIP communication that dictates the RTP setup.
02:34.02DoktorGregwith the new ups the lcd screens seem brighter
02:34.11DoktorGregand the computer seems to be running faster...
02:34.20DoktorGregam i imagining that???
02:34.24*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
02:35.23DoktorGregPMantis, since i doubt you will walk me through sniffing packets
02:35.25DoktorGregon irc
02:35.41DoktorGregcan you point me to a tutorial on sip packet sniffing?
02:35.48*** join/#asterisk L|NUX (n=linux@202.5.145.58)
02:35.55orlokDoktorGreg: ethereal and tcpdump
02:36.35*** join/#asterisk d-tech (n=dtc@72.245.233.107)
02:36.40*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
02:36.50PMantisUnfortunately, I don't know of a tutorial.. I just felt my way through the packet headers a couple times.
02:37.44ManxPowerThe SIP RFCs will help you understand the contents of the SIP packets, but really, the RFC is poorly written and confusing
02:39.21DoktorGregwhat phones, becaues im having problem with spa 1001
02:42.30[TK]D-FenderPMantis : Linksys is missing quite a few things.  First, 2 lines, no presence, flimsier feel than Polycom / Cisco, poor usability of LCD.
02:42.56[TK]D-Fenderlinksys does have a backlit screen on the 922 though.
02:43.37[TK]D-FenderI'd rate the Polycom IP 501 as a much better purchase for that range
02:43.59PMantis[TK]D-Fender, Oh really? I was looking at the 942. I need presence (like an agent setting themselves as not avail for queue calls, but still taking direct extension calls).
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02:44.10[TK]D-FenderSorry, meant to type 942 there :)
02:44.20PMantisheh
02:44.36[TK]D-FenderPMantis : Linksys doesn't do presence.  Only SIP-B which * does not support, and only across 2 lines.
02:44.44*** join/#asterisk fabiokl (n=fabio@200.175.4.169.adsl.gvt.net.br)
02:44.58[TK]D-FenderWhic means you can only monitor *1* if and when * actually gets around to supporting it anyways
02:45.10PMantishah
02:45.19PMantisok, that's not good.
02:45.25[TK]D-FenderWithout paying to activate the other "lines
02:45.30[TK]D-FenderNo its not....
02:45.32*** part/#asterisk fabiokl (n=fabio@200.175.4.169.adsl.gvt.net.br)
02:45.36PMantisANy "gotcha's" with Polycom 501 ?
02:45.51[TK]D-FenderIf the price point was lower it'd have a place in my suggestion scale.  Just not yet.
02:45.54tainted-price
02:46.49[TK]D-FenderPMantis : Yes. You have an initial choice on the 501 for PoE included or wall-wart.  switching will cost a little extra.  You should also get it from a cert'd dealer to ensure support.
02:46.57[TK]D-FenderPolycom IP 501 = $170
02:46.57PMantis[TK]D-Fender, I'm not sure I understand your sentence. That mean you don't have a Polycomm phone ?
02:47.24[TK]D-FenderPMantis : I own every model of Polycom desk IP phone.  I've also owned an SPA-941
02:47.39PMantisok
02:47.43[TK]D-FenderI ditched my 941 to financeits Polycom replacements.
02:47.52[TK]D-FenderAnd glad of it.
02:47.56PMantislol
02:48.17[TK]D-FenderI run all 600's at work, and a 501 + 301 at home now.
02:48.46PMantisOk, what's the difference between the Polycom phones, and will the differences matter at all for an inbound tech cupport call center?
02:49.11[TK]D-FenderDon't get me wrong, the SPA-941/2 are decent phones, just that they don't fit in the right price bracket and are more suited to home hackers
02:49.52[TK]D-FenderPMantis : For inbound typicall agents will only be on 1 call at a time.  They also typically won't be speakerphone users and likely on headset.
02:49.57PMantisWhat I'd *really* like to see is a single button agent logoff (perhaps logon, but what about roaming agents?)
02:50.07PMantis[TK]D-Fender, Agreed
02:50.15[TK]D-FenderPMantis : That in mind I'd strongly suggest the IP 301 + Plantronics M12 amp + headset
02:50.56[TK]D-FenderPMantis : Polycom has some specialized agent functionality that is planned to be merged into 1.4 due this summer.
02:51.03[TK]D-FenderFor login/out/pause
02:51.19PMantisOOoooooooooooh
02:51.25[TK]D-FenderPMantis : Naturally as its only supported in a patch I haven't tested it personally.
02:51.35PMantisand the 301 supports all an agent would need ?
02:52.00DoktorGregok, have ethereal installed and working
02:52.04[TK]D-FenderPMantis : All the typical SIP stuff (hold, blind/att xfer, conference), and so on
02:52.10DoktorGregnow how do i spy on my ata adaptor?
02:52.30PMantisI guess it would help me to see a side-by-side comparison of features on the phones.
02:52.56PMantis[TK]D-Fender, Here's a provider that's selling a 301 w/ a headset..
02:52.57PMantishttp://www.voipsupply.com/product_info.php?&products_id=1021
02:53.04[TK]D-FenderPMantis : Special note : RJ12 headset jack so you WILL want to get them a Plantroncs (GN Netcom, etc) amp + headset.  It makes all the difference VS el-cheapo 2.55mm ones
02:53.37*** join/#asterisk ptiggerdine (n=ptiggerd@203-206-88-247.dyn.iinet.net.au)
02:53.59PMantis[TK]D-Fender, You saying that after opening the link I sent?
02:54.02[TK]D-FenderPMantis : How big a call center, and what kind of cll volume?
02:54.32[TK]D-FenderPMantis : No, I was typing before your line came over
02:55.02PMantis[TK]D-Fender, brand new call center, hasn't opened doors yet. Will be inbound technical support (paid). They're expecting about 15-20 agents to start.
02:55.13[TK]D-FenderI'm looking at that link now.... headset looks cheap and I don't see it AMP'd... straight does not measue up if you're in a potentially noisy environment
02:56.05PMantisDo you know what this means? "Handset operation through independent jack"
02:56.20[TK]D-FenderI don't know the brand of that headset so I distrust it by nature, plus no mention of AMP.
02:56.20PMantisLOL
02:56.51[TK]D-FenderPMantis : that means the headset isn't used in-line with the handset, it has its own RJ12 jack.
02:57.03Strom_CRJ9
02:57.11PMantisheh
02:57.15Strom_CRJ12 is six-position four-conductor
02:57.17[TK]D-FenderOthers use the RJ12 handset to wire in and you need to leave the handset off-hook.
02:57.19PMantisRJ11/12 is the wall jack size
02:57.32Strom_CRJ9 is four-position four-conductor
02:57.42[TK]D-FenderStrom_C : RJ12 = mini handset connector.
02:58.02Strom_Csays who?
02:58.13DoktorGregoh man for headsets use softphone and usb headset
02:58.26PMantisha
02:58.45PMantisNot when a client says they want hard phones. :)
02:58.51[TK]D-FenderStrom_C : Last I checked.  could be mistaken.  Either way the headset jack is the same style as the handset jack :)
02:58.56DoktorGregzactly
02:59.26DoktorGregi just let people in my office try the usb headsets
02:59.35DoktorGregand they said they wanted the better isolation
02:59.38Strom_C[TK]D-Fender: yes, they're the same.  but the 4-position 4-conductor plug is RJ9.
02:59.48Strom_CRJ-11 and up are six-position
02:59.53DoktorGregbut we have noisy office
02:59.57[TK]D-FenderStrom_C : I'll want to brush up on it to be sure.
03:00.18Strom_CRJ-11 == 2-conductor, RJ12 == 4-conductor, RJ-14 == 6-conductor
03:00.27DoktorGregnext im gonna map a both a hard phone and a soft phone to particular desks
03:00.28Strom_C(all six-position)
03:00.45DoktorGregso they can use either seamlessly
03:01.09MikeJ[Laptop]rj smar-j
03:01.13[TK]D-FenderStrom_C : Ummm. RJ is the jack type, not jsut the # of conductors....RJ11 = 3 pair, typically 1-2 used
03:01.23Strom_Cgah, NO NO NO NO NO.
03:02.03DoktorGregcan anybody tell me how to spy on my ata with ethereal?
03:02.39filegive me wireless any day.
03:02.42[TK]D-FenderStrom_C : I'll jsut assume that my numbering is wrong OK?
03:02.49MikeJ[Laptop]http://en.wikipedia.org/wiki/RJ-11
03:02.52file[TK]D-Fender: Strom is obsessive compulsive
03:03.01MikeJ[Laptop]and wrong I think
03:03.31PMantisHeh, All I know is that RJ11/12 is what's in my wall that I plug y deskphones into.
03:03.32*** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk)
03:04.11MikeJ[Laptop]yeah.. rj11 is 1 pair, 14 is 2, 12/25 is 3
03:04.13[TK]D-FenderAh whatever!  Not something I'd really want to argue over right ro wrong.
03:04.25Strom_CI think I found the requisite document, but it'll take a few minutes to download from the TIA
03:04.42PMantisHeh, if it works.. no discusion needed.
03:05.03PMantis[TK]D-Fender, So whatever the RJ # is, it's a headset connector....
03:06.54PMantis[TK]D-Fender, Thanks for taking the time to go over phones. I think I'll be suggesting the IP-301's.
03:06.59[TK]D-FenderPMantis : Yeah, that :)
03:07.45[TK]D-FenderPMantis : I have tried a plantronics headset that came with an adapter to it that plugs direct to the phone without an amp, but it sounded weak.  Unless your reps are well isolated I wouldn't really suggest it.
03:08.03PMantis[TK]D-Fender, so 1.4 will directly support polycom status updates? Is that officially written anywhere?
03:08.08[TK]D-FenderPMantis : If you follow my lead you'll find you'll actually spend more on headset tech than the phones themselves :)
03:08.44PMantisYeah, I've seen the prices of good headsets...
03:08.45[TK]D-FenderPMantis : It's planned.  What will actually happen is anyones guess, but its in the mailing lists.  Look up "polycom ACD"
03:09.09coppicePMantis: no. you've seen the prices of "professional" headset :-)
03:09.09[TK]D-FenderPMantis : And they're worth it.  I didn't believe so at first... but sometimes you jsut have to do a job right.....
03:09.20*** join/#asterisk kainam (n=Jake@202.137.160.110)
03:10.04coppicethe prices of things like plantronics are totally unreasonable, but they are tough enough to stand life in a call-centre
03:10.24PMantisGN-Netcoms are good, too....
03:10.42[TK]D-FenderPMantis : Both are good... GN is only nominally cheaper
03:10.46coppiceand Hello Voice. those 3 pretty much are the market
03:11.43[TK]D-Fendercoppice : Hadn't heard of them...
03:12.10[TK]D-Fendercoppice : Of course I learned what I had to to equip my team when their manager started whining and might have sooner than later :)
03:12.36coppicethey are kinda weird, because their of their supply chain. you have to mail order them or something. a lot of call centres use them, though, with good results
03:13.49MikeJ[Laptop]coppice, you dropped your unicall!
03:13.55MikeJ[Laptop]over there --->>
03:13.59fileoh god
03:14.04MikeJ[Laptop]:P
03:14.14[TK]D-Fendercoppice : I only find the in-your-face advertised stuff and read up.... anything "behind the scenes" is out of my scope.
03:14.15filein my highly hyper caffeinated state, I laughed out loud at that
03:14.23MikeJ[Laptop]it's late
03:14.33PMantishehe
03:14.34fileyeah, it's late...
03:14.44[TK]D-Fenderfile : no more Red Bull for you!
03:14.52PMantisIt is here, too... but *somewhere* it's early.
03:14.57fileI don't drink red bull, I tried it when I was at Digium HQ and it gave me a horrible headache
03:15.03xachenwhen electricity pulses through my body I laugh for no reason. does that count? :P
03:15.13MikeJ[Laptop]russellb, so autoconf done yet?
03:15.29russellb?
03:15.34russellbyes, autoconf is complete.
03:15.36PMantisfile, red bull sucks... XS Energy is *tons* better
03:15.41russellbtook me 30 years to write it
03:15.42MikeJ[Laptop]LIAR!
03:15.55coppicePCP is more effective
03:16.02MikeJ[Laptop]hehe
03:16.14fileI like Bawls
03:16.24coppiceso does my wife
03:16.31MikeJ[Laptop]ummmmmmm
03:16.35PMantisROFL
03:16.35MikeJ[Laptop]wow....
03:16.48filethat's hot
03:16.58ManxPower#Asterisk: After Dark
03:17.06fileafter hours
03:17.18PMantisI've seen #ltsp do that, but not #asterisk
03:17.33PMantisof course, I'm in #ltsp a TON more
03:19.13MikeJ[Laptop]file, what has Corydon done to you!!
03:19.20fileno no
03:19.28fileit's "what has Corydon tried to do to you!!"
03:19.35russellbeverything?
03:19.36MikeJ[Laptop]phew.....
03:19.42MikeJ[Laptop]I was worried for a second
03:19.50MikeJ[Laptop]so you had mace?
03:20.07russellbhe has the whole being really far away thing on his side
03:20.11fileif by mace you mean muffins
03:20.12filethen yes
03:20.29MikeJ[Laptop]were they old muffins with spikes coming out of them?
03:20.36fileyes
03:20.47MikeJ[Laptop]well that works
03:20.58filebaked right in even
03:21.45MikeJ[Laptop]so who did the scary kp as uncle sam picutre
03:21.48MikeJ[Laptop]picture
03:21.57fileMikeJ[Laptop]: are YOU on the DTMF task force?
03:22.03russellbi haven't seen it
03:22.10MikeJ[Laptop]no, the picture scared me away
03:22.17MikeJ[Laptop]:P
03:22.19filemakes sense
03:22.34orlokhey, in the asterisk book, they are using a Zaptel card for outgoing
03:22.39orlokspecified as Zap/4
03:22.40MikeJ[Laptop]but I am a solid supporter of proper dtmf support
03:22.50MikeJ[Laptop]orlok, one sec
03:22.50orlokwhat would i do for a sip registrar?
03:22.58fileah yes, solid DTMF support
03:23.08*** join/#asterisk jsmith (n=jsmith@smithfam.dsl.xmission.com)
03:23.19russellbfile: where's that pic
03:23.31MikeJ[Laptop]orlok, oh look.. theres jsmith.. I think he wrote that book
03:23.37russellblol
03:23.42*** join/#asterisk trbldwine (n=trbldwin@71.194.161.170)
03:23.45russellbjsmith: run!
03:23.55jsmithDocumentation?  On Asterisk?  Get out...
03:23.57MikeJ[Laptop]russellb, and he fell for it
03:23.59coppiceif a VoIP platform can't get something basic liek DTMF right, there doesn't seem a lot of hope for it :-)
03:24.18jsmithcoppice: You should know ... DTMF is overrated
03:24.28fileI said we should just get rid of DTMF support altogether
03:24.34MikeJ[Laptop]coppice, TROLL!
03:24.36russellbfile: i second that
03:24.38MikeJ[Laptop]:P
03:24.38filesomething can't be broken if it doesn't exist
03:24.46Strom_Cno no no no, then how will I decode what people are dialing just by listening?
03:25.02MikeJ[Laptop]orlok, SIP/
03:25.03jsmithfile: But we need to keep support for the ABCD digits in DTMF... otherwise, only the government will have control of them
03:25.16orlokMikeJ[Laptop]: SIP/username?
03:25.17coppicefile: so if a bridge falls down becaus a bolt is missing, rather than broken, its OK? :-)
03:25.23filecoppice: yes
03:25.36filecause like - who needs bridges
03:25.40filewe have legs... we can jump!
03:25.44MikeJ[Laptop]well... SIP/somthing
03:25.53MikeJ[Laptop]it could be a friend!
03:25.57MikeJ[Laptop]or an ip
03:26.11*** join/#asterisk trbldwine (i=trbldwin@71.194.161.170)
03:26.13MikeJ[Laptop]or a dns name
03:26.19MikeJ[Laptop]file, what else can you use
03:26.21coppicefile: I'll consider that in an hour or so when crossing the worlds largest road+rail bridge :-)
03:26.30filecoppice: excellent
03:26.38fileMikeJ[Laptop]: or... a peer
03:26.43MikeJ[Laptop]user?
03:26.47filecan't call a user
03:27.06MikeJ[Laptop]why not :P
03:27.11jsmithUsers, Peers, and Friends, oh my!
03:27.26filebecuz Uncle File says so
03:27.37MikeJ[Laptop]jsmith, be my friend?
03:27.37fileand you know better then to disobey me!
03:27.48MikeJ[Laptop]what else could you SIP/ to?
03:28.16coppiceisn't VoIP wonderful. "how will we convey DTMF across channels which corrupt the voice". "why, every possible way anyone can think of, of course" :-)
03:29.12MikeJ[Laptop]ok.. here is a fun game.. how many sip dtmf methods can you name
03:29.24filespecific to SIP?
03:29.27coppicedumb and dumber?
03:29.28russellbinband, RFC2833, INFO, NOTIFY
03:29.31MikeJ[Laptop]inband, 2833, notify, info,
03:29.34MikeJ[Laptop]theres more...
03:29.34russellbi win
03:29.36MikeJ[Laptop]ummmm
03:29.38jsmithMikeJ[Laptop]: More?
03:29.39fileit's a trick question
03:29.41MikeJ[Laptop]yeah
03:29.53fileinband isn't a SIP DTMF method, neither is RFC2833
03:29.56MikeJ[Laptop]file, what are the others
03:30.01russellbfile: i know that.
03:30.01MikeJ[Laptop]blah....
03:30.06MikeJ[Laptop]you knew what I mean
03:30.07fileI just know the 4
03:30.09russellbfile: but it's still used in conjunction with SIP, you goof
03:30.10MikeJ[Laptop]there are at least 2 more
03:30.29coppicewhy are they not SIP methods? SIP says the media will use RTP
03:30.31filehow many other ways could you convey it...
03:30.50filecoppice: I was picking on his use of words
03:31.04orlokhmm
03:31.06fileyou could encode it via XML and send it along
03:31.15orlokwhats the dissalow= for in sip.conf?
03:31.15russellbfile: that's notify!
03:31.23filerussellb: do they really do it via XML?
03:31.25orlokdisallow, even
03:31.29russellbfile: i believe so, yes.
03:31.32filedear god
03:31.36russellbexactly.
03:31.55russellbit's quite a fsck up, if you ask me.
03:32.20russellboh noes!
03:32.29jsmithorlok: That's to disallow certain codecs from being used
03:32.59russellbfile: at least there is only ***ONE*** way to send DTMF with IAX2
03:33.26filewell
03:33.32russellbone correct way.
03:33.35*** join/#asterisk Ixthod (n=Ixthod@198.174.206.41)
03:33.39fileyes
03:33.46filethat chinese chip supports inband though...
03:33.53russellbwell that's just stupid
03:34.04orlokjsmith: ahh, so that stops all by default, then i've got the allow ulaw and alaw
03:34.05orlokcool
03:34.15russellbfile: i hope asterisk refuses to support it
03:34.29filewe don't fire up a DSP on the channel so it never gets recognized internally
03:34.31russellbfile: "crappy IAX2 implementation detected, immediately closing connection"
03:34.47fileINVAL! INVAL! INVAL!
03:35.01russellbyou're speaking in IAX2!
03:35.05fileACK
03:35.15russellbREGREQ
03:35.22fileAUTHREQ
03:35.31russellbAUTHREP
03:35.32russellb:/
03:35.42russellbi haven't looked at it in quite a while
03:35.53fileI'm focusing on something else...
03:36.04MikeJ[Laptop]his muffin
03:36.05russellbi am ... too ...
03:36.07filebut it's
03:36.40fileREGREQ, REGAUTH, REGREQ, REGACK
03:36.40PMantisCan * send a text message to a Polycom IP-301 for things like # callers in queue, # agents available, etc?
03:37.04PMantis...for display on the LCD, of course.
03:37.09jsmithPMantis: Good question... I have no clue
03:37.25PMantisheh, but at least you admit it.
03:37.35filePMantis: theoretically yes
03:37.42orlokjsmith: uhh, is it bad if i just printed out the whole PDF? ;0
03:37.45fileyou'd just have to code it
03:37.57jsmithorlok: Nope... that's what it's there for.
03:38.12MikeJ[Laptop]I just found somthing funny: "Cisco Unity treats any incoming RTP payload above 90 as a DTMF event.
03:38.12MikeJ[Laptop]"
03:38.20MikeJ[Laptop]heheh
03:38.21fileMikeJ[Laptop]: seriously?
03:38.26*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-241-16.red.bezeqint.net)
03:38.34jsmithMikeJ[Laptop]: You've got to be kidding
03:38.42MikeJ[Laptop]http://doc.trecom.tomsk.su/cisco/cc/td/doc/product/voice/c_unity/whitpapr/sipcomp.htm#1047608
03:39.04coppiceorlok: yes, its bad. you shouldn't consume wood to make paper. you should consume it in a furnace making electricity :-)
03:39.09fileha
03:39.28orlokheh
03:39.34orlokdont let our clients hear that :)
03:39.42orlokwe are in the printing industry
03:39.56jsmithMikeJ[Laptop]:
03:40.07jsmithMikeJ[Laptop]: That' looks like an old doc -- 2002?
03:40.29jsmithMikeJ[Laptop]: That's before Cisco even acknowledged that SIP existed in the real world
03:41.05filehello world!
03:41.09PMantisfile, jsmith, others, How does an agent normally monitor a queue, to see how many people are waiting, etc?
03:41.30r0d3nt|m!dlrow olleh
03:41.36orlokexten => _0XXXXXXXXX,1,agi(selintra,OutRoute,Outgoing)
03:41.42MikeJ[Laptop]jsmith, maybe.. still scary
03:41.51orloki should have more entries in my config file for OutRoute and Outgoing, correct?
03:42.16mds2anyone know if you can change the default dialtone on Cisco 79xx phones?
03:42.21*** join/#asterisk mog_home (n=achika54@68.62.237.103)
03:43.36jsmithPMantis: Using an app that talks to Asterisk via the Manager Interface
03:43.46jsmithmds2: Yes, slightly, see dialplan.xml
03:43.54PMantisjsmith, Recommend one?
03:44.08PMantisjsmith, For a Linux desktop...
03:44.25mds2jsmith: thanks, have been playing with that.  I'm curious about changing the default primary dial, busy and reorder tones to something more.. local
03:44.49jsmithmds2: Good luck :-(
03:44.53mds2jsmith: from what I've read dialplan.xml will let you change secondary dialtone only. can it do more than that?
03:45.05jsmithPMantis: Nope... most are custom apps
03:45.07mds2hum
03:45.12jsmithmds2: Not that I know of
03:45.19mds2righto, thanks anyway
03:46.19MikeJ[Laptop]file, found another: http://www.ietf.org/internet-drafts/draft-ietf-sipping-kpml-07.txt
03:46.40mog_homefile!
03:46.51filemog!
03:47.02jsmithMikeJ[Laptop]: Using SUBSCRIBE, huh... that's new...
03:48.53*** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com)
03:49.00filelet's all just pretend that doesn't exist, mmmk?
03:49.03MikeJ[Laptop]oh wait... then there is 2833 in info.. forgot about that one
03:49.25websaefile: back when you did work for asterlink, how long did it take to get a 800 number without it being rushed?
03:49.53xachenPorted or Vanity?
03:50.03websaevanity
03:50.32*** join/#asterisk bmg505 (n=leon@c1-177-3.rndf.isadsl.co.za)
03:50.55xachen7-14 days
03:51.07orlokDo i have the syntax for this right?
03:51.07websaedo you work for asterlink?
03:51.10orlokexten => 123,1,Dial(SIP/0385121550,0438015327,1)
03:51.28xachenyes
03:51.31filehahaha
03:51.38MikeJ[Laptop]there are lots...
03:51.43filechan_iax2.c:7700 socket_process: Immediately destroying 666, having received INVAL
03:51.49file666!
03:51.59MikeJ[Laptop]yes
03:52.10orlokchannel of the beast
03:52.23coppiceDefining a KPML that can't replace MIDI is stupid :-)
03:52.46jsmithcoppice: chan_midi.so?
03:52.46brookshiredoes sip keep getting more and more bloated everyday or what?
03:52.56jsmithbrookshire: No kidding...
03:52.58filebrookshire: just like... er nevermind
03:53.11brookshirefile!!!!! !! !!!!!!!!!!!!!!!!!!!
03:53.11orlokhmm
03:53.14brookshire!!!!!!!!!!!!!!!!!!!!!! !!
03:53.15brookshire!
03:53.17brookshire!!
03:53.20orlokwhen i dial 123, i get a message "Caller ID is Blocked"
03:53.24brookshireoh.. and !!!! !
03:53.57coppicebrookshire: SIP was developed by people who thought telephony didn't have to be as complex as H.323, while H.323 was developed by people who had spent years finding it does.
03:54.14MikeJ[Laptop]ok.. to some up, dtmf methods: inband, 2833, info, 2833 in info, megaco based event detection in INFO, NOTIFY, and kpml..
03:54.36brookshirewhy does there need to be 90 billion ways to detect dtmf too?
03:54.42coppiceif you look at SIP and H.323 today, which looks cleaner?
03:54.54jsmithcoppice: None of the above...
03:55.10MikeJ[Laptop]coppice, of the 2?
03:55.11coppicejsmith: yeah, it a close call :-)
03:55.17MikeJ[Laptop]h323
03:55.20brookshiresip happens
03:55.22MikeJ[Laptop]hands down
03:55.22brookshirethough
03:56.01MikeJ[Laptop]number of lines of rfc is always a fun one to compare
03:56.14*** join/#asterisk downunder33 (n=downunde@219.95.158.235)
03:56.31jsmithWhy'd you guys talk me into hanging out in here anyway?
03:56.39coppicecomparing spec lines says nothing really.
03:56.42filewasn't me.
03:56.46mog_home#asterisk is the coolest jsmith...
03:56.57jsmithYeah, if I were smart like y'all
03:57.11coppicethe real fun specs leave out most of the detail (e.g. T.38). of course, that limits their length
03:57.40jsmithExactly... specs are what people use to distract suits from the details
03:58.01fileI like a protocol that I can just change on a whim...
03:58.01coppicespec == fly in my soup
03:58.25MikeJ[Laptop]file, heh
03:58.40MikeJ[Laptop]DO IT
03:58.41filethat nooooobody has to be compatible with!
03:58.45filebecause it's MINE!
03:58.50jsmithchan_file.so
03:58.53fileand that protocol would be, FTP2!
03:58.54coppicefile: XML is ideal for that. a language with no semantics :-)
03:58.55MikeJ[Laptop]so you can talk to yourself
03:58.58fileFile Transfer Protocol... 2!
03:59.10*** join/#asterisk redondos_ (n=redondos@190.48.46.149)
03:59.20coppiceis FTP2 anything like MSFTP?
03:59.28MikeJ[Laptop]are you saying file has no semantics?
03:59.29filecomplete opposite
03:59.42MikeJ[Laptop]so youll need to make it IAX3
03:59.42fileFTP2 encodes everything in pig latin
03:59.44MikeJ[Laptop]errrr
03:59.46MikeJ[Laptop]FTP3
04:00.10brookshireoh why oh why is myspace sooo slow
04:00.16PMantislol
04:00.22russellboh why oh why are you using myspace?
04:00.32brookshirebecause myspace > #asterisk
04:00.39MikeJ[Laptop]ummmm
04:00.42MikeJ[Laptop]your lame
04:00.44MikeJ[Laptop]:P
04:00.45fileMikeJ[Laptop]: I refuse to make IAX3 ... yet
04:00.46brookshirehaha
04:00.48*** kick/#asterisk [brookshire!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (off to myspace, then!)
04:00.54*** join/#asterisk brookshire (i=mbrooks@hijacked.us)
04:00.59brookshirerussell: boo!
04:00.59russellb:-p
04:01.07filerussellb: you're supposed to be asleep
04:01.13russellbfile: oh yeah ...
04:01.16brookshiredon't make me drive to the carolinas
04:01.23russellbbrookshire: you wouldn't
04:01.28brookshireyou're right
04:01.30coppicecompatibility is for wimps. real mean reverse engineer all protocols on the fly
04:01.51filecoppice: you rock! :P
04:01.53brookshireactually.. i should convience some people to take a road trip
04:02.25brookshireconvince also
04:02.33coppicei saw a road trip one. a volvo with a blowout did 3 sommersauts
04:03.00russellbthat's not cool.
04:03.12brookshirei've seen a wreck worse than that :(
04:03.17coppicei was amazes to find a 2 tone car could do that
04:03.19russellbwell let's not talk about it!
04:03.38russellbor i guess you can if you want, i'm going to sleep now :-p
04:03.44fileyay
04:03.47filebye bye russell!
04:03.48orloki'm suprised a volvo had the velocity to flip three times
04:03.51coppicethis wasn't a bad wreck. it was a volvo - they all walked away :-)
04:03.57brookshireyay! i agree with file!
04:04.16fileyay
04:04.47coppiceorlok: you obviously haven't driven a high end volvo. they aren't slow. heck, they made The Saint's car :-)
04:05.47orlokcoppice: farkin hemi moite
04:06.59jsmithorlok: I don't know what you're trying to do with this syntax, but it's wrong: exten => 123,1,Dial(SIP/0385121550,0438015327,1)
04:07.15orlokjsmith: thought it might be
04:07.21MikeJ[Laptop]once upon a time there was the little protocol that could... and it got bigger, and bigger, and bigger,  and it became SIP!
04:07.26orlokjsmith: i've got two sip channels set up i want to use for outbound dialling
04:07.49orloki was trying to set it up so 123 would dial 0438015327 via the 0385121550 sip account
04:08.09orlokjsmith: asterisk book is great, but the fact that the start focuses on zaptel stuff is annoying
04:08.13orlokbut i didn tpay, so meh :)
04:08.19fileMikeJ[Laptop]: it growed up?!?
04:08.28MikeJ[Laptop]yes
04:08.30jsmithSIP/0385121550/0438015327 or SIP/0438015327@ 0385121550
04:08.32filehow cute
04:08.34MikeJ[Laptop]well.. it's a teenager
04:08.35orlokahh, cool
04:08.42fileentered puberty yet?
04:08.47MikeJ[Laptop]use the #
04:08.49coppicewhy would anyone sane want to filter entered DTMF at source? you can only have 10 digits per second. Its hardly going to be an issue to send anything that is entered
04:09.03MikeJ[Laptop]who filters
04:09.04MikeJ[Laptop]??
04:09.28coppicefile: well, its certainly already been f**ked up a lot
04:09.38filecoppice: very true
04:10.14*** join/#asterisk esculapio_ (n=ESCulapi@145stb68.codetel.net.do)
04:10.14MikeJ[Laptop]ererr @
04:10.45esculapio_tecnico, hola esta hay
04:10.45MikeJ[Laptop]naptime, or finish the coding I need to do?
04:10.48jsmithorlok: The next edition of the book will have more SIP coverage
04:11.28jsmithorlok: At the time it was written, the SIP support in Asterisk was... well... let's just say it wasn't my favorite channel type.
04:11.39esculapio_websae, no soy dominicano
04:11.43orlokheh
04:11.49websaeque pasa?
04:12.00websaeno hablas ingleis?
04:12.06orlokjsmith: we sell high quality broadband, and our broadband supplier is owned by one of the arger pabx (an dslam) manufacturers
04:12.29orlokjsmith: so  we have high quality, low latency broadband, plus sip/voip services provided to us by the same company
04:12.29esculapio_websae, no muy poco
04:12.41websaeque necesitas esculapio_?
04:12.46orlokso theres not that much reason to be using ata's
04:13.02orlokhmm.
04:13.09*** join/#asterisk shmxtra (n=shmxtra@219.95.158.235)
04:13.09orloki'm getting "caller id is blocked"
04:13.13esculapio_websae, mi llamadas en mi asterisk no salen ni entran solo suena ocupado
04:13.28esculapio_websae, pero tengo comunicacion interna
04:13.42esculapio_websae, me puedes ayudar
04:13.56websaeyo lo veo
04:14.21websaesolomente hablo espanol un poco, pero yo atare ayudarte
04:14.23orlokwtf does "caller id is blocked" mean
04:14.26Strom_Cpor favor, habla ingles aqui o usa los mensajes privitos
04:14.31orlokupstream doesnt like my cid or lack of it?
04:14.39orlokdonde esta la pollo
04:14.46orlokme no hablo espanol
04:14.58Strom_Ces como afeitando con un gato en tu grabadora
04:15.00websaeStrom_C: porque no hablas espanol aqui?
04:15.26Strom_Cwebsae: because most in here don't speak spanish and it's mainly spam and clutter for the majority of users
04:15.40websaeStrom_C: necesitamos ayudarlos que necesitan ayudar
04:15.55fileit looks like random characters put together to me... ha
04:15.56Strom_Cwebsae: use private messages in that case
04:16.07jsmithesculapio_: Tal vez puede encontrar mas ayuda en #asterisk-es
04:16.19Strom_Cor that
04:16.30MrDigitalno espanol hablo engles
04:16.56jsmithStrom_C: Neither did I, until about 30 seconds ago
04:17.04Strom_Chah
04:17.17asterboyhablo un poco
04:17.24coppiceah, maybe this KPML isn't so dumb. it allows things like "A # of at least 3 seconds" to be defined. that means effective digit length related input methods can be made useful in limited contexts.
04:18.28websaeWell I am glad we can segregat in here :)
04:18.31websaethat's good
04:18.38MikeJ[Laptop]what's goto line number in emacs?
04:18.52websae*segregate
04:18.55jsmithMikeJ[Laptop]: CTRL-ALT-OpenApple-Meta-ALSKDFJA
04:19.00fileEsc+X goto-line
04:19.10coppiceMikeJ: something just as obscure as in vi, of course
04:19.24jsmithcoppice: Naw, it's easier in vi
04:19.28MikeJ[Laptop]thank you
04:19.40*** part/#asterisk downunder33 (n=downunde@219.95.158.235)
04:20.05coppiceI love these vi and emacs wars. the participants seem blissfully unaware they suck equally :-)
04:20.10MikeJ[Laptop]not working
04:20.21filehit Esc, then X, then type in goto-line and hit enter
04:20.23coppiceyou didn't add pixie dust
04:20.25filethen enter the line number and hit enter
04:20.56tainted-coppice how dare u reveal the truth
04:21.05sevardi can't stand emacs at all and vi hobbels along, i'd much rather use pico or nano.
04:21.12fileI don't care what people use for a text editor... use what works
04:21.25MikeJ[Laptop]not working :(
04:21.30Strom_Csevard: but but but vi is the lubeless anal rape you grow to love!
04:21.32sevardi use pico and nano because i don't want to do fancy fucking things with my editor, i just want to edit a file.
04:21.48fileMikeJ[Laptop]: :(
04:21.56MikeJ[Laptop]just says no match
04:22.00tainted-u text editor trolls
04:22.01sevardStrom_C: That's a whole other ballgame.
04:22.03filefor goto-line?
04:22.05coppicevi is far more useful than emacs, only because its the editor that's always there. however "its always there" is also the only thing Windows has going for it
04:22.20Corydon76-homeEmacs is a nice operating system, but it lacks a decent editor.
04:22.29filethe hate, the hate!
04:22.36MikeJ[Laptop]I hit escape and x, it says M-x 123
04:22.37sevardCorydon76-home: i've heard that one, it's great.
04:22.39filenext you'll be telling me Linux sucks
04:22.43MikeJ[Laptop]I hit enter, it says no match
04:22.51fileMikeJ[Laptop]: clear the 123
04:22.56fileand type in goto-line
04:23.07coppiceI wish someone would add key stroke recording to SciTE.
04:23.22Corydon76-homeI tend to threaten Emacs users with making it their shell.
04:23.25sevardI use pico and nano becaues software should be easy to run.  I don't want to open a emacs or vi reference book that weighs more than a small child to find the command to jump to a line.  i just want to edit a fucking file.
04:23.27MikeJ[Laptop]oh that's dumb
04:23.44QwellCorydon76-home: threaten them with making it their editor
04:24.51coppicethe sentence for copyright infringement should be something like entering the bible from a printed copy by emacs.
04:25.05Corydon76-homeI would use emacs, as it was the first editor on Unix I was ever taught.  However, I prefer having enough memory left over that my kernel doesn't need to be swapped out of memory.
04:25.39rdgztCorydon-w: So, how's life with a computer with 16 megs of mem these days?
04:25.39PMantisfile, Unanymous vote?
04:25.46filedarn right
04:25.55coppiceslipping into bed sounds very keystone cops
04:26.09filecoppice: wha?
04:26.28PMantisheh, speled with an 'i'... I knew that didn't look right.
04:27.03fileyay bed
04:27.08coppicewhen emacs was written, 16 megs was a luxury very few had
04:29.29sevard16 megs still is.
04:29.57sevardOMFG HI
04:31.05*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:34.51*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
04:35.05*** join/#asterisk melte (n=melisate@219.95.158.235)
04:35.17brookshireHIHIHIH!
04:37.08coppicesevard: yeah. if you want anything as small as 16M, its a specialist product at a high price :-)
04:37.11MikeJ[Laptop]kram, how many sip dtmf methods can YOU name!
04:37.12*** part/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com)
04:39.07*** join/#asterisk melt1 (n=melisa@219.95.158.235)
04:41.33*** part/#asterisk melt1 (n=melisa@219.95.158.235)
04:41.36coppiceMikeJ: do combinations count? like one digit by RC2833, the next by NOTIFY, etc? :-)
04:42.37MikeJ[Laptop]heh
04:43.14MikeJ[Laptop]I wonder how bad you could confuse a UA by sending overlapping 2833 digits
04:43.56MikeJ[Laptop]ok.. sleepy time for me..
04:44.24coppicehave you seen how easy it is to confuse most ATAs? hard lockup seems their usual way to get pissed off :-)
04:47.03Corydon76-homeThey must not be coded in ADA...
04:47.51coppicethat's right. they absolutely must not under any circumstances be coded in ADA
04:57.49*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
04:58.40*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
05:00.36*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
05:09.14websaeit always gets quiet right around now
05:09.28Zeeekshhhhhhh!
05:09.30websaeI suppose everyone needs a nap once in awhile
05:09.38Zeeekkeep it down!
05:09.47websaeI'll keep
05:09.48websaemy
05:09.48Zeeekor get it up, if that's your thing ;)
05:09.50websaesentences
05:09.51websaeshort
05:10.08websaelol
05:10.15websaehow's it going Zeeek?
05:10.17websaewhere you from?
05:10.38coppicelunch seems just the thing right now
05:10.38ZeeekMinneapolis, MN, USA
05:10.52Zeeeklunch? I just had my first coffee
05:14.11Corydon76-homeSo, is it cold or hot in Mpls?
05:15.12websaeMilwaukee, WI here
05:15.39ZeeekActually I'm in Paris now. But my son says it never gets cold in the winter anymore
05:16.05websaeit's about 50ish here
05:16.07websaein Wisconsin
05:16.16websaeZeeek, what are you doing in Paris?
05:16.33ZeeekI've lived here for the last 25 years working on asterisk :)
05:16.36CunningPikeLast tango
05:16.43CunningPikeSorry - couldn't resist
05:16.45*** join/#asterisk wenko (n=wenko@142.232.8.200)
05:16.46websaewe just started offering DIDs from Paris, France
05:16.55websaeoooo
05:16.59ZeeekI invented asterisk and gave the idea to Mark in 1980
05:17.08websaehaha
05:17.22Zeeekoh, what's your Paris offer?
05:17.31ZeeekI have a few already (testing)
05:17.52ZeeekI have an interesting logistic problem I'm working on now
05:17.56websae$13 USD/month DID and unlimited
05:18.11ZeeekYou want to kknow what the competition does?
05:18.19websaethey blow it away :)
05:18.26Zeeeknaw, not really
05:18.32websaegood thing I don't compete with them in the U.S.
05:18.32coppicewebsae: 50ish? wow, that's desert temperature :-)
05:18.36ZeeekWengo €7 unlimited
05:18.44websaecoppice, where are you from?
05:19.31coppicefrom? from about 10,000km from here :-)
05:20.45websaeoh
05:20.47websaefantastic
05:21.17websae12:21AM Monday, May 01, 2006
05:21.26Zeeekthis is like ham radio all over again
05:21.43websaethat it is
05:21.46coppice01:21PM Monday, May 01, 2006
05:21.59websaeyou're only a timezone away hehehe
05:22.09websaewell quite a few timezones away
05:22.11websaeare you in AU?
05:22.20CardoeSo what's the advantage of AEL over the conf files?
05:22.28coppicenope. .au would be 03:21PM
05:22.31Cardoeare the conf files going away? should I go with AEL?
05:22.34Zeeekyou can pretend you're programming in C
05:23.03coppicelots of people pretend they are programming in C
05:23.05websaeand be really cool then
05:23.14jsmithCardoe: No, the conf files aren't going away
05:23.29ZeeekI never pretended to program in C. I kludge a lot though
05:25.22Cardoeso basically don't bother with ael
05:26.50jsmithCardoe: You have your choice on whether to create your dialplan in AEL or the old style.  Internally, AEL gets translated back to the old style, so it's no going away.
05:26.57jsmithCardoe: In short, it's about choice.
05:27.57coppicetime for lunch, then off to disneyland. turn our noses up an disneyland, and play on the lake next door to it :-)
05:28.10*** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
05:29.55websaeI want lunch
05:34.13Zeeekhey kram, randy here
05:35.05*** join/#asterisk Ciber311 (n=Ciber@user-1087e94.cable.mindspring.com)
05:35.59Strom_Cnothing makes your mind work hard quite like speaking a language you learned in school but never spoke conversationally in the first place
05:38.00Zeeekwhat mind? What is work? What is school?
05:38.32coppiceStrom_C actually, that just makes my mind switch off
05:38.54Strom_Chaha
05:38.57Strom_Cyeah, I know what you mean
05:39.01Strom_Cyou forget simple things
05:39.08Zeeekthe first time I actually had a conversation in a foreign language, I was nearly drunk
05:39.17Zeeekit flowed better that way
05:39.33websaeZeeek: they always do
05:39.48websaeEven when you speak your native language as well sometimes
05:40.03Zeeekin fact I had a very funny experience related to that night that I think I shall regale you all with
05:40.37coppicethe first time I ever needed a foreign language in earnest, it was one my school would never have considered teaching
05:41.07ZeeekMy "business" colleague, a technician, accepted the late night hotel clerk's offer of finding "a girl" (at like 2AM)
05:41.40ZeeekWe were sitting in the bar and she shows up and sits down next to me and says "so what's your sign?"
05:42.05ZeeekI had to say "Oh it wasn't me that called, it was my friend"
05:42.48ZeeekShe: "so um, what do we do?" Me: I'm going to get up and you'll take my seat next to him." And that's how my French career started
05:53.06*** join/#asterisk naS_- (n=andrew@182.136.233.220.exetel.com.au)
05:53.25*** join/#asterisk codebreaker (n=codebrea@xserver.flexserv.de)
05:56.07*** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim)
05:56.08rdgztTypical reason for ztcfg saying no such device or address?
05:56.26Zeeekdriver not loaded
05:56.33rdgztUsing a TDM400P with 4 FXOs.
05:56.40rdgztThat should be the tor2 driver, right?
05:56.47wasimwctdm
05:56.53Zeeekwasim, my man
05:56.59wasimtor2 is the old isa card
05:57.05wasimbonjour monsieur Zeeek
05:57.10Zeeekcomment va?
05:57.14wasimbien
05:57.45rdgztOk, modprobing wctdm didn't give any error messages, but neither did it seem to change anything.
05:57.57wasimztcfg -vvvvvvvvvvvv
05:58.41rdgztTells me the channel map with four channels, that look ok, and then says
05:58.42rdgzt4 channels configured.
05:58.42rdgztZT_CHANCONFIG failed on channel 1: No such device or address (6)
05:59.02wasimwhat does dmesg show
05:59.26rdgztAm I looking for anything specific here?
05:59.52rdgzt[   45.812423] Zapata Telephony Interface Registered on major 196
05:59.53rdgzt[   45.812510] Zaptel Version: 1.2.5 Echo Canceller: KB1
05:59.53rdgzt[   46.839809] Registered Tormenta2 PCI
05:59.53rdgzt[   47.130273] usbcore: registered new driver wcusb
05:59.53rdgzt[   47.130344] Wildcard USB FXS Interface driver registered
06:00.02wasimand your zaptel.conf should have fxsks=1-4
06:00.02L|NUXwasim : bro i need your little help :)
06:00.03rdgztIs the only thing Zaptel-related I can see.
06:00.09rdgztwasim: That it does.
06:00.11wasimL|NUX: what happened
06:00.36L|NUXwasim : bro see i have senrio i hope you can understand me better :)
06:01.19L|NUXwasim : suppose if some one call me on extension e.g i have this in my extensions.conf exten => 1,1,Dial(SIP/1,20,tr)
06:01.55L|NUXwasim : when its time out then it will Goto new context [email] and execute systemcommand :)
06:02.26wasimL|NUX: only if 1,2,Goto(email|exten|prio)
06:02.31L|NUXi tried
06:02.35L|NUXbut in system
06:02.55L|NUXwait
06:03.44wasimrdgzt: you have neither usb nor tor2
06:03.49L|NUXi am using this in [email] context
06:03.49L|NUXexten => s,1,SetVar(CALLFILENAME=/var/spool/asterisk/monitor/${TIMESTAMP})
06:03.49L|NUXexten => s,2,System(/bin/echo -e "'Incoming call from : ${CALLERID} \\r Received: ${DATETIME}'" | /bin/mail -s "'Phone call'" f4fahmed@gmail.com)
06:03.54L|NUXand its not working :(
06:04.13rdgztSo that still doesn't work.
06:04.15rdgztThat's really weird.
06:05.45L|NUXany idea wasi
06:05.48L|NUXwasim bro
06:06.07wasimL|NUX: build the command a single step at a time
06:07.05L|NUXwell this thing is working
06:07.05L|NUXlike
06:07.17L|NUXif put it in [sip] context like this
06:07.21L|NUXi remove s
06:07.24L|NUXand add only 1
06:07.27L|NUXand its working
06:07.51wasimofcourse, 1,2 != s,1
06:08.18L|NUXO_o
06:08.32L|NUXso i will have to use s,3,setVar....
06:09.38rdgztModules seem to be loaded correctly, but ztcfg still tells me the device doesn't exist.
06:09.41rdgztAny other ideas?
06:09.53wasimL|NUX: no, 1,2 ... where is CALLFILENAME being used?
06:10.40L|NUXhumm
06:10.50wasim[4294681.327000] Zapata Telephony Interface Registered on major 196
06:10.50wasim[4294681.351000] ACPI: PCI Interrupt 0000:06:02.0[A] -> GSI 18 (level, low) -> IRQ 18
06:10.53wasim[4294681.351000] Freshmaker version: 71
06:10.56wasim[4294681.352000] Freshmaker passed register test
06:10.58wasim[4294682.052000] Module 0: Installed -- AUTO FXO (FCC mode)
06:11.02wasimrdgzt: you should get 0,1,2,3 moudles like that
06:11.03L|NUXso its means i have to logged call file in [sip] context
06:11.35*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
06:11.37rdgztwasim: That's weird, so what does it mean if I don't get that?
06:11.49wasimbusted card, bad PCI slot, lots of possibilities
06:12.20rdgztNah, this card worked earlier today, but I uninstalled zaptel and asterisk stuff that I had gotten from Ubuntu packages, and decided to build from source, to solve another problem.
06:12.28rdgztSo I'm pretty sure the hardware's fine.
06:12.47rdgztI haven't changed anything hw-related.
06:12.52wasimok, did you make install
06:12.55rdgztYes.
06:13.21wasimand lspci shows 06:03.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
06:13.37rdgztYup.
06:13.54wasim<PROTECTED>
06:13.54wasimzaptel                187140  59 wctdm
06:13.58rdgztWell, it says "communication controller", strictly speaking, but apart from that, yes.
06:14.30rdgztIt seems it loaded all the modules on boot, actually, for some reason, so yes, zaptel, wctdm, and many more.
06:15.23rdgztrmmodding all the modules and then just modprobing zaptel and wctdm and leaving out all the others makes no difference.
06:18.43*** join/#asterisk Assid (n=assid@203.115.64.12)
06:19.04Assidyello
06:21.40codebreakerif my $database-server is offline. will asterisk then cache the cdr-logs until this server is back online or will they got lost?
06:21.40Zeeekexit
06:21.43Zeeeksu
06:21.46Zeeekwhen4fsre6
06:21.56ZeeekOh, nooooooo
06:22.17codebreakerrooted :)
06:22.26Assidhahh
06:23.14jqlwell, it'll be on google by tomorrow
06:23.19jqlchop, chop
06:26.23*** join/#asterisk BugKham (n=BugKham@125.24.6.174)
06:27.24Assidanyone managed to get their sipbroker working
06:27.30*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
06:31.16Assidisnt that message supposed to come when someone messages him ?
06:32.20wasimAssid: no, it won't cache the logs for the db
06:32.57Assidhuh?
06:33.06Assidno no.. i was talking about krams' away
06:35.08wasimoh sorry, s/Assid/codebreaker
06:36.01Assidyou ever used sipbroker?
06:41.47*** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
06:44.15Assidseriously.. no one?
06:46.53tainted-Assid everyone's asleep
06:46.59tainted-try again in a few hours
06:47.34*** join/#asterisk k-man (n=jason@unaffiliated/k-man)
06:47.38k-manhello
06:48.00codebreakerwasim: thanks. so its better to have a local postgres running and feed my "important" master via fetch :) then i will nly loose logs when the local db is down. but htne i shutdown the hole node.
06:48.12Zeeek"I'm sorry, your answer is unavailable at this time. Please try again later" Dahp. Dahp. Dahp.
06:48.19k-manhow does one ensure availability of ADSL so that you don't have phone lines going down when using voip?
06:48.36tainted-k-man are u having downtime?
06:48.42k-manno
06:48.47k-manwell
06:48.48tainted-Zeeek why are u getting that?
06:48.50Zeeekk-man we have two ADSL lines
06:48.51k-mannot yet
06:48.58k-mani want to get some voip lines
06:49.11tainted-this must be 2 adsl line night
06:49.14k-manso i want to upgrade our ADSL to ensure it never goes down
06:49.15Zeeekreally
06:49.23tainted-u can't
06:49.24k-mancurrently we get downtime every so often
06:49.26k-mannot sure why
06:49.34Zeeekwell SDSL is a step in that direction
06:49.40tainted-what sort of downtime
06:49.42k-manoh?
06:49.45k-mansounds expensive
06:49.47tainted-power outage
06:49.53k-manno
06:49.56Zeeekyou need a "professional" contract. Way more expensive
06:50.05tainted-lol
06:50.10k-mani just know that about 1-2 times a week our network goes down
06:50.17tainted-sounds like a secret handshake
06:50.18k-mani have no idea if its isp or hardware related
06:50.20Zeeeknetwork or connection?
06:50.25codebreakerk-man: but this wont help if your adsl-hardware-providing place goes down. you have two providers but they both use the same hardware
06:50.26tainted-k-man how far from central office are u
06:50.45k-manum.... 1-2kms
06:50.55tainted-what country
06:51.01k-manaustralia
06:51.14Zeeekkoalas on the line
06:52.04Zeeekactually it depends on the cause, codebreaker. When our line goes off, the other is still on
06:52.22ZeeekI chose to put astrisk on the one that doesn't go down often
06:52.33Zeeekironically, it's the cheaper consumer line
06:54.08Zeeekso... nufone...
06:58.06*** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net)
06:59.15k-manok
06:59.37k-manso do isps have some sort of rating in terms of reliability and also, what about modem reliability?
06:59.43k-manare some ADSL modems  cheaper than others?
06:59.49k-mancheaper?
06:59.52k-mani meant better
07:00.03k-manin terms of their reliability
07:00.16wasimbut ofcourse
07:00.18dlynes_k-man: 3com's are good
07:00.29dlynes_k-man: 3com office connect
07:01.35Zeeekhttp://www.dslreports.com/forum/voip
07:01.50Zeeekk-man you know about that site?
07:02.01dlynes_Also, there's a difference in the dsl providers...there's the telcos that don't do anything about all the viruses, spyware, worms, spambots, ... on their networks...then there's other dsl providers that'll shut customers off with that kinda crap abusing their network
07:02.10k-mannothing
07:03.12k-manoh, no, i don't know about it
07:03.38Zeeekit has a lot of consumer comments about all DSL (in the uS)
07:03.44k-manoh
07:03.45k-manok
07:03.48k-manwe hvae something like that here
07:03.53k-manwhirlpool.net.ayu
07:03.57k-man.au
07:04.05Zeeekbut it might have useful info even outside the us. It has speed test for example
07:04.21Zeeekyeah whirlpool is actually pretty good
07:04.58dlynes_yeah...telstra probably blows though
07:05.04dlynes_it's the national telco
07:05.11dlynes_optus probably blows too
07:07.29Zeeekhey check this out: http://www.dslreports.com/shownews/74008
07:07.38Zeeekabout the king of spammers
07:08.58k-manyeah, telstra is crap
07:09.07k-manand they are expensive
07:09.14k-manand have a monopoly on the local loop
07:14.22Assiddamn
07:14.51hads|homek-man: I thought au had unbundled?
07:14.59Assidi dunno what extension should i make to handle this incoming sip uri
07:15.00k-mannafaik
07:15.37Assidif i have a assid@blah.org .... what kinda extension do you need for handlng that?
07:16.18Zeeekhow about exten => assid,1,Dial(yer_phone)
07:16.38Zeeekin a guest or default context
07:17.47Assidi gotta go that for every sip extension ?!?!?
07:17.51Assiderr.. do..
07:17.58orlokk-man: upgrading your dsl doesnt protect against backhoe fade
07:18.09k-manbackhoe fade?
07:18.11k-manwhats that?
07:18.14orlokk-man: oh, you know telstra.
07:18.16orlokan aussie
07:18.24ZeeekAssid no you could just accept all names
07:18.26orlokbackhoe fade.. no sync/dialtone
07:19.02k-manoh
07:19.03orlokk-man: you need redundant connections basically
07:19.09k-manoh
07:19.12k-mani see
07:19.25k-mani was thinking that it might be worth getting a seperate ADSL connection just for voip
07:19.26orlokyeah, preferably of different types,
07:19.33k-manand keeping other traffic off it
07:19.36orloktheres lots of different faults
07:19.37AssidZeeek: so how would that be ? s,1,Dial(s/Local) ?
07:19.44k-manyeah
07:20.14orlokk-man: like, dslam line cards can die, somebody can use the pair for something else, the isp can have an upstream fault
07:20.21ZeeekAssid you could do something like _XXXXX. to keep it a minimum number of characters
07:20.30k-manyeah
07:20.34orlokso you want a different sort of media through another provider, ideally
07:20.37Zeeekotherwise anything at all will ring phones
07:21.01Zeeekbut if you wanted assid@ to ring one phone and farida@ another, you'd need the actual extensions
07:21.03Assiddamn.. my user is spherelinx-satish
07:22.50*** join/#asterisk BoRiS (i=boris@S010600112f38a61e.wp.shawcable.net)
07:23.19Zeeekand?
07:23.46Zeeekguys I had the worst and wierdest problem last year with one of our DSL lines
07:24.10Zeeekit would retain sync but pass no traffic for 10-20 seconds *exactly* evey 8 minutes
07:24.38ZeeekEvery time I made a call I had to tell people if I disappear wait 2à seconds and I'll be back!
07:24.54Zeeekthat went on for 3 weeks before they fixed it
07:25.04ZeeekI'm positive it was in the dslam
07:25.14*** join/#asterisk bzbw (n=wlwzhang@68-190-223-129.dhcp.mtpk.ca.charter.com)
07:27.02BoRiSAnyone has an updated IPP g729 patch that works with the latest cvs changes?
07:27.18AssidZeeek: i have it wanting to go to default context for the calls.. anyway to force it to use another context?
07:32.03AssidZeeek: also.. how would you handle different context (vhost)
07:33.22*** join/#asterisk ramo (n=ramo@59.92.137.242)
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07:50.14ZeeekAssid I wonder what ${EXTEN} contains when it arrives at the _XX. extension? I think I'll try that when I get a second
07:51.17codebreakercan i do something if phone 123 connects tell it to use serverB instead to connect. also something like a redirect?
07:51.54*** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net)
07:52.17key2!seen kram
08:02.44Zeeeksee 'switch'
08:04.20*** join/#asterisk kamileon (n=kamileon@68.62.190.253)
08:04.48AssidZeeek: it takes the extension from before the @
08:04.59Assidor... if you use sipbroker.. its the one after the broker code
08:06.27Assidbrb
08:08.18*** part/#asterisk Dr-Linux (n=Linux@202.59.73.131)
08:20.39Zeeekyes it works fine
08:21.10Zeeekyou'd need to have an address like user_domain@domain.tld
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08:57.20naS_-anyone able to help with g729 pass through trunking using IAX2?
09:04.59*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
09:15.17AssidZeeek: supose i have assid@abc.com and assid@xyz.com .. but i want 2 different phones to ring.. both on the same * box
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09:29.09alexbrhi all
09:30.17alexbris it possible to use asterisk with a standard isdn card?
09:31.13jqlyes
09:31.28alexbrand with an analogic modem?
09:32.40*** join/#asterisk r0d3nt|m (i=r0d3nt@tinfoilhat.net)
09:33.41jqlno clue. it's supposed to work with anything supported by the standard linux isdn drivers
09:33.58alexbrok, thanks a lot
09:34.28alexbrand do you know about any localized documentation?
09:34.57jqlI wouldn't know where to find that
09:35.39*** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no)
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09:46.19*** join/#asterisk ghenry (n=ghenry@195.38.86.72)
09:47.20ghenryfor sip and NAT, I only need to have ports 5060 and 5061 UDP forwarded to a particular client? I can't get it to work any other way
09:48.54*** part/#asterisk alexbr (n=alexbr@adsl-ull-83-239.47-151.net24.it)
09:55.16tainted-what do u mean
09:55.19tainted-was that a question?
10:00.45mutdoes the zaptel echo can control hw echo can on the sangoma cards
10:00.45mut?
10:01.35ghenrytainted-: sorry. that was a question
10:03.59ghenryeverywhere I have read says onlt port UDP 5060
10:04.07ghenrybut it doesn't work without 5061 too
10:04.16ghenryor is that wrong?
10:07.49hwtdoes astertest work with more recent versions (1.2.x) of *?
10:08.09hwtor are there other, perhaps more advanced tools, available?
10:08.41tainted-ghenry it's wrong
10:08.57ghenryI don't understand then
10:09.03tainted-mut yes sangoma has echo contol
10:09.15tainted-hwt what is astertest
10:09.29tainted-ghenry udp works on whatever port u want it to
10:09.30ghenryI have opened up ports 5060 UDP, and could reg, but asterisk couldn't sip_poke
10:09.34mut...
10:09.42ghenryneeded to open up 5061
10:09.44mutyes, does zaptel control the echo cancel on it tho
10:09.45tainted-ghenry u want 5060 TCP
10:09.53mutor is there a seperate sangoma util for it
10:09.59ghenryasterisk can't do TCP for SIP
10:10.00tainted-and a range of ports in UDP for rtp to go through
10:10.09tainted-ghenry wrong
10:10.16tainted-where the fuck are u reading that
10:10.20ghenryso the book and every other docs I read
10:10.30*** join/#asterisk rkr245 (n=ravi@office.callsat-telecom.com)
10:10.36ghenryO'Reilly, voip-info.org etc.
10:10.37tainted-u mean the SIP signalling or the audio
10:10.47mutasterisk only does udp sip..
10:10.52mutfor signalling
10:10.54tainted-u don't want the audio stream in tcp
10:11.08mutand rtp is udp as well
10:11.10ghenryI don't know. That's what I am asking. Newbie ;-)
10:11.11tainted-signalling = tcp, audio = udp
10:11.15ghenryah
10:11.15mutno
10:11.17mutit's udp
10:11.18hwttainted-: http://www.asteriskguru.com/tutorials/astertest.html
10:11.29mutasterisk doesn't do tcp SIP
10:11.35ghenryCouldn't find a Firewall hint sheet for ports etc.
10:11.38ghenryThought so
10:11.39tainted-mut block tcp 5060 and see if u can receive calls
10:11.47mutit is
10:11.50mutand i do
10:11.53hwtmut: it will soon, though.
10:11.58ghenry1.4 will
10:12.00ghenryI read
10:12.07hwtghenry: yup.
10:12.11hwtghenry: to support sip-tls.
10:12.17naS_-does anybody know much about IAX2 trunking?
10:12.18ghenrycool
10:12.28ghenrySo, firewall guide/howto?
10:12.53mutall you should need open is 5060
10:12.55mutudp
10:13.09mutand whatever ports you're using for audio/rtp
10:13.10ghenryIf every client softphone needs port 5060, how can you have more than 1 client behind a NAT router?
10:13.17tainted-where does it say asterisk SIP signalling is 5060 UDP
10:13.27Zeeekyou use 5061 for the second client
10:13.29ghenryNot sure mut. I think sip.conf says 8000
10:13.36ghenryah Zeeek
10:13.41ghenrySo, start again.
10:13.53Zeeekno change to asterisk, just the client
10:14.08Zeeekdomain.name:5061
10:14.13ghenryI can only register my client sooftphone when ports 5060 and 5061 UDP point to the same client
10:14.24ghenrythat's not right
10:14.29Zeeeknot normal
10:14.42ghenrythat's the port it sends the OPTIONS back on
10:15.00Zeeekoh, I just remembered why my laptop IAX wasn't working
10:15.25Zeeek4569 is forwarded - what an idiot
10:16.02tainted-http://www.voip-info.org/tiki-index.php?page=Asterisk+firewall+rules
10:16.07tainted-i stand corrected
10:16.13tainted-5060 udp
10:16.27tzangertainted-: jeez you're up early
10:16.51tainted-tzanger coding the night away
10:17.04tainted-tzanger must be 5-6am where u are
10:17.07tzangeryep
10:17.07muttzanger: you have any idea? sangoma hw echo can controlled by zaptel
10:17.09ghenryThanks tainted-
10:17.10tzanger6:17am
10:17.14mutor is there a sang util
10:17.23tzangermut: I don't think so, I tihnk it's handled by the hwectools
10:17.28ghenryanyone integrated Asterisk with a Avaya VoIP existing PBX?
10:17.52mutknow what they are?
10:18.09tzangerit's a tarball that gets applied ot the sangoma drivers from the sangoma ftp site
10:18.26ghenryhmm tainted- doesn't mention UDP port 5061
10:18.35ghenryso not sure why it doesn't work without that open
10:18.57tainted-ghenry what do u need it to work on?
10:19.07tainted-what clients are u using
10:19.11tainted-softphones, hardphones?
10:19.14*** join/#asterisk kavit (n=kavit@210-84-40-39.dyn.iinet.net.au)
10:19.17ghenryI just testing a basic sip.conf with ekiga
10:19.47ghenryCan only register and make a call when 5061 UDP and 5060 UDP is open
10:21.08tainted-what is ekiga
10:21.21ghenrygnomemeeting renamed
10:21.40ghenryvery nice
10:22.00*** join/#asterisk AsteriskAlbania (n=info@217.24.244.130)
10:22.28AsteriskAlbaniaASTERISK & RADIUS any information will be appriciated
10:23.10tainted-ghenry can u configure ekiga to different audio ports?
10:23.19ghenryaye
10:23.26ghenryanything you like
10:24.19*** join/#asterisk lucifr (n=chatzill@c-24-126-108-87.hsd1.ca.comcast.net)
10:24.26lucifrHello..
10:24.53Zeeekshould be e-geeka !
10:25.07tainted-try setting to something else
10:25.11ghenrytainted-: Ah, no you can't . sorry Might be able to, but can't see where
10:25.12tainted-and then calling
10:25.29ghenryit won't register without UDP 5061 open
10:26.18lucifrI'm trying my new Asterisk installation and I wan to have PSTN termination... Does anyone know who provides it, at least for testing?
10:26.37Zeeeklook on the wiki
10:26.40tainted-it won't register?
10:26.46*** join/#asterisk saftsack (n=saftsack@p54A7F622.dip.t-dialin.net)
10:26.49lucifrmmmmmm....
10:26.54tainted-lucifr what sort of testing
10:27.01Zeeeklucifr in fact you can get 0.25 free account to test voipjet
10:27.07lucifrZeeek, what just I look for?
10:27.12Zeeekthe wiki has a liong list though
10:27.18*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
10:27.20tainted-ghenry how do u know? 'sip show peers'?
10:27.23lucifrZeeek, where?
10:27.26Zeeekproviders
10:27.37Zeeekok wait a second, here take my hand.
10:27.37ghenryhow do I know?
10:27.55Assidhey zeek: how do you set it up such that zeek@abc.com and zeek@xyz.com ring on 2 different sip channels.. using the same * box ?
10:28.10Zeeekhttp://www.voip-info.org/
10:28.29ZeeekAssid I was just reading about that
10:28.30lucifrtainted-, I don't have a phone line here at home, so I would like my friends calling from a regular phone and ring on my computer... Can it be done?
10:28.33ghenrytainted-:  D   N      5061     UNREACHABLE
10:28.49Assidcause when you make exten => zeek,1,Dial(SIP/zeek-abc)
10:28.53Zeeeklucifr this is the last time: GO LOOK AROUND - see the site I wrote above
10:29.01Assidthat would be like universal for all zeeks
10:29.09tainted-lucifr yes, with a soft client like firefly or x-ten
10:29.28ZeeekAsid it has to do with SIP_HEADER(TO) and/or SIPDOMAIN
10:29.34tainted-lucifr u wouldn't need an asterisk server in that case, only a pstn->voip provider
10:29.37ghenrystrange tainted- now it says port 5067
10:29.38lucifrZeeek, I got it...
10:29.55Zeeeklucifr yes in fact you can buy a phone and use a provider
10:30.12tainted-ghenry u have serious firewall issues
10:30.19tainted-put the thing in a dmz and figure it out
10:30.24ghenryyup
10:30.27tainted-then plug it back behind the nat
10:30.31Assid* should actually let you setup this on context based.. so suppose you want abc.com it should be like [domain-abc_com]
10:30.40Zeeeklucifr here is the exact sopt: http://www.voip-info.org/wiki-VOIP+Service+Providers
10:30.44tainted-it's not limiting to 5061, ur just diagnosing incorrectly
10:30.53lucifrwell, I want to do more than just that... I want to play with Asterisk...
10:30.54Assidlike how you do for voicemail
10:31.01Zeeekok, see above
10:31.08ghenrythanks. later all
10:31.14lucifrZeeek, thanks
10:31.40Zeeekor google for voip, voipjet, voicepulse, wengo, voiptalk, sipgate, nufone, junction networks, euriax
10:32.00Zeeeklucifr it depends on what country you want the number in
10:32.05AssidZeeek: but is  it possible to catch everythig and then handle based on the sipdomain.. and thereby throwing it to the context?
10:32.20ZeeekAssid accornding to what I was just reading it is
10:32.28lucifrZeeek, I see
10:32.36Assidyou got the urls' open with you ? i can refer to?
10:32.50Zeeeklooking cause I already forgot about that
10:33.39Zeeekhere's something interesting but doesn't give the answer: http://slacker.com/~nugget/projects/asterisk/page7
10:33.53ZeeekOur very own Nugget
10:38.39ZeeekI can't find that page for some reason
10:39.06Assidweell ${SIPDOMAIN} does exist
10:39.54Zeeekthe actual method I saw was exten => zeeek,1,GoToIf
10:40.41Zeeek$[${SIP_HEADER(TO)}=domain1]]
10:40.54Assidso.. i think what we can do .. is .. exten _.,1,Macro(handledomain,${SIPDOMAIN})
10:40.55Zeeekwith one less ] :)
10:41.08Zeeekthat looks reasonable enough
10:41.18Zeeekor look it up in the astdb
10:41.35Zeeekdomain = family and user = user
10:42.11Assidand then handledomain would have something like s,1,Dial(SIP/${arg1}-user)
10:42.13Assidor something
10:42.45Assidmy astdb knowledge sucks
10:43.44RoyKmethinks astdb should be replaced by an sql solution
10:44.35Assidisnt sip_header for the whole sip uri ?
10:45.17Assidhttp://www.voip-info.org/wiki-Asterisk+variables -- doesnt really document that variable
10:47.08*** join/#asterisk ringe (n=runar@ti531210a080-6380.bb.online.no)
10:50.04AssidZeeek: you tried wengo
10:50.05Assid?
10:52.13Alystairare snom phones really good?
10:52.23Zeeekwengo is pretty good
10:52.28Assidthey are decent Alystair
10:52.48Assid1.0c/min right?
10:53.50RoyKAlystair: imho snom phones are good and expensive, but if you or your customers can afford them, it's very good indeed
10:54.09ringeSay I've got a Via EPIA based Asterisk server. Now I'd like to just connect a USB phone and place a call directly from that. Is it supported?
10:54.39AssidRoyK: just curious.. how would you rate them against polycom?
10:54.46Assidringe: you would need a softphone
10:54.53RoyKAssid: i don't know polycom
10:55.09ZeeekPolCom is very good
10:55.09RoyKringe: there've been talk about chan_usb, but only talk
10:55.29Zeeekplus, it keeps your dentures tight
10:55.56ZeeekAssid where are you
10:56.01Assidhrmm.. always wondered how they compared to one another
10:56.03RoyKringe: http://www.voip-info.org/wiki/view/USB+Phone
10:56.03AssidZeeek: india
10:56.15Zeeekyou want to call france?
10:56.36Zeeekor be called from it ?
10:56.47Zeeekwengo is france
10:56.59Assidme?nah.. i do play with * boxes for my friends.. they call europe often.. but i wouldnt mind a US terminator too
10:57.04Zeeekthere is a new provider called euriax for europe
10:57.19Zeeekeuriax is very cheap
10:57.29Zeeekbut I'm wondering about their routing
10:58.10Ahrimaneshey Zeeek :)
10:58.11ringeRoyK: That's clarifying. So a better approach would be to just connect a hardware phone to the same ethernet. Maybe using a crossed cable to connect directly to eth0.
10:58.14Assidwhat about us-pstn calls?
10:58.19ZeeekAhrimanes - beer?
10:58.30AhrimanesZeeek: nah, not right now, hows life?
10:58.31ZeeekAssid there are a million providers for that
10:58.32Assidringe: definately
10:58.52Assidyeah.. but not all million are good.. and nicely priced
10:58.54RoyKringe: some client on ethernet, it doesn't really matter an ATA, hardphone or softphone
10:59.04Zeeekvoipjet is good for price:quality
10:59.26Assidyeah. didnt have a problem with them.. BUT now.. they need verified paypal accounts
10:59.29Zeeekthe best networks are relativel expensive, retail like 3-4 cents/min
10:59.45Zeeekyeah paypal sucks but it's better than nothing
11:00.07Zeeekyou want to pay 4c/minute?
11:00.13Assidthey dont have a verified account.. and paypal doesnt verify on CC, they need bank account to cverify
11:00.18Assidnah.. 1.5 odd
11:00.40Zeeekretail-wise not many will do less than 2c
11:00.41Assidvoicepulse is 2.4.. im trying to find a replacement for voipjet
11:00.57Zeeekvoicepulse is good but expensive on the penny level
11:01.14Zeeekand what about lag times for India?
11:01.34Assidnot that bad really.. i make quite a few calls.. dont see any reall issues
11:01.42Zeeekusing voipjet?
11:01.45Assidi dont use as much as those guys.. they use it in their office
11:01.51Zeeekwhat's the time in ms usually?
11:01.54Assidme personally? my usage is on sipdiscount
11:02.15Assidfrom my asterisk box.. to me? or.. asterisk box to provider ?
11:02.25Zeeekboth, what the heck
11:02.36ZeeekI show you mine if you show me yours ;)
11:02.59Assidhehe
11:03.07Zeeekjust curious
11:03.17Assidaround 321ms me to asterisk box..
11:03.19ZeeekI'm keeping a chart of all providers , lag and unreachable times
11:03.25ZeeekWHoA!
11:03.33Zeeektalking to the moon!
11:03.56Zeeekwhere's the box?
11:03.59Assidthis box to voicepulse -- 59ms
11:04.00ZeeekJupiter?
11:04.06Assiderr.. mercury
11:04.27Zeeekfunny, my lag to most providers is 50-100ms
11:04.33Zeeek30-40 to my box
11:04.49Zeeekthe French providers are 20-30ms from the box
11:05.04Zeeekso the echo is much shorter :)
11:05.05Assid60 ms.. from this box to voicepulse
11:05.17mutanyone ever used a sangoma a104d? the wac_ec_client command dies when i try to stat it
11:05.37mut<PROTECTED>
11:05.37mut> syntax error (argv=(null),offset=0)
11:05.44Zeeekmy best US providers are around  100ms
11:05.48Assidround-trip min/avg/max = 23.9/24.5/25.8 ms -- from my friends box -- to voicepulse
11:06.05Zeeekwhere is it though, in the US?
11:06.14Assid1 in texas.. 1 in ny
11:06.32Zeeekyeah ok. My box in the US is 3ms away from one of the providers!
11:06.42Assidhis box is in ny.. mine is in datacenter in texas
11:06.48Zeeekgotcha
11:06.59Zeeekmakes sense the 30ms to vpulse then
11:08.41ZeeekAssid that's interesting, those times. I thought I was bad off with 125ms to my second box
11:08.43lucifrDoes any of the provider support more than 1-channel? I mean, I want to be able to setup multiple phones at home and have several people make calls!
11:09.01Zeeeklucifr if you read their sites they tell you
11:09.02kamileonwhere do you guys colocate ?
11:09.03Assid320ms is from my in india to houston /texas
11:09.14ZeeekI understood that Assid
11:09.23Zeeekand mine is Paris-> Virginia
11:09.24lucifrZeeek, yes, I'm going through the list!
11:09.28Assidhosted the box with ev1
11:09.42Zeeeklucifr they usually specify it in the FAQ
11:09.50lucifrok
11:09.57lucifrtnx
11:10.06Zeeekor write them. It's very revealing to call or write before you buy
11:10.20Zeeektry calling cust "care" before you sign up
11:12.17Zeeeklucifr where are you located? That will also determine who you want to go with
11:12.24lucifrZeeek, thanks... you're my heroe!
11:12.27lucifr:)
11:12.42ZeeekI've been thru the shooping experience for two years
11:12.45AssidZeeek: most providers for me is 50-70ms
11:12.52lucifrZeeek, I'm in the West side of the states
11:12.58Zeeekyou mean for you box?
11:13.02Assidyeah
11:13.07Zeeekthen you add the 300ms to you
11:13.11Assidyep
11:13.16Assidbut frankly
11:13.18Assidits nto that bad
11:13.32Zeeeklucifr look up junction ( jnctn.net )
11:13.43nettieHi guys, I noticed that when we receive a call from overseas the + or the double 0 is omitted in the CID
11:13.44Zeeekor voipjet.com
11:13.49lucifrZeeek, thanks
11:13.54Zeeekjust a thought
11:13.58nettieany of your guys know hot to tell asterisk to add it please?
11:14.39Zeeeknettie I believe all that can vary from country to country so there's no perfect way
11:14.59Zeeekunless maybe detecting the lack of a '1' in the beginning
11:15.21AssidZeeek: you connected to sip broker?
11:15.35Zeeekno sir i am not. Never heard of them til today
11:16.13Zeeekis it free?
11:16.17Assidits like a dns based service exchange..
11:16.52jql00 is dropped in some pstn calls as well. it's irritating...
11:16.53Zeeeki'm looking at the site now
11:16.59Assidlike i have black.abc.com .. and when i signup. i get something like *746 for example as a number for my box
11:17.19Assidand if i am on extension 201.. then you just dial *746201
11:17.36Assidit iwll make a sip/uri call to 201@black.abc.com
11:17.48Zeeekok I see the point now
11:18.41Zeeekis there a service that allows you to make a SIP URI call from a site?
11:18.52Zeeekfor testing
11:19.03Assidwell.. it has a test call thing
11:19.07Assidyou cant hear nothing
11:19.12Assidbut atleast you know if its working
11:19.13Zeeekgood enuf
11:19.21Zeeektest the SIP URI?
11:19.24Assidyep
11:19.35Zeeekis it free before I fill out 50 forms?
11:19.57Assidno forms.. user/pass/uri/email
11:19.58Assidthats it
11:20.03Assidand then you get confirmation email
11:20.10Assidto activate
11:20.17Zeeekand pay $100 ?
11:20.20lucifrZeeek, I hope you can help me with this.... At the site you mentioned earlier it says on their website "Up to 25 Simultaneous calls - FREE". Does that mean 25 simultaneous channels open to make phone calls?
11:20.51Assidpay $100 for a free service? you feeling rich ?
11:20.52ZeeekI don't know but I *assume* they mean you are billed for the minutes of *each* call
11:21.07Zeeeklucifr was that junction?
11:21.13lucifryes
11:21.33nettieZeeek well all country doesnt have the + ore the 00
11:21.34ZeeekI think they bill by the MINUTE, so 25 1 second calls would be billed as 25 minutes
11:21.56Zeeeknettie but you see the 00 always? In that case it's easy
11:21.56nettieZeeek if it's france I get 330498784533
11:22.02nettieno
11:22.06nettienever
11:22.10Zeeekok you're screwed then
11:22.26nettieyouthink it's a problem of my carrier?
11:22.33lucifrI wish they (Junction Networks) had unlimited outbound calls!
11:22.37AssidZeeek: that was why i wanted to know how to capture the domain..
11:22.40Zeeekhow about "if you are calling froma  foreign country, please hit the pound key"
11:22.47nettiewhich doesnt send me the correct cid?
11:23.00Assidi got 2-3 vhosts on the same box
11:23.17ZeeekAssid yeah. If I can test mine I'd have the answer
11:23.21nettieZeeek eheh nah, this just to have the correct cid?
11:23.22nettienoway
11:23.23nettieehehe
11:23.49Zeeekyeah I know but I don't see how to determine it because of the variation in number lengths
11:24.14Assidnettie: just consider all the numbers as starting with 1
11:24.16Assiderr
11:24.17Assid+
11:24.25nettieno
11:24.28nettiethat's not good
11:24.32Assidso.. if you get from 3304.......  the number is +3304
11:24.42nettiebecause cells are 338-4567898
11:24.53nettieso it wont work
11:25.03Assidyes.. and when a cell calls you.. it will be 13384567898
11:25.14nettienah
11:25.18nettieit's paing
11:25.19nettieehehe
11:25.25Assidsure it is
11:25.26Assidtry it
11:25.46nettieI maybe try to ask my carrier to send me the proper cid for int number and see what they say me
11:26.10Assidokay one more development change and then i can keep this aside
11:26.20Assidi wish i had more time.. i wanna play with that sip uri
11:26.27Assidunfortunately.. i got work to do
11:26.38Zeeekso how do I test the number Assid?
11:26.41AssidZeeek: http://www.astmasters.net/howtos.html incase you wanna read
11:26.53ZeeekI saw that one but I don't think it was the best
11:27.17AssidZeeek: right side. there is a ezdial
11:27.23Assidelse call *75493001
11:27.34Assidprovided you set up your dial plan and sip already
11:27.45Assidthats for outgoing
11:27.54Assiddid you get a sip broker number?
11:28.11Assidhttp://faq.sipbroker.com/tiki-index.php?page=Asterisk+Configuration <--
11:30.35Assidi cant seem to add a dyndns.org account
11:36.41Zeeekyes
11:36.48Zeeekim on the phone now
11:37.25*** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz)
11:39.32Assidworks?
11:40.13Zeeekno it was my wife calling on DID from New York
11:40.21Assidoh
11:40.23Zeeekexcellent quality call by the way
11:40.34Assidthrough which network?
11:40.35ZeeekJunction Networks
11:40.45Zeeekbut 0.04c/min :(
11:41.01Zeeektoo bad voipjet doesn't have tollfree DID
11:41.07Zeeekand nufone is down
11:41.19Zeeekand teliax is too far away
11:41.21Assiderr.. 2.9c/min
11:41.23Assidthats what it says
11:41.29Assidoh wait
11:41.31Assidtollfree
11:41.33Zeeeknot on TOLLRFREE incoming my friend
11:41.46Zeeekprolly 3.9
11:41.58Zeeekbut like I said I think they round up to the minute too
11:42.04Assidjunction charging for incoming too!! normal numnber
11:42.06*** join/#asterisk Itburnz (n=UNITY@itburnz.de)
11:42.11Zeeekso getting back to the url test
11:42.26Itburnzgood afternoon everyone
11:42.27Assidshit.. i gotta get back to finishing this code
11:42.38AssidZeeek: let me know how it works out
11:42.40Zeeektoo bad all I want to do is ring the sip url
11:42.43Assidtyr and macronise it
11:42.43Zeeekok
11:42.53Itburnzhey does anyone of you know about a problem regarding asterisk 1.2.7.1 & spanDSP ? i get the error "app_rxfax.so: undefined symbol: t30_get_far_ident" - was wondering if anyone here knows a solution
11:43.04Assidringing sip url aint a problem
11:43.50Assidthe bad thing is.. you need to have the dialplan defined into [default]
11:44.56Assid8610 ? yours?
11:45.10Zeeekcan you dial sip url?
11:45.16Assidyep
11:45.24Assidyou signup already?
11:45.34ZeeekI already have code in extensions for it
11:45.47ZeeekI didn'yt set up sipbroker lookup though
11:45.57Assidin default?!?!?
11:45.58ZeeekI don't want or need that service
11:46.19Zeeekyeah as I said I have code in the default section to receive calls for certain aliases
11:46.27Assidwell. its just that people can call you for free.. throough a 'central' location
11:47.01Assidgimme your sip uri..
11:47.12Zeeekthry this:
11:47.31Zeeektsturl@r.declic.com
11:48.29ZeeekI used to use e143.org
11:48.33Zeeekoops, not exactly
11:50.07lucifrZeeek, how about just getting IP phones for my mom overseas and configure that phone to call me over in the states?
11:50.26Zeeekthat's the best solution for what you want I'd say
11:50.57lucifrZeeep, good... now we are getting somewhere...
11:51.25Zeeekyou might want to get a cheap IAX hardphone for her
11:51.36Zeeekthat way there is zero setup problems
11:51.48Zeeekof course she needs a high speed internet connection
11:52.05ZeeekAssid, I am not getting any calls
11:52.13*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
11:52.13lucifrZeeep, how about a "device" that I can configure and also convert signals from a regular phone (cutting cost here) :P
11:52.15Assidsip debug
11:52.17*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)
11:52.18Assidpastebin
11:52.37Zeeeklucifr yes, that'll work well, it's called an ATA
11:52.41lucifrZeep, she has DSL overseas
11:52.48Zeeekbut again, requires Internet access
11:52.51Zeeekok
11:53.14ZeeekWould she have others she'd like to call in the US?
11:53.20lucifrZeeep, computer required for ATA?
11:53.28Zeeekno that's the point
11:53.37lucifrZeeep, not necesarely..
11:53.38ZeeekI hate being tied to a PC phone
11:53.56Zeeekand you want to set up an asterisk box to play?
11:54.29ZeeekAssid try again, I'll turn debug on
11:55.12lucifr"and you want to set up an asterisk box to play?" Are you asking me?
11:55.18Zeeekyes
11:55.23ZeeekI'm trying to help
11:55.28lucifrI'm not following you..
11:55.32Zeeekdetermine what the best choices are
11:55.48RoyK<PROTECTED>
11:56.06Assiddid you get it?
11:56.11*** join/#asterisk v3rmap (n=puser@unaffiliated/v3rmap)
11:56.16Zeeeklooking
11:57.03Zeeeknot getting anything even remotely suspicious
11:57.14Zeeeklike 'tst' or 'declic'
11:57.20v3rmapHi, I've installed asterisk on Ubuntu, and started it. But I see thru nmap that the port no. 5060 is closed. An x-lite phone is also not able to connect and times out. What could be wrong?
11:57.55lucifrZeeek, what I really want is for her to call me from a IP phone and Asterisk redirect the call to my wireless IP phone and reach me anywhere... Now, she is not computer savvy and I'm trying to cut cost...
11:57.59v3rmapI've modified extensions.conf and sip.conf to add users and extensions.
11:58.06*** join/#asterisk homac (i=holle@strace.org)
11:58.15Zeeeklucifr I understand, millions do this daily :)
11:58.33jqlv3rmap: did you try running it manually as asterisk -f -vvvvvv or some such?
11:58.44*** part/#asterisk homac (i=holle@strace.org)
11:58.45lucifrZeeep, I only need a ATA and that's it?
11:58.53v3rmapjql: I just started it manually as "asterisk"
11:59.00Zeeekby wireless, you mean cellphone?
11:59.15lucifrwho me?
11:59.24jqlv3rmap: adding -f keeps it in the foreground, and -vvvvv makes it print lots of verbose trace info
11:59.25Zeeeklucifr wireless ip phone? What is it?
11:59.34v3rmapjql: asterisk is running and I can go to the asterisk command interpreter by using "asterisk -r"
11:59.48v3rmapjql, thanks I'll try -vvvv
12:00.04jqlrather than nmap, does netstat -a show udp:5060 open?
12:00.33lucifrit's a wi-fi phone that register itself to a Asterisk box through a open wireless network so Asterisk can redirect calls.. (inbound and outbound(\\\)
12:00.44v3rmapjql, netstat -a does not show any line that has 5060 in it.
12:00.53jqlwell, that's bothersome
12:01.06Zeeeklucifr oh so you'd wander around town and talk thru hotspots?
12:01.40*** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.56)
12:02.14lucifrZeeep, not that popular now.... but since Google will be rolling out free wi-fi across the US... it will be nice...
12:02.35Zeeekok, I'm not folliwng that too closely at the moment
12:02.43MoutaPTHi, any one could advice me how do i check what went wrong in my production server that just went down?
12:02.44lucifr:)
12:02.48AssidZeeek: did you get it?
12:02.57Zeeekanyway,n yeah an ATA and a normal phone plugged in to it will do the trick
12:03.06ZeeekAssid, nothing at all
12:03.12Assiddude.. it has to work
12:03.20ZeeekWhat is this "debug server" they mention?
12:03.24Assidtry calling this.. 3001@mercury.sphrerelinx.com
12:03.36MoutaPTI've started it now, and also didn't get the msg like your pc has been incorrectly shut down...
12:03.42Assidignore that
12:03.48Assidjust sip debug your * box
12:04.01MoutaPTAssid talking to me?
12:04.02Zeeekok but anyway I haven't put ANY code in extensions
12:04.15ZeeekAssid I did that and there was nothing trying to come in
12:04.15AssidMoutaPT: nope
12:04.20lucifrZeeep, what a good and reliable ATA device?
12:04.40Assidsip:tsturl@r.declic.com
12:04.45Zeeekyou'll have to do research on that. Sipura has a lot of happy users
12:04.54jql<-- sipura user
12:05.07*** join/#asterisk madounet (n=madounet@juv34-2-82-226-155-19.fbx.proxad.net)
12:05.15coppicegood *and* reliable. gee, some people want *everything* :-)
12:05.18Zeeekexten => tsturl,1,NoOp(${EXTEN} ${CONTEXT} ${SIPDOMAIN})
12:05.20lucifrZeeep, beautiful....
12:05.28jqlI actually shopped for cheap
12:05.32jqlthat it worked was a bonus
12:05.41Zeeeklucifr watch out for power supply voltages if you buy it!
12:05.49Assidr.declic.com --- is that your box
12:06.01Zeeekit's an alias for now, yeah
12:06.16lucifrok, thx
12:06.23lucifrbrb
12:06.42*** join/#asterisk tdonahue (n=tdonahue@www.vonworldwide.com)
12:08.38ZeeekAssid, actually I see what I wanted to know because Dial, while complaining of a loop condition when I dial myself, also says "Thank to ..." and gives the domain name
12:09.07Zeeekthat same uri is likely available in the SIP_HEADER(TO) variable
12:09.37*** join/#asterisk the_magic_bean (n=the_magi@209.43.15.211)
12:09.45*** part/#asterisk the_magic_bean (n=the_magi@209.43.15.211)
12:10.18*** join/#asterisk ghost99 (n=neville@222-152-219-77.jetstream.xtra.co.nz)
12:10.49*** join/#asterisk the_magic_bean (n=the_magi@209.43.15.211)
12:10.56mutman o mn
12:11.08mutwhy does mcdonalds breakfast have to be so expensive
12:11.26lunkbecause you bought too much stuff that will kEEL YOU
12:11.27nextimebleah
12:12.07lunki'm having a bfast of champions right now, redbull and poptarts
12:12.12lunkfear the sugar rush
12:13.08nextimecappuccino and croassant, and a black strong expresso coffe, is all what you need for a good bfast :)
12:13.23muti got bad acid reflux
12:13.27lunkcappuccino and expresso?
12:13.29muti can't drive coffee/cappa
12:14.06mutdrive = drin
12:14.07mutk
12:14.08mut:P
12:14.17nextimelunk yep, cappuccino as a real bfast, and coffee for your brain wakeup
12:14.19lunkobviously you need to
12:14.36lunkah
12:14.54nextimelunk : i'm italian, expresso is a must for me :)
12:14.59v3rmapfolks, can anyone tell me why asterisk is not listening at port 60 on my system?
12:15.17lunkah, someday i will visit italy
12:15.24v3rmapHere is the sip.conf: http://pastebin.com/692092
12:16.12nextimelunk : you will probably try the best food in the world here :)
12:16.37lunknextime: millions of fat americans would argue with thtat, but i'll take your word for it
12:16.59*** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au)
12:16.59Assiddamn.. my noop didnt get executed
12:17.09lunkmy great grandmother on my mom's side came from sicily around 1910 or so
12:17.24v3rmapOh, sorry, I meant asterisk is not listening on port 5060. And the sip.conf is:  http://pastebin.com/692092
12:17.39nextimelunk : yeah, so you have some italian blood in your body :)
12:18.04lunkoh yea, but my brother got all the outward genes, i just got the rage \o/
12:19.15lunkanyway, when dealing with .call files, if your trunk is in use and a second call file is added to the outbound spool
12:19.25lunkdoes it fail when the trunk is in use (or at capaciy)?
12:19.34nextimeanyway, you can find good food in any part of the world, the real problem of fat americans is the food education, not the food itself
12:20.11lunki want to be able to drop like 10 .call files and not worry about them
12:21.14nextimelunk, you can set retry timeout 1, 2 3 time or so...
12:21.47nextimeand when it exceed it disappear from your spool dir
12:21.58lunkyea, that seems kinda lame
12:22.11lunki'd like to know that every .call file is actually routed outbound at least once
12:22.25Greek-Boyif a call comes in on a zap interface can caller id show on the IP phone?
12:22.34lunkGreek-Boy: yes
12:22.37mutwould anyone consider 15 min breaks in an 8 hr work day a lot?
12:22.44mut3 15 min breaks*
12:22.56lunkmut: no, but it depends on the situation
12:23.10mutin an accounting office
12:23.17lunkoh hell no
12:23.24lunkthat crap is boring enough as it is ;)
12:23.26mutno?
12:23.27Greek-Boynice
12:24.28nextimelunk : i think that you can set an extension that do Dial and some other work instead of dialout directly from the .call file, no?
12:24.43lunkprobably
12:24.51lunkideally i'd like to use Queues in a backwards way
12:24.57nextime( maybe i'm saying something stupid, but it can work )
12:25.20lunknextime: if i want it to show up in the log at all i have to route it to an extension
12:25.23Assiderr
12:25.27Assidsomething is totally wrong here
12:25.48nextimeAssid : probably my synapses
12:25.51ZeeekAssid so you are supposed to be able to dial these numbers as *12345 ?
12:26.21Zeeekis srvlookup needed?
12:27.01Assidyes
12:27.13Assiderr.. am having very very weird issue
12:27.25Assidi disabled my extension from default
12:27.28*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
12:27.30Assidand yes.. reloaded it
12:27.37Assidstill the calls are coming on my extension
12:27.57ZeeekAssid do you have any really wide expressions like _. ?
12:28.12ZeeekI had this happen yesterday:
12:28.14Assidin default.. NOTHING
12:28.37Zeeekexten => _${SOMEVAR}.,1,do something
12:28.40Assidand i know for a fact it tries to match with default cotext
12:28.49Zeeekwell, if SOMEVAR happens to be empty....
12:29.00Zeeekthat will talke ALL calls
12:29.10Assidnothing
12:30.06Assidim removing the default context
12:30.09Assidlets see what it does
12:30.19Assid!@#!#@@#$#@ its still calling
12:30.21Assidwtf
12:30.33Assidsip cached?!?!?
12:30.37*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
12:30.53orlokAssid: reload or restart?
12:31.13Assidtried both
12:31.21Assidi even shut down asterisk for 10 seconds
12:32.27Zeeekdid you kill a chicken during a full moon, though?
12:32.34Assidnah
12:32.38Zeeekbecause without that, no dialplan is safe
12:32.45Assidthere was chicken flu
12:32.48AssidH5N1
12:33.02Zeeekso I got sipbroker working. Now what?
12:33.11Assidyuou did?
12:33.17Assidminne just went kaboom
12:33.20Zeeekwell I got the test messages
12:33.27Assidi removed the extension .. it still doesnt work
12:33.35Assidewrr..it still works
12:33.37*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
12:34.25Assidfinally dies
12:35.13kavitvideo conferening and oh323, stuff nightmares are made of
12:35.27kavitno, oh323, stuff nightmares are made of
12:35.28kavit:(
12:35.56lucifrAnybody familiar with the Linksys PAP2 Phone Adapter (ATA)?  Is it locked to "Vonage"?
12:36.11Zeeekthe models ending in NA are unlocked
12:36.17Zeeek(I theeeeeenk)
12:36.45lucifrhmmm...
12:36.46Assidumm.. it doesnt wanna check in the default context no more
12:36.47*** join/#asterisk pigpen2 (n=mark@fw.seamans.cc)
12:36.56AssidZeeek: is yours checking in the default context?
12:37.00lucifrok, thanks Zeep
12:37.03AssidMay  1 18:06:29 NOTICE[1004]: pbx.c:1738 pbx_extension_helper: Cannot find extension context 'spherelinx'
12:37.17pigpen2Hi all, would I assume correctly that Asterisk does not have an ACD (Automatic Call Distribution) solution yet?
12:38.41Assidoh wait
12:38.41Assidi think i know why
12:39.06ZeeekAssid, these international number I see: you dial *code*number or what?
12:39.24Zeeekon this page: http://www.sipbroker.com/sipbroker/action/providerWhitePages
12:40.43[TK]D-Fenderpigpen : No you wouldn't.
12:41.12[TK]D-Fenderpigpen : I take it you looked REALLY hard :)
12:41.29[TK]D-Fender(#2) that is...
12:42.02Assidyes yes.. i got it
12:42.05coppiceone of the greatest cons of the telecoms world was how PBX vendors sold customers on the idea that ACD is a really really hard problem.
12:42.06[TK]D-FenderWish I could make autocomplete do a best case match agianst "last spoke"
12:42.15kavitpigpen2: asterisk is really virtual monkeys on a virtual switchboard jacking random lines
12:42.15Assidi understood how to do vhosting on this based
12:42.21Assidsoo freaking easy
12:42.51[TK]D-Fenderkavit : Virtual monkeys were too expensive to develop so we outsourced it to Indian monkeys :)
12:43.01pigpen2[TK]D-Fender, well, after searching the wiki and googling, all I could find is a page referencing commercial products....
12:43.14pigpen2so my assumption was based on research.
12:43.27pigpen2personally, ACD sounds like a well built queue.
12:43.27[TK]D-Fenderpigpen2 : Dear God you really didn't look to well... try "asterisk queues" on the wiki.....
12:43.38coppiceif an ACD solution requires monkeys, i doubt typical call centre staff would really be up to the job
12:43.40lunkhow hard is it to install the PGSQL application after-the-fact?
12:43.54kavit[TK]D-Fender: hahaha saved some money I bet
12:43.57[TK]D-Fenderpigpen2 : There is this miraculous module called "app_queue" thats been around for a great many years....
12:44.12pigpen2yeah...as long as you don't use chan_agent
12:44.15[TK]D-Fendercoppice : Note we didn't say TRAINED monkeys :)
12:44.26[TK]D-Fendercoppice : These are RANDOM connections after all!
12:44.45coppicesimply saying monkeys implies a certain basic IQ
12:45.00pigpen2[TK]D-Fender, sorry to get you on the wrong side of the bed....I was unsure if ACD was like queues.....
12:45.07[TK]D-Fenderpigpen2 : What about chan_agent?  Can work depending on your needs.  You can also to static agents, etc.
12:45.16[TK]D-FenderACD = Queues
12:45.18pigpen2yes...static works fine...
12:45.30kavithttp://www.voip-info.org/wiki/view/Agents+without+agent+channel
12:45.38kavitpigpen2: there you go
12:45.48ZeeekAssid, it does work
12:45.51[TK]D-Fenderpigpen2 : There is also an add-on package calld ICD you can lookup on the WIKI which takes a different approach.
12:45.58Assidyeh
12:46.03coppiceeven if during the daytime the agents might appear to outwit a monkey, in the middle of the night on the graveyard shift high on crack could they still manage it?
12:46.05Assidyou just use simple goto
12:46.14[TK]D-Fenderkavit : Thats pre 1.2.x code and basically all stupid dial-plan.....
12:46.43kavit[TK]D-Fender: don't shoot the messenger :(
12:47.03[TK]D-Fenderkavit : Sorry, you're all I've got :)
12:47.15pigpen2[TK]D-Fender, thanks...sorry about the confusion...I was just unfamiliar with the term "ACD"...queues, well...lets just say, chan_agent made my life hell for about 2 weeks.
12:47.32*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
12:47.33[TK]D-Fenderpigpen2 : What was your probelm, and what do you need?
12:47.38kavit[TK]D-Fender: alright I will just start bothering you with oh323 video conferencing questions
12:48.24pigpen2[TK]D-Fender, well, using chan_agent was causing -some- sort of an issue that was causing a dead lock of asterisk.  Going to static agents took care of it.
12:48.34[TK]D-Fenderkavit : Sorry, I have successfully avoided H.323 since discovering * and SIP :)
12:49.06the_magic_beananyone have any idea how to backup/restore voicemail?  I have the audio files but im not sure how asterisk stores the 'database' connecting messages to users to files.
12:49.17[TK]D-Fenderpigpen2 : Could be that it was a bug that has since been resolved.... What kind of dead-lock.  Like hard-code or something making the dial-plan stick?
12:49.56[TK]D-Fenderthe_magic_bean : Depends, sounds like your using a database to store it as opposed to the native mailbox method
12:50.06pigpen2[TK]D-Fender, well, this was about 2 weeks ago...maybe 3.  I opened a ticket with Digium, and they said "yeah...use static's....it is a know issue"
12:50.37kavit[TK]D-Fender: yeah but people want everything, I told them if you want a video conferencing bridge go Polycom, but curiosity got the better of me and here I am trying to see if it can be done with Open source products
12:50.47codebreakerwhat is the first place to look when "res_odbc.c:563 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno= 0 [unixODBC][Driver ManagerData source name not found, and no default driver specified"
12:51.15the_magic_bean[TK]D-Fender: well not really sure about that, I set the box up, and i did not do anything with a pgSQL or mySQL, so i would assume it is whatever native mailbox method is
12:51.32*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
12:51.40[TK]D-Fenderkavit : You could use Ekiga for video....
12:51.41kavitcodebreaker: unixODBC cannot find a data source and a driver?
12:52.18kavit[TK]D-Fender: no I want to see if I can get it running over ISDN or play nice with another polycom av bridge
12:52.36[TK]D-Fenderthe_magic_bean : Then you just need to back up the voicemail foder (by default /var/spool/asterisk/voicemail
12:52.53[TK]D-Fenderkavit : Oh... its too late :) hehe sorry!
12:53.03kavitcodebreaker: what DB are you using?
12:53.09the_magic_bean[TK]D-Fender : thank you sir
12:53.37kavitcodebreaker: install a driver for that db and if you want ease of configuration use ODBCConfig, its a gui tool that is packaged with unixODBC
12:54.25kavit[TK]D-Fender: yeah there is not one decent open source solution that can do video conferencing with h323 properly
12:54.33codebreakerkavit: i use postgres and tried http://www.asteriskguru.com/tutorials/realtime_pgsql.html
12:55.11kavitcodebreaker: did you check if the driver was installed?
12:57.33codebreakerkavit: jupp is installed. i now go to increase ma logs on the postgres.
12:58.55AssidZeeek: are you setting up the srv?
12:59.10ZeeekI used to but it's off now
12:59.22Zeeekif you mean srvlookup
13:00.06Assidyeah
13:00.12Zeeekso people using anyone those services can call me free? I can't get that to work yet
13:00.15Assidim trying to setup the dns to use it
13:00.28Assiderr.. well.. did you setup your sip broker?
13:00.30Zeeekdyndns allows all those records
13:00.49Assiderr.. im doing it frommy own dns
13:01.04ZeeekI don't see what the alias is
13:02.53Zeeekis alias the same as member number?
13:03.25Assidno no.. whats your broker number
13:03.29Assidits as soon as you login
13:03.37Zeeekthat say member number
13:03.46*** join/#asterisk zaf (n=zaf@65.255.203.114)
13:03.52Assidyeah
13:03.53Zeeekoh that was a number I made up^
13:03.54Assidwhats that
13:04.01Assidits part of the sip uri
13:04.08Assid*xxx-sipuser
13:07.43*** join/#asterisk DoktorGreg (n=Greg@70.91.121.89)
13:08.06ZeeekAssid if the code given for wengo is *248 what do you dial after?
13:08.17Zeeekthe whole *011blahblah?
13:10.34Greek-Boywith IP phones that have two RJ45 ports for passing one link to the PC, does the PC and the phone have to be on the same network?
13:10.37[TK]D-Fenderkavit : Have you tried Ekiga?  Does H.323 & SIP
13:10.37Assiderr..
13:10.42Greek-Boyi mean same subnet
13:10.45Assid*248 and your DID
13:11.00[TK]D-FenderGreek-Boy : Not if your switch has vlan support.
13:11.40Greek-Boyis it better to put the phones on a seperate unmanaged switch or two use vlans?
13:11.41[TK]D-FenderGreek-Boy : Not that it would save you from bandwidth issues, and assumes the phone supports it as well.
13:11.51[TK]D-FenderGreek-Boy : Seperate LAN naturally.
13:12.11Greek-Boyok but are unmanaged switches better?
13:12.20Greek-Boycoz all bandwidth is dedicated to the phones
13:12.51[TK]D-FenderGreek-Boy : depends on your tastes & needs.  Technically better, yes, worth it?  Depends on you. I personally don't care for it.
13:13.04[TK]D-FenderGreek-Boy : I wouldn't think twice if you're getting a seperate lan.
13:13.10Greek-Boyi see
13:13.10Greek-Boy:)
13:13.13[TK]D-FenderGreek-Boy : Describe your deployment
13:13.52AssidYES... it works
13:14.12AssidZeeek: i got this to work.. you can have virtual hosting this
13:14.15Assidvery very easy
13:14.18Assidsimple macro
13:14.23kavit[TK]D-Fender: yeah but asterisk doesnt do conference rooms with Video and h323 methinks
13:15.24ZeeekAssid so let's see it!
13:15.25Greek-Boy[TK]D-Fender 122 phones that I need to connect to asterisk, I dont have wiring in place but I have time to get the wiring done. But I was thinking if I should put phones and PC's on same network as CAT6 wiring for PC's is already in place but I think i'll do it seperate
13:15.30techman97_andygood morning all - has anyone setup a faxing module in * where I can accept faxes and convert them to a TIF/PDF/JPG/whatever?
13:15.51coppicenope. nobody ever did that
13:15.56techman97_andy=P
13:16.03Zeeekhahaa
13:16.13kavitGreek-Boy: with some phones you can piggy back the computer off the back of the phone
13:16.19Zeeekthey did but it's not perfect :)
13:16.22kavitGreek-Boy: look into 802.1Q
13:16.26kavitvlan
13:16.42[TK]D-Fenderkavit : No... it wouldn't manage it, it'd only allow the endpoints to call each other.
13:16.43Assidgimme a min.. setting up my line
13:17.00[TK]D-FenderGreek-Boy : are your PC really network heavy?
13:17.04kavitso you can use the same physical network but seperate vlan segments
13:17.26Greek-Boy[TK]D-Fender, not really heavy. It's a gigabit network
13:17.31kavit[TK]D-Fender: yeah I was trying to get asterisk to pass ISDN video calls to gnugk and openmcu
13:17.33kavita lot of issues
13:17.51kaviti am just going to give up or wait until someone develops something
13:17.54[TK]D-FenderGreek-Boy : I mean are your PC's going to be constantly working on huge files?  because pluggin in-line with your phone will drop everything to 10/100
13:18.18[TK]D-Fenderkavit : Well you are definately working outside my field of experience....
13:18.26codebreakerkavit: fixed. a typo in /etc/odbc.ini "only" one s to much in [asterissk]
13:18.35Greek-Boynot huge files all the time, just database stuff
13:18.49kavit[TK]D-Fender: i am working outside my own field of experience as well
13:19.30kavit[TK]D-Fender: I though I would get back into hardcore programming and write a few enhancements to openmcu, I looked at the oh323 code
13:19.41kavitand just shook my head and gave up
13:19.43[TK]D-FenderGreek-Boy : Typically I'd trust a decent phone to take care of you just fine.
13:19.51[TK]D-FenderGreek-Boy : What models are you considering?
13:19.53kavitcodebreaker: ah simple error yet a show stopper
13:20.31Greek-BoyST-302 IAX2 IP Phone
13:20.37Greek-Boybut also polycom
13:20.56kavitGreek-Boy: I did an Cisco VoIP roll out for a major client and we used vlans, really for normal workstation stuff piggybacking off the phone is good enough
13:21.03Greek-Boyi also looked at linksys sipura phones
13:21.09AssidZeeek: satish@spherelinx.com
13:21.10kavitpolycom are generally good
13:21.11Assidcall that
13:21.17Assidsip uri
13:21.23Greek-Boyi'll consider vlans kavit
13:21.38Greek-Boybut its generally best practice to seperate voip network, right?
13:21.59kavitGreek-Boy: what for? unless you are hosting stuff no
13:22.01[TK]D-FenderHahah, craptastic SIP hardphone suggestion for * article on NewsForge :D
13:22.23coppicepage filler of the week
13:22.33[TK]D-FenderGreek-Boy : Ummm... Don't know that phone and trust it even less... Go with Polycom or Cisco for sure if you're going in-line.
13:22.44Greek-Boywhat wiring works better? CAT5 or CAT6? I know this might sound like a stupid question but a few network components out there work better with CAT5 and this is why i'm asking even though CAT6 can handle higher throughput
13:22.51AssidZeeek: alternatively 3001@spherelinx.com
13:22.56Assidor
13:23.00[TK]D-FenderGreek-Boy : Irrelevent since they're stuck on 10/100.  Cat5 = fine
13:23.07Assid*75493001
13:23.09kavitGreek-Boy: also a word of advice, make sure all the workstations and the network card drivers support 802,1Q or you will find vlans will mess everything up
13:23.21Greek-Boyhmmm, ok
13:23.22Assidthey all will work
13:23.34AssidZeeek ?
13:23.36Greek-Boywhich phones are better between cisco and polycom?
13:23.58kavitI had to mess around for 3 days rechecking all my cisco switch config and turns out the customers ARCHAIC SOE was to blame
13:24.02kaviti was not happy
13:24.20kavitGreek-Boy: CIsco if you have the cash
13:24.34kavitGreek-Boy: their phones are generally very reliable.
13:24.45Greek-Boyeven better voice quality then polycom?
13:25.11Alystairhmmm
13:25.29kavitGreek-Boy: voice quality has a lot of factors that come into play like bandwidth, codecs etc but on an internal network
13:25.33Greek-BoyI also need to make sure that the phones support PoE as i want power to be distributed from a central point. I will have a 10KVA UPS to supply power to all phones, hope that will keep them going for atleast an hour or two
13:25.40kavitGreek-Boy: it shouldnt matter
13:25.49kavitGreek-Boy: Cisco supports PoE
13:26.04kavitGreek-Boy: even their Linksys models do
13:26.14[TK]D-FenderGreek-Boy : I'd say they're about equal, but Polycom being a far better value.
13:26.24*** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk)
13:26.28kavitAlystair: whats your budget? I found Grandstream GXP 2000 to be ok for a low priced phone
13:26.33Assidi have only one issue
13:26.37Assidi cant hear crap
13:26.44ZeeekAssid sorry I was feeding my face
13:26.45Assidprolly cause of routing
13:26.51Assidbut more or less works
13:26.53kavitAssid: go to #windows :P crap talks there
13:26.54[TK]D-FenderGS = Plagued.  A "to be avoided" brand onmy list...
13:27.02Zeeekhahha
13:27.13Greek-Boyi was thinking of buying d-link PoE 48-port switches but I might as well as get cisco for that too since i'm going to be buying cisco phones
13:27.29kavitGreek-Boy: cisco are very expensive
13:27.35robin_szohh goody, new GXP 2000 firmware at last, and it fixes my GUI problem
13:27.52kavitGreek-Boy: esp 7940 and upwards
13:28.01[TK]D-FenderGreek-Boy : Save a bundle of cash and get a D-Link DES-1526 24 port PoE switch and some Polycom PoE phones....
13:28.18robin_szand Ive won the lotto, and been appointed chief shower-keeper for the female showers at the next olympics
13:28.44Zeeekrobin_sz better the next Vegas revue, doncha think?
13:28.49mutwhy, theres no boobs at the olympics
13:28.50kavitGreek-Boy: if you really want to go the cisco path cisco switches work really well with Vlans
13:29.08Zeeekgo rent "Showgirls" again :)
13:29.09*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
13:29.10kavit[TK]D-Fender: i wouldnt trust dlink with proper vlan and qos
13:29.11Greek-Boy[TK]D-Fender, then I'll have to buy 5 switches for all the phones, lol
13:29.12robin_szVegas? no . thats a clean living mormon town ...
13:29.29ZeeekWatch the Sopranos
13:30.53DoktorGregSopranos has gotten boring
13:30.53[TK]D-Fenderkavit : They support it and we are talking a minimal install here.
13:30.53Zeeekis anyone actually using e164 ?
13:30.54DoktorGregits time for them to have a masacre and kill off everyone
13:30.54AssidZeeek: did you try
13:30.54ZeeekWhat season you on?
13:30.54robin_szactually, I think there is more chance of either if those things happening than my GXP getting fixed :(
13:30.54[TK]D-FenderGreek-Boy : True, but look at the COST of those switches... $400 USD
13:30.54Hmmhesaysso i specifically arrange for a ride to work this morning
13:30.54ZeeekDoktorGreg Tony already is bleeding to death in the kitchen
13:30.54Hmmhesaysand she doesn't show up
13:30.54kavit[TK]D-Fender: ah ok, well I have just had a lot of issues with D-Link hardware
13:30.55DoktorGregshut up!
13:30.55DoktorGregomg
13:30.55coppicethen they'll rehash all the stories as the Tenors, and everyone will think its fresh and new
13:30.55[TK]D-Fenderkavit : I've been trouble free.....
13:30.55Greek-Boy[TK]D-Fender, they dont come in 48-port?
13:30.55ZeeekDoktorGreg the last season first epsode is free on the net by the way
13:31.01kavitGreek-Boy: is it your money? or is it your companys money?
13:31.04Zeeeklegally thur google video
13:31.05DoktorGregthanks
13:31.08*** join/#asterisk C4T3l (n=rcall01@216.54.143.2)
13:31.09Greek-Boycompany money
13:31.12Greek-Boylol
13:31.12Greek-Boy:)
13:31.19robin_szspend it like rain then
13:31.25[TK]D-FenderGreek-Boy : Not yet... but we're talking minimal patching here...  how much are 48 port Cisco-capable PoE switches going to cost you?
13:31.38ZeeekAssid I haven't tried anything but I can see the routing works because asterisk detects the loop
13:31.50Assidcool
13:31.52ZeeekI never talk on the phone anyway, I just TEST them ;)
13:31.57Assidhaha
13:32.02Assidwell
13:32.05coppicedo cisco have 48 port switches with 48 ports of PoE now?
13:32.05Assidit works nice
13:32.18Zeeekunlike faxes which I wouldn't send for fun
13:32.25kavitGreek-Boy: if you work for someone else, i suggest you bulk buy cisco hardware and talk your account manager into getting you signed up for a few discounted courses so you get paid to go and sit in a course for a week or two and your boos gets jealous
13:32.38DoktorGregI need like a 100 person conference call for a day
13:32.40kavitboss even
13:32.48ZeeekDoktorGreg see junction networks
13:32.50Greek-Boylol kavit
13:32.53Greek-Boynice thinking
13:32.53Greek-Boy:)
13:33.02DoktorGregZeeek, thx
13:33.25Greek-Boy[TK]D-Fender i know what u mean, its all about cost-effectiveness. If the company pays for cisco then I'll consider that route
13:33.26Zeeekthere is a company that does just conferences too but I can't remember their name
13:33.37*** join/#asterisk ComputerWarm (n=donc@HS196-230-97.nt.net)
13:34.01kavit[TK]D-Fender: they cost a lot cisco switches, we had a catalyst 6500 switch chassis
13:34.10*** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
13:34.31kavitGreek-Boy: i have been contracting for the last few years and you learn how to milk the clients who make life hard for you
13:34.34lucifrnite every1
13:34.49codebreakersomebody a pointing link for voicemail in database without *local* wav files?
13:34.52lucifrwait... good morning.... lol
13:35.06lucifrlet me say.... goodbye
13:35.15*** part/#asterisk lucifr (n=chatzill@c-24-126-108-87.hsd1.ca.comcast.net)
13:35.45kaviti feel like a dark evil cloud of misspelling has been lifted off this channel
13:36.18[TK]D-FenderGreek-Boy : Yeah, typically lets say that PoE Polycom IP 501's = $200 USD and the Swich = $400 (16$/port) = $216 / ext.
13:36.27Assidwhats better.. making a new extension.. or a goto ?
13:36.28DoktorGregomg charge PITA premiums
13:36.37DoktorGregPITA premiums are the norm
13:36.59[TK]D-FenderAssid : That makes no sense...
13:38.27Assidokay i have exten => assid,1,Dial....... and i have exten => 3001,1,Dial....... both have the same dial
13:39.11Assidjust wondering if it makes any difference to the processor
13:40.28lunkhow can i pass in an arbitrary dial-plan variable with a Call file?
13:41.15C4T3lhello world
13:42.02Assidanyone behind a nat?
13:42.11Assidand can help me test a sip uri call?
13:42.45Zeeekhow long do these ezdial calls take to happen?
13:42.52[TK]D-FenderAssid : Goto or Macro......
13:43.31ZeeekAssid, by the way if you don't hear anything, notice the canreinvitie=yes in their sip entries? If yiou're behind nat that may be why
13:44.06Hmmhesaysor if you have a t1/e1 card in with nothing plugged into it
13:44.16the_magic_beanwhen using RealTime for configuration and whatnot with a sql db, does asterisk only store things like users in the sql db or would voicemail files get stored in sql also?
13:44.19AssidZeeek: i only get that problem with sip uri direct calling
13:44.37*** part/#asterisk ComputerWarm (n=donc@HS196-230-97.nt.net)
13:45.48Zeeekso I should be able to call FWD and use the code to call back to asterisk. I think I did that once and there was like 50000ms of lag, no kidding
13:45.59Hmmhesays50 seconds?
13:45.59Zeeek5000
13:46.10Zeeekno 5-10 seconds
13:46.24Hmmhesaysnice math
13:46.26Zeeekand it was an octave lower than my voice!
13:46.31filewow, my bank actually has Euros
13:46.52Zeeekwhat? here a ms is 1/10,000 of a second!
13:47.01Hmmhesayshaha
13:47.08Zeeekon national holidays especially
13:47.11Hmmhesayssorry I missed a booty call this weekend, i'm a little uptight
13:47.21filepoor Hmmhesays
13:47.32Zeeekmissed because?
13:47.40HmmhesaysI was incarcerated
13:47.41kavit[TK]D-Fender: i like the new ekiga, it is good
13:47.53Zeeeke-geek-a
13:48.00[TK]D-Fenderkavit : I"m still waiting for a Win32 build as I don't use LInux as a desktop platform...
13:48.18Dr-Linuxwhat is Ekiga?
13:48.42kavit[TK]D-Fender: maybe its time to stop playing Battlefield or world of war craft :P
13:48.53codebreakerthe_magic_bean: there a two ways i ve heared. one ist storing only users and an other approuch saves also all files( this one i am also looking for) when i find the link i will you inform
13:49.51Hmmhesaysjail, any type of jail sucks
13:50.15fileHmmhesays: didn't get no lovin' in jail? :P
13:50.42Hmmhesayswell, it was min security, but no conjecalK(sp?) visits
13:51.09vader--morning
13:51.17Hmmhesaysseriously if you live in fargo and haven't been to jail at least once... my god you must be bored
13:51.28vader--hi d-fender, hmm, dr
13:51.30[TK]D-Fenderkavit : I only have 2 games, HL2 & Diablo2.  And I harly ever play either :)
13:51.58[TK]D-FenderHmmhesays : this is for that car trashing your GF gave you?
13:52.12Hmmhesays[TK]D-Fender: X girlfriend yes
13:52.27kavit[TK]D-Fender: last game i REALLY got into was Carmageddon 2
13:52.28[TK]D-FenderHmmhesays : Avoid drunks.....
13:52.33kavitAGES ago
13:52.42codebreakerthe_magic_bean: http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
13:52.51Hmmhesays[TK]D-Fender: no, now when I got out there is a 20 hidden in my wallet for a cab ride no matter what
13:53.15lunkwhat is the syntax for the Data: prompt in a .call file?
13:53.19ghost99hi all Wondering if someone can help, I don't seem to be able to dial out, I have the following exten => _9.,1,Dial(Zap/1-1/${EXTEN:1}) in the context [sip] in extensons.cof wher should it be ?
13:54.08[TK]D-FenderHmmhesays : Seriously blows... whats the insurance situation?
13:54.23[TK]D-Fenderghost99 : Bad Zap format
13:54.24Hmmhesaysinsurance paid for it, I have no car now
13:54.27Hmmhesaysno license
13:54.38[TK]D-Fenderghost99 : exten => _9.,1,Dial(Zap/1/${EXTEN:1})
13:54.39Hmmhesays$1000 dollar fine, 5 days in jail
13:54.53fileHmmhesays: but now you're back!
13:55.01Hmmhesaysfile: i go back tonight at 6
13:55.03[TK]D-FenderHmmhesays : Wait, what did you hit, and WHO was covered?
13:55.14filewell, back for a bit
13:55.21DoktorGregdoes anybody have a fax retreive script?
13:55.23[TK]D-FenderHmmhesays : And you killed your points on this did you?
13:55.27HmmhesaysI was covered, and she pulled the steering wheel hard to the right on a residential street
13:55.32DoktorGreggo up to fax machine
13:55.40DoktorGregdial into an extension
13:55.50HmmhesaysI was just over the limit, passed my field sobriety test, failed breath
13:55.54DoktorGregextension starts transmitting stored faxes
13:56.01Hmmhesaysso... I got busted for dui
13:56.08Hmmhesaysthat's why all the fines
13:56.14DoktorGregHmmhesays, just get a lawyer
13:56.25Hmmhesayscity prosecutor wouldn't budge
13:56.40HmmhesaysI have some other stuff on my record from my mischevious years
13:56.51Hmmhesayslike I said, this is fargo... you'd know if you lived here
13:56.57DoktorGreglol
13:57.00[TK]D-FenderHmmhesays : Not your first DUI I take it having lost your license, or are you too new a driver?
13:57.06DoktorGregIm originally from vally city
13:57.12MikeJ[Laptop]hmmmm he says
13:57.25fileuh oh it's MikeJ
13:57.26Hmmhesaysthen you know DoktorGreg: there is nothing to do around here except work, and get in trouble
13:57.27fileeveryone hide
13:57.38DoktorGregcookie salad!
13:57.48Hmmhesays[TK]D-Fender: no, had one way back when I actually deserved it
13:57.52C4T3li'm running * 1.2.7.1 and having problems with one way communications using SIP i'm assuming its a firewall issue.  As far as I know both ends are free of firewalling.  The far end can recieve calls and listen to pre-recorded messages on my *.  But I can't hear anything on my end.  Any suggestions?
13:57.56HmmhesaysI mean, really deserved it
13:57.57DoktorGregI get about 20 new cookie salad recipes every time i go back
13:58.23DoktorGregsheesh, move to mineapolis
13:58.35DoktorGregNorth Dakota is a dead end
13:58.41ghost99[tk]D-fender: I tried it that way also took out of sip context and put into global context in extensions.conf and still no luck .. any more cluses ?
13:58.41HmmhesaysDoktorGreg: actually i was on my way out of here when jamie helped my car into the junkyard
13:59.13[TK]D-FenderHmmhesays : Next time plow her back to her side....
13:59.26[TK]D-Fenderghost99 : HUH!?
13:59.29Hmmhesays[TK]D-Fender: there will never be a next time
13:59.37Hmmhesaysplus the girl i'm seeing now is pretty cool
13:59.38[TK]D-FenderHmmhesays : Not with HER at least...
14:00.07Hmmhesays[TK]D-Fender my life is fscked enough becuase of this, it will not happen, EVER again
14:00.38DoktorGregthe girl also owes you a sympath fuck!
14:00.44jake1932C4T3l: if you run sip debug, or rtp debug, you can see the ip addresses and ports of the endpoints
14:00.57HmmhesaysI'd rather she pay half the fines
14:00.57[TK]D-FenderDoktorGreg : Oh I think she fucked him pretty good already
14:01.04Hmmhesayscould pay for a much hotter hooker
14:01.06HmmhesaysLOL
14:01.25Hmmhesays[TK]D-Fender: in every way conceivable
14:01.31C4T3ljake1932: thanks. I've only been using asterisk for a month so theres alot i'm not aware of
14:01.33Hmmhesaysmore bad than good
14:02.22jake1932C4T3l: are both enpoints (your asterisk box and the device) both on the same network?
14:02.31*** join/#asterisk Lino` (n=Lino@i577BD559.versanet.de)
14:02.36C4T3ljake1932: No. diff nets
14:02.46jake1932C4T3l: both have public IPs?
14:02.48Hmmhesayson the upside, I think i'm going to make it to cluecon this year
14:02.53C4T3ljake1932: yes
14:03.20jake1932C4T3l: maybe you can pastebin a sip debug. - i'll take a quick look
14:04.19Hmmhesaysalthough it has been interesting, I had to make someone my bitch and shank a guy the first day
14:04.29Itburnzhey does anyone of you know about a problem regarding asterisk 1.2.7.1 & spanDSP ? i get the error "app_rxfax.so: undefined symbol: t30_get_far_ident" - was wondering if anyone here knows a solution
14:04.40Hmmhesaysmissing some header
14:04.54Hmmhesaysor the name of that function changed
14:04.56C4T3ljake1932: i wont be able to do that right away. I'm at work right now (this is an unrelated work prob. )
14:05.15jake1932oh
14:05.21C4T3ljake1932: It's really just for fun at home for this point
14:05.24Itburnzhrm, it's a fresh asterisk installation
14:05.32Hmmhesaysahh tech for fun at home
14:05.59Itburnzat least that explains why recompiling didnt do the job
14:06.01C4T3lI guess its not everyone's idea of fun
14:06.17C4T3llol
14:07.09[TK]D-Fendersame damn thing!
14:07.18Hmmhesaysfantastic
14:07.25Hmmhesaysalthough i'd probably fail my ua tonight
14:07.35fileHmmhesays: so who is your bitch?
14:07.38[TK]D-FenderUA?
14:07.43Hmmhesayspiss test
14:07.45[TK]D-FenderAh
14:08.00Hmmhesaysalthough it is min security, they frown upon that type of thing
14:08.05[TK]D-Fenderfile : Careful with words like that.... he just got out of jail....
14:08.20file[TK]D-Fender: true
14:08.28Hmmhesays[TK]D-Fender i got 3 days left
14:08.32Hmmhesaysi'm out on work release
14:08.43[TK]D-FenderHmmhesays : like house arrest, only let out to work?
14:09.02[TK]D-FenderGotcha
14:09.02Hmmhesayskind of yeah
14:09.02HmmhesaysI have them convinced i'm self employed and on call 24/7 though
14:09.06[TK]D-FenderHmmhesays : What did you hit exactly?
14:09.12fileneed us to call you randomly?
14:09.18Hmmhesaysfile: that's what cron is for
14:09.18eric_hill:)
14:09.26*** join/#asterisk IceManRISK (n=kart@201.15.214.182)
14:09.28file:D
14:09.45Hmmhesaysit's actually pretty funny, they come running with my phone when it rings
14:09.51kavit[TK]D-Fender: yeah I just gave up on the whole h323 thing for now
14:09.52[TK]D-FenderYeah you're chronic SOMETHING all right...
14:09.58kavittoo much pain
14:09.59Hmmhesaysa 1991 volvo
14:10.37[TK]D-FenderHmmhesays : Parked or mobile?
14:10.40Itburnzhrm i installed spandsp in /usr/include/spandsp... how i can check where the app_rxfax.so tries to access the t30 header ?
14:10.52HmmhesaysParked: we were driving down a residential street at 2:30am
14:11.10*** join/#asterisk austinnichols102 (n=austinni@70.46.69.131)
14:11.11HmmhesaysI gave her a pretty good shove off me and she grabbed the steering wheel
14:12.41*** join/#asterisk azzie (n=az@azzie.net)
14:13.18HmmhesaysI missed that reference
14:15.43Hmmhesayswas it funny or something else
14:16.51[TK]D-FenderHmmhesays "Stupid girls"
14:17.04Hmmhesaysahhh
14:17.22*** part/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker)
14:17.34*** join/#asterisk LoRez (i=lorez@freenode/staff/lorez)
14:17.34Hmmhesaysno guitar for 5 days, i'm going to have to learn to play all over again, lol
14:18.33*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
14:18.35[TK]D-FenderI think it fitting you start out with some Elvis :)
14:18.50*** join/#asterisk BadPacket (n=root@unaffiliated/badpacket)
14:18.55HmmhesaysI was thinking "old time rock n roll" by bob seger
14:19.05filepesky kids
14:19.07[TK]D-FenderI was thinking "Jail House Rock" :)
14:19.12HmmhesaysLOL
14:19.18Hmmhesaysi think i could play that lefty
14:20.03[TK]D-FenderYou're not "over the hill yet".... if only she hit the accelerator at the same time you'd have gained sufficient velocity :D
14:20.34Hmmhesaysok stop making references I don't get
14:20.53Hmmhesaysjail is making me dumberer
14:20.54[TK]D-FenderGoing airborne instead of stopping cold on the crash.
14:20.59Hmmhesaysahh lol
14:21.06fileairborne express!
14:21.43[TK]D-FenderI am tending to speak in the "implied" tense.  Works good with those that handle context & stream of consciousness well....
14:21.45kavitis asterisk business edition  released yet?
14:21.56[TK]D-FenderNot to be confused with "Field & Stream" ;)
14:22.01[TK]D-Fenderkavit : Serveral
14:22.04HmmhesaysI don't fish or hunt
14:22.05kaviti read Marks interview somewhere
14:22.07filekavit: it's been released for some time..
14:22.23kavitfile: i dont get out much :(
14:22.25MikeJ[Laptop]does ABE have version numbers??
14:22.31MikeJ[Laptop]kavit, get out more
14:22.33fileit's got letters
14:22.38[TK]D-FenderF
14:22.39MikeJ[Laptop]what letter are they on?
14:22.40[TK]D-Fender:O
14:22.48file... A
14:23.10fileI wonder what happens when it reaches Z
14:23.16[TK]D-Fenderfile : What is the approximate std release equivalent for it?
14:23.16MikeJ[Laptop]like hurricanes?
14:23.20HmmhesaysAA
14:23.22HmmhesaysBB
14:23.23[TK]D-Fenderfile : LIke Excel.... AA :D
14:23.31[TK]D-FenderHmmhesays : no, AB :)
14:23.34Hmmhesayssometime in 2050
14:23.40file[TK]D-Fender: of A? I have no clue
14:23.42Hmmhesayswhen hopefully i'll be long dead
14:23.48Hmmhesaysor living on a beach somewhere
14:23.50[TK]D-FenderBelieve me Asterisk will not last that long.
14:24.08[TK]D-FenderHmmhesays : Hefner seems to be doing OK for his age...
14:24.15Hmmhesaysi'm not that rich
14:24.24Hmmhesays1 he can pay for good medicine
14:24.25MikeJ[Laptop]there may not even be phones in 50 years :P
14:24.34Hmmhesays2 surrounded by beautiful women constantly
14:24.35C4T3ljust a quick question:  What's an acceptable load avg for a Xeon (dual core) running * with 200+ users?
14:24.48C4T3lsip users
14:24.56fileC4T31: that depends
14:24.58C4T3ler peers
14:25.03[TK]D-FenderC4T3l : Are they oall on coffee break?
14:25.07MikeJ[Laptop]C4T3l, till it makes stuff break it's good, once it starts breaking, it's bad
14:25.18C4T3lgood one
14:25.19Hmmhesaysyou should be able to cook an egg on your processor(s)
14:25.21fileSO - IAXtel survived the night with my latest mods...
14:25.30MikeJ[Laptop]file, I can fix that
14:25.46*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
14:25.48fileI bet
14:26.07MikeJ[Laptop]it's on how many boxes?
14:26.09MikeJ[Laptop]just 1?
14:26.14C4T3ljust 1
14:26.26MikeJ[Laptop]C4T3l, not you, iaxtel
14:26.35C4T3lsorry :)
14:26.43fileyessir, just 1
14:26.52MikeJ[Laptop]I wonder how many calls I can send to file at once
14:26.55MikeJ[Laptop]hmmmm
14:27.03kavitfile what sort of mods?
14:27.04Hmmhesaysgotta be at least 5 or 6
14:27.08BugKhamthe debug shows chan_sip.c: Failed to authenticate user "asterisk" when calling from another server
14:27.16BugKhamiax calls work fine
14:27.19*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
14:27.30filekavit: some updates... that'll get merged into trunk once they get tested by someone else who has been having troubles as well
14:28.12BugKhaminsecure=port,invite on the [general] section, what else should I look at?
14:28.14filewhomever coco_tseng is, you're not registered
14:28.15kavitah, file do you work for digium?
14:28.24MikeJ[Laptop]file broke it!
14:28.26filekavit: yes
14:28.32kavitor do you commit out of the kindness of your heart
14:28.50fileI'm commited!
14:28.52kavitheh file, planning to ask for a raise now that the business grade enterprise version is out?
14:28.58MikeJ[Laptop]kavit, he has no kindness inhis heart.. only bitterness
14:29.05ringeI have to open the firewall to be able to place two-way calls between two astersisk servers, right?
14:29.06filekavit: it's been out for awhile
14:29.10filelike, a year
14:29.30*** join/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg)
14:29.42MikeJ[Laptop]there is now a business grade enterprise version of file?
14:29.50kavitMikeJ[Laptop]: he must have tried to fix h323, enough to make anyone bitter
14:29.55fileisn't he talking about BE?
14:29.57*** join/#asterisk jahani (n=k@adsl196-213-242-217-196.adsl196-16.iam.net.ma)
14:30.00Hmmhesayswho can we get on the case we need ....
14:30.04fileif not then I have no clue what you're talking about
14:30.05MikeJ[Laptop]fix h323?
14:30.22kavitfile: I didnt see the year on the article, need sleep
14:30.30MikeJ[Laptop]go sleep
14:31.14*** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
14:32.41filesleep is evil, don't listen to him
14:33.04ManxPowerCell phone companies suck
14:33.10Hmmhesaysyes
14:33.19fileyes, they do
14:33.23kavitcertainly not, I just need to replenish the redbull content of my blood stream
14:33.34ManxPowerVerizon expects me to pat a $583 bill, which I didn't even know I had until a week ago, without any copy of the charges.
14:33.58muttoo many 1-900 numbers eh
14:33.59filethey won't send you the stuff explaining how it's $583?
14:34.02kavitManxPower: tell them very politely where they can stick it
14:34.03filenice.
14:34.05Hmmhesayswhere the hell did alltel wireless come from
14:34.16ManxPowerkavit, I'll send them a letter by certified mail.
14:34.42ManxPowerHmmhesays, they are a regional cell company in the southeast (maybe other places).
14:34.56Hmmhesaystheir commercials claim they have the largest network in the US
14:35.05ManxPowerkavit, they said that they already send it when they sent out the original bill who knows how long ago.
14:35.07Hmmhesaysand even use characters from other cell companies
14:35.08kavit"Customers are not supposed to be scared of their companies, Companies are supposed to be scared of their customers"
14:35.58kavitManxPower: thats a standard reply, their default stance is "We are always right, we can never be wrong, we have more money than you we will do what we want"
14:36.41kavitManxPower: atleast they arent partially government owned like telstra here... probably the worst telco on this planet
14:37.34mutheh
14:37.40mutcenturytel is the worst telco on the planet
14:38.16kavitwhich leads me to believe telcos suck as a rule.
14:38.47Hmmhesaysit'll be cool when I can download full length dvd's to my cell phone
14:38.57austinnichols102telcos suck (tm)
14:39.19fileHmmhesays: using multiplexing teletectonic data transfer?!?
14:39.27Hmmhesaysum sure
14:39.31Hmmhesayswhatever the fark that is
14:39.44fileit's from ImaginaryTech(tm)
14:39.45muttap your foot on the ground, morse code man!
14:39.57Ikarusfile: shaking the earth in different directions ?
14:39.58Hmmhesaysfile: cool, sounds like it has something to do with smashing your phone against a rock
14:39.59*** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk)
14:40.01IkarusAM of FM ?
14:40.51Hmmhesaysyou think you'd have better luck on am with a rock
14:41.47Hmmhesaysbut what do I know
14:42.30kavithey Tesla wanted to distribute power like that
14:42.47Hmmhesaysthey also wanted to rock your faces off
14:43.00Alystairhaha
14:43.01*** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin)
14:43.11Alystair"multiplexing teletectonic data transfer"
14:43.22I-MODmut: whats up with centurytel?
14:43.23Hmmhesaysso that would be hitting a rock with two phones at once
14:43.35brettnemwait a minute.. telcos don't suck..
14:43.39Alystairyou send faxes via earthquake
14:43.55brettnemITSPs suck
14:43.57kavitHmmhesays: i dont know which Tesla you are talking about, but Tesla had no time for music
14:44.27Hmmhesaysgo download "modern day cowboy"
14:44.41kavitah
14:44.53kavitI meant Tesla the inventor of AC current
14:45.01Hmmhesayskavit: you have no sense of humour
14:45.29brettnemyou long hair hippie people
14:45.39kavitHmmhesays: actually I have never heard of Tesla in context of music before
14:45.47Hmmhesayskavit: you're missing out
14:46.00brettnemI like tesla and not just for the coil
14:46.06Hmmhesaysthe screaming vocals of jeff keith and guitar of frank hannon and tommy scheoch
14:46.10kavithahaha
14:46.10coppicebrettnem: I used to be a long haired hippie, but my hair is shorter these days
14:46.33Hmmhesaysthey had some of the better dual lead solos of the late 80's and early 90's
14:46.47brettnemcoppice: heh, me too.. until I realized I was that "dorky long hair computer guy" and I seriously needed a change.. so I cut my hair got a wife and 2.5 kids..
14:46.51wasimb
14:46.58Hmmhesayswhere do you buy one of those
14:47.05Hmmhesaysdoes it come in a package?
14:47.10brettnemHmmhesays: Edmond scientific!!
14:47.47Hmmhesayshmm
14:48.00kavitbrettnem: did you feel like RMS ?
14:48.13LostFrogshop, even.
14:48.23brettnemHere you go
14:48.26brettnemhttp://scientificsonline.com/product.asp_Q_pn_E_3070301
14:48.32Hmmhesaysthey're so cute, little legs walking around by themselves
14:49.05Hmmhesays50,000V what a rip off
14:49.06brettnemkavit: I may have been root, and maybe even square back then, but never mean
14:49.10brettnemhaha
14:49.17brettnem"Watts up!"
14:49.24brettnempff
14:49.28Hmmhesaysgo pick up an ingnition coil out of the junkyard for 10 bucks
14:49.49Hmmhesaysnewer ones put out about 100Kv
14:50.22*** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk)
14:50.22LostFrogBuy that and your amateur radio neighbors will hate you!
14:50.38brettnemwhoo http://bellsouthpwp.net/B/u/BunnyKiller/tcoil.htm
14:51.29brettnemawesome
14:51.36coppicethe latest trick in India to avoid paying for electricity is a little box. in it is a line output stage from a TV, about to generate about 30W at 25kV. They link this to the live wire of the mains, putting massive noise on it. This blows up most of the electronics in the house, including the electricity meter.
14:52.09Ikaruscoppice: here they then proceed to bill you at the high end of "average" for the size house
14:52.36IkarusWithout giving you any oppertunity to prove you didn't do it on purpose (unless you reported it within a month or so)
14:52.46coppicehow? you have a dead electricity meter. there's no way to prove how it got like that
14:52.47*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
14:53.24Hmmhesaysunless you hit it with a 20lb sledge
14:53.36coppicethe real downside is you can't keep doing it. the occassional dead meter is one thing, but failing between every reading it quite another
14:53.53brettnemheh
14:54.02brettnemthis thing is cool
14:54.03brettnemhttp://bellsouthpwp.net/B/u/BunnyKiller/bigpig.html
14:55.05DoktorGregslash dot and mac evangalists are a funny mix
14:55.18*** join/#asterisk brif8 (n=Techno@lazyjtrainingcenter.com)
14:55.21DoktorGreg"independance day was dumbest movie ever"
14:55.29HmmhesaysI liked that movie
14:55.32DoktorGreg"But it had a mac in it"
14:55.37Hmmhesaysback when Will Smith was all the rage
14:55.53DoktorGreg"like you can write an alien virus on a mac"
14:56.08DoktorGreg"But you CAN write an Alien virus on a max!"
14:56.19DoktorGreglol
14:56.28DoktorGregthe gist of a thread on slashdot right now
14:56.42*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
14:57.04*** join/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au)
14:57.06DoktorGregThe the post that explains how you write alien viruses on Mac gets modded +4 insiteful
14:57.06HmmhesaysI go to fark for my down time
14:58.33*** join/#asterisk kink0 (n=k@62.37.205.161)
14:59.10kink0hello
14:59.43kink0I still suffering memory leakages problems with Asterisk 1.2.7 + h323, anybody can help ?
15:00.52trelane_docelmo, what's the first rule of slashdot?
15:00.58*** join/#asterisk nite (n=nite@gateway.digium.com)
15:01.30LostFrogtrelane: You don't talk about /.?
15:01.50AlystairSo, what are the nicest phones to use? I'm thinking Polycom?
15:01.55trelane_LostFrog, THATS RIGHT!
15:02.07eric_hillLostFrog: Funny, that's the same as the second rule!
15:02.10*** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41)
15:02.12*** join/#asterisk bweschke (n=bweschke@66.152.225.74)
15:02.25trelane_eric_hill, amazing how that works :)
15:02.36kavitFirst Rule of Slashdot is Do not talk about slashdot?
15:02.42eric_hillAlystair: The ones that work for what you need.
15:02.45trelane_kavit, right.
15:02.54mostyi thought the first rule of slashdot was check your brain at the door
15:03.01trelane_mosty, wrong
15:03.02Hmmhesaysdoes anyone else in here hate myspace.com
15:03.06LostFrogkavit: Watch "Fight Club."
15:03.18trelane_Hmmhesays, yes, not that that's on topic
15:03.27chapeaurougeAlystair, using polycom here too.. very nice
15:03.36Hmmhesaystrelane_: so?
15:03.51mostyi'm trying to figure out how to barge in on calls, is that possible in asterisk?
15:03.56Hmmhesayschanspy
15:04.06Hmmhesaysor a manager redirect to a conference room
15:04.11Hmmhesayseither will get you there
15:04.39kavitLostFrog: I have read it
15:04.42GerbilWrkI have an issue where two asterisk boxes are calling eachother via sip, and each have at least one g729 codec available. Their context is set to disallow all, and allow only g729. The call goes through, but not using g729
15:04.43kavitBooks are better
15:04.44mostyHmmhesays: thanks
15:05.06Hmmhesaysnp: i had to say something on topic to keep trelane_ from skewering me
15:05.16*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.66.Dial1.SanJose1.Level3.net)
15:06.05trelane_what skewer?
15:07.25*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.6.66.Dial1.SanJose1.Level3.net)
15:11.53GerbilWrkanyone have any ideas about the codec issue?
15:11.55C4T3lGerbilWrk: what codec is used instead?
15:12.21*** join/#asterisk zaf (n=zaf@dsl081-237-122.lax1.dsl.speakeasy.net)
15:12.25*** join/#asterisk Splas (n=jwb@206.252.198.101)
15:12.40GerbilWrkaccording to sip show channels, the form is g729 to the phones
15:12.57C4T3lGerbilWrk: have you attempted sip debug?
15:13.09GerbilWrkbut, show g729 shows 0/0 inuse
15:13.42mostygerbilwrk: you have registered the g729 licences with asterisk?
15:13.52*** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
15:14.10ghost99could i ask if anyone could take a look at my dialplan etc and fix a big i have , I keep getting error 404 on my softphone and can't dial out .. would be nice if someone could help me out
15:14.26GerbilWrki registered the codecs per digiums instructions, and asterisk does show there are 3 available on one server, and 1 on the other
15:16.24mostygerbilwrk: if a call is g729 at both ends, asterisk doesn't need to use any g729 licences, only the phones do. perhaps if you set one of the phones to use a different codec, you would see asterisk using a licence to transcode to and from g729
15:16.41*** join/#asterisk DrDeke (i=Rusty@causticsoda.engin.umich.edu)
15:17.44GerbilWrkso the g729 codecs are only to the phones? not server to server?
15:18.20mostyif both ends of a call talk the same codec, asterisk does not interfere
15:18.43mostyhowever if they are both talking in a different codec, asterisk will translate
15:19.04DrDekeCan anyone point me in the direction of a SIP or IAX softphone that can use G.722?
15:19.05GerbilWrkso, in reality, once the calls connected, it's just phone to phone right?
15:20.13mostygerbilwrk: depends how asterisk is setup
15:20.46noname32hey question in the log does this mean it is using codec 2? Oooh, format changed to 2
15:21.49*** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com)
15:22.47vader--have any of you guys posted anything to www.voip-info.org?
15:23.10DrDekeEver? Yes, I have.
15:25.27vader--i wanna post some info on a part number for a dell poweredge 2800 to make the digium TDM analog cards work with it
15:25.39vader--the dell poweredge 2800 and 2850 don't ship with molex connectors
15:26.01vader--so for you to be able to use FXS modules ya need this part
15:26.09vader--not sure where to put i on the site though
15:26.20DrDekehmm
15:27.03Hmmhesayswow, this itsp doesn't send a phone number lol
15:27.06Hmmhesaysfantastic
15:27.53DrDekewhich ITSP is that?
15:28.29*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
15:28.29*** mode/#asterisk [+o anthm] by ChanServ
15:29.24*** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net)
15:29.39*** part/#asterisk austinnichols102 (n=austinni@70.46.69.131)
15:32.58Hmmhesaysaxvoice
15:33.18Hmmhesaysone of the eleventy billion generic ones out there
15:33.26DrDekeahh, yeah
15:33.30Hmmhesayslevel 19
15:33.38Hmmhesaysand we're not talking D&D
15:33.54DrDekehahaha
15:34.04brettnemAX voice?
15:34.25brettnem"pay your bill, or you'll get the AX!"
15:34.32DrDekeI find that my "customers" (family & friends) could almost not care less about voice quality so long as I give them free telephone calls.
15:34.38DrDekeNevermind that I'm about tier-5 ;)
15:34.40Hmmhesaysyeah pretty much
15:34.43mutin iax
15:34.50mutif i do bandwidth=high
15:34.57mutdoes that do ulaw codec or what?
15:35.01HmmhesaysA@H doesn't deal well with names for incoming calls
15:35.21DrDekeYeah, it allows ulaw/alaw (the 64kbps payload codecs). You could just as well leave out the bandwidth line and set "disallow=all allow=ulaw allow=alaw" and whatever else you want
15:35.47*** join/#asterisk pythos (i=pythos@unaffiliated/pythos)
15:35.52pythosmornin!
15:35.59DrDekemornin
15:36.00DrDeke'
15:36.19Hmmhesaysboobies
15:36.26Hmmhesays(.)(.)
15:36.32vader--do they just let anyone modify anything on voip-info.org's wiki?
15:36.38Hmmhesaysif you're signed up
15:36.47*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
15:36.49vader--what happens if someone delete's something?
15:37.00VoicePulseIt keeps a history of changes that can be undone.
15:37.07vader--gotcha
15:37.08DrDekevader-- The wiki software keeps track of all changes that people make, and it is trivially easy to undo vandalism.
15:37.32vader--i found a section for me to put that info
15:37.36vader--http://www.voip-info.org/wiki/view/Asterisk+hardware
15:37.43pythosok, so I got my TDM400P with 2 fxo and 2 fxs's installed, and such, as per report by DMESG.  Now what do I do?  <ok, Im lost in terms of what to do next, but I DID at least get the card recognized, etc.>
15:37.46*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
15:38.22*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
15:38.29vader--did you modify your zaptel.conf to setup the ports?
15:38.52vader--ya have to tell zaptel.conf what ports are fxs and what ports are fxo
15:38.57pythosnope.  I just barely got the hardware stuff figured out.
15:39.31pythosIs there a good primmer, not to techie?  Im kind of slow, as its Monday here all week long
15:39.43HmmhesaysI'll tell you my dirty little secret
15:39.50Hmmhesaysdon't tell anyone or you'll be just another regret
15:40.15pythosI got fxs on moduels 0 and 1, and fxo on 2,3
15:40.23*** join/#asterisk techie (n=gus@antibala.com)
15:40.25puzzledhi
15:41.06jake1932dirty little secret
15:41.18SpaceBasspythos, nerdvittles.com and www.archatechs.com both have good primers up currently... focused on asterisk@home but the concepts are the same
15:41.34Hmmhesaysok there is something confusin in this freepbx dp
15:41.44Hmmhesaysexten => s,1,Set(FROM_DID=s)
15:41.45Hmmhesaysexten => s,n,Goto()
15:41.45Hmmhesaysexten => _X.,1,Goto(ext-did,s,1)
15:41.48Hmmhesayswhy does that work
15:42.06Hmmhesaysand the first person to give me shit about a 3 line paste gets their foot nailed to the floor
15:42.30Hmmhesaysj/k //not really ///but kinda
15:42.37jake1932what's the purpose of 2?
15:43.06Hmmhesaysit gets longer hold on
15:43.39Hmmhesayshttp://pastebin.ca/52584
15:44.11Hmmhesaysthe call comes in _X. catches it, sends it to s,1
15:44.42Hmmhesaysnevermind I figured it out
15:45.53*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
15:46.11*** join/#asterisk Tagor (n=Tagor@s55928c6d.adsl.wanadoo.nl)
15:46.14TagorHi
15:46.31TagorI'm using asterisk (with an external SIP provider) and grandstream phones
15:46.53TagorNow I would like to make a script that first calls the grandstream and then onces it's picked up, it should call an external number
15:47.04TagorIs there an easy way to do that?
15:47.36wasimcall files baby
15:47.59jake1932exten => s,n,Goto()?
15:50.01TagorThanks wasim
15:50.27*** join/#asterisk shaynes (n=shayne@c-67-161-190-26.hsd1.ca.comcast.net)
15:51.10*** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net)
15:51.27shaynesCan anyone help? I am setting up an IVR using Asterisk@Home (latest-ver) and I am having trouble using a WAV as the announcement. Are there any special requirements of the WAV file?
15:51.42wasimshaynes: mono, 8khz, 8 bit
15:51.56CunningPikeshaynes: What he said
15:52.05CunningPikeJust beat me to it lol
15:52.48shayneswasim: no way.
15:52.50CunningPikeWow
15:52.50[TK]D-Fendershaynes : Please read the channel topic.
15:53.22shaynes[TK]D-Fender: you're too kind.
15:53.32[TK]D-Fendershaynes : Not possible :)
15:53.38CunningPikeIs that channel new?
15:53.46CunningPikeI hadn't noticed it before
15:53.47shaynes[TK]D-Fender: figured. ;)
15:54.05Katty[TK]D-Fender: we should talk :P
15:54.18Katty[TK]D-Fender: this isn't asterisk related.
15:54.33[TK]D-Fendershaynes : Sorry if I sounbd abrupt... you did come in kinda cold without so mcuh as a "hello all", and flew right into the "grey" topics...
15:54.45[TK]D-FenderKatty : :O.  Always open.
15:54.46Hmmhesayswow this version of freepbx is retarded
15:54.56[TK]D-FenderKatty : and "mew", and pm if you like
15:54.56shaynes[TK]D-Fender: "grey?" -- open source, open mic?
15:54.59Katty[TK]D-Fender: yeah well it'll take a bit.
15:55.05[TK]D-FenderKatty : Sure
15:55.07Kattyk
15:55.10Kattyalso! mew
15:55.25shaynesHello All!
15:55.45[TK]D-Fendershaynes : Grey in the sens that all things AMP related tend to lead towards being shuffled off.
15:56.17blitzragehas anyone else been experiencing VM in 1.2.7 deleting the audio files, but leaving the .txt file behind?
15:56.30pythosHmm, anyuone help me with auto-ignore problem in bitchx?
15:57.17*** part/#asterisk shaynes (n=shayne@c-67-161-190-26.hsd1.ca.comcast.net)
16:01.23sevardI'm having a little issue with DTMF on my IVR.  It seems I have to dial extensions at the prompt very slowly
16:01.32*** join/#asterisk miguel3239 (n=miguel32@ns1.nashuacs.com)
16:01.34CunningPikeblitzrage: Very occasionally, and I can't remember what caused it
16:01.53CunningPikeIt hasn't happened for ages
16:02.00sevardI have
16:02.02sevardexten = s,4,Set(TIMEOUT(digit)=3)
16:02.03sevardexten = s,5,Set(TIMEOUT(response)=15)
16:02.08sevardIs that not suggested?
16:02.09blitzrageCunningPike: yah, we're talking about it in dev -- seems to be in 1.2.7 -- is that what you're running?
16:02.26CunningPikeWe're running 1.2.1 in production right now
16:02.35CunningPikeTest is 1.2.7.1
16:03.14CunningPikeThis would have been quite a while back for us - maybe pre-1.2 - and I think it was something we were doing rather than something in *
16:03.33CunningPikeWe have our vm files on a mounted share
16:04.00CunningPikesevard: We use timeouts of 2
16:04.10CunningPikeOtherwise you have to wait or press #
16:04.24*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
16:04.55sevardCunningPike: digit and response?
16:05.01*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
16:05.10CunningPikesevard: Yes
16:05.20*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
16:05.56CunningPike15 is way long
16:06.15CunningPikeMorning wunderkin
16:06.50sevardCunningPike: I get the issue, even with the 2 time out when I enter in extension 2003 I get invalid extension 203, or invalid extension 00, or invalid extension 003, etc.
16:07.01pythosk
16:07.05CunningPikeAh - that's a DTMF problem then
16:07.15*** part/#asterisk mosty (i=mostynm@adsl-137-244.swiftdsl.com.au)
16:07.18sevardCunningPike: right :)
16:07.21CunningPikeWhat type of DTMF signaling are you using
16:07.29sevardCunningPike: I am unsure how to find out
16:07.33Hmmhesaysthis freepbx seems to be using a query string that I cannot find in any of the source files
16:07.41wunderkinhey
16:07.42CunningPikeWhat phones are you using
16:07.52sevardCunningPike: I have some calls originating from ATAs, from some SIP phones, some cell phones and some land line phones from the pstn
16:08.15CunningPikesevard: And they're all exhibiting problems?
16:08.26sevardCunningPike: mostly the ones from the cellphones
16:09.13CunningPikesevard: We've noticed that a couple of times as well - not very often though....
16:09.18*** join/#asterisk CoffeeIV_ (n=CoffeeIV@64.149.168.97)
16:09.33CunningPikesevard: Let me check something
16:09.40sevardCunningPike: it happens very often with the cellphone, when a cell dials in 90% of the time they can't ring the extension because of this problem
16:10.06CoffeeIV_I have an asterisk built from source from CVS a while ago,and I am trying to figure out exactly which verison it is -- include/asterisk/version.h doesn't seem to have anything useful in it, how can I tell ?
16:10.44sevardasterisk -V
16:10.45CunningPikesevard: In your sip.conf, what codecs are enabled?
16:10.56sevardCunningPike: only ulaw
16:11.06CunningPikeAh - try allowing gsm as well
16:11.09sevardCoffeeIV_: asterisk -v
16:11.12CoffeeIV_asterisk -V gives "Asterisk CVS-HEAD" which isn't reall helpful
16:11.13sevardCoffeeIV_: asterisk -V
16:11.14*** join/#asterisk lzhang (n=lewiszha@67.95.13.46)
16:11.19sevardCunningPike: why gsm?
16:11.31lzhangis sip.conf the only place where I specify codecs?
16:12.07CunningPikesevard: Cell phone calls are gsm? Not sure of the exact reasoning, but it works for us :D
16:12.21dlynesCunningPike: cell phone calls are gsm v4 though
16:12.27dlynesCunningPike: asterisk i think uses gsm v2
16:12.47CunningPikedlynes: Would that cause transcoding though?
16:12.55dlynesno idea
16:13.05dlynesI just know asterisk doesn't use gsm v4 because it's patented
16:13.29CunningPikeOK
16:13.37sevardCunningPike: i'm not sure if that's helping ;/
16:13.40dlynesThe compression on gsm v4 is significantly greater, while still allowing better quality
16:14.15sevardCoffeeIV_: I'm not sure, try xdd asterisk | less and see if you can find anything
16:14.18sevardCoffeeIV_: or strings
16:14.22sevarderm, xxd
16:14.25dlynesCoffeeIV_: how about show version in the CLI?
16:14.27*** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com)
16:14.55dlynesJust realized
16:14.57CoffeeIV_show version at the *CLI> gives "CVS-HEAD" again
16:15.08sevardCoffeeIV_: strings asterisk | less, scroll down, see if you get what i get
16:15.09dlynesIf it's CVS HEAD, he must be running 1.0.9.2 or earlier
16:15.17CunningPikesevard: Are these all on a PRI?
16:15.20sevardCoffeeIV_: or just download a new version
16:15.27sevardCunningPike: is what on a PRI?
16:15.30wunderkinyeah, does it really matter? it is going to be old anyway
16:15.57CunningPikesevard: Your troublesome DTMF calls - I'm wondering is the gain screwing you up
16:16.36sevardCunningPike: It's a bit more complicated than that :) I have sipura ATAs coming into to a TDM400P
16:16.39dlynessevard: is your dtmf problems occurring with at least one leg on pstn, pri, or t1?
16:16.51dlynessevard: yeah, i guess it is :)
16:17.11sevarddlynes: I'm not sure what you're asking or how to give you those answers.  school me :)
16:17.14dlynessevard: try cunningpike's suggestion then...it's probably your gain control
16:17.19CoffeeIV_strings can't give me anything that isn't in the #defines in the code, which seems to "CVS-HEAD" . . .. I found a reference to version 1.2 in UPGRADE.txt, I guess that is best I can do
16:17.21sevardWhere is my gain?
16:17.29dlynessevard: zapata.conf
16:17.31CunningPikeIn zapata.conf
16:17.37dlynessevard: that's for your tdm400p
16:17.46sevardrxgain=12.0
16:17.46sevardtxgain=3.0
16:17.46CunningPikeDamn I gotta drink more coffee
16:17.47dlynessevard: your sipura also has a gain control, I think
16:17.52brodiemwhat is the recommended codec for SIP over broadband?
16:18.03dlynesbrodiem: g729
16:18.13sevardit matters how much broadband you have
16:18.15CunningPikesevard: Also try dtmfmode=rfc2833 for your ATA entries in sip.conf
16:18.22wunderkinCoffeeIV_: what is your reason for finding out?
16:18.25brodiemsevard well just as a generalization
16:18.32CunningPikeAnd make sure the ATA is set appropriately also
16:18.51dlynessevard: oh yeah...one other thing, too...if you're using sipuras, make sure you disable auto dtmf detection on them
16:18.55brodiemdlynes, is that the codec thatrequires a paid license?
16:19.00CunningPikedlynes: Good point
16:19.02sevardWill do.
16:19.23dlynessevard: they have a major problem with correctly detecting dtmf...you often get dtmf codes in your conversation cause it thinks it hears dtmf
16:19.31sevardhahaha
16:19.37viperdudeukhow do i stop outgoing calling from voicemail?
16:19.40dlynessevard: and even the newest firmware doesn't fix it
16:19.49sevardthe sipuras that come into the tdm400p aren't defined in sip.conf
16:19.56dlynessevard: it occurs on both sipura 2000's and sipura 3000's
16:20.05CunningPikesevard: Right - sorry
16:20.06sevardYeah, I have 2002s
16:20.07dlynessevard: not to mention sipura 2002's and pap-2's
16:20.24Hmmhesayswow that version of freepbx had gone retarded
16:20.50Hmmhesaysmissing a freaking column in the database
16:20.52dlynessevard: 2002 and pap-2's are just 2000's with a different cover and newer default firmware
16:20.54Hmmhesaysi had to manuall add it
16:21.16sevardHmmhesays: why do you use that crap
16:22.06dlynesbrodiem: yes...if you want to avoid codec licensing, try ilbc, but it requires more cpu power, albeit, it consumes less bandwidth than g729
16:22.23CoffeeIV_wunderkin: my reason for finding out is that someone asked me . . . they seemed statisfied when I told them "probably 1.2", and I'll be installing a newer version in a few weeks anyway
16:22.37brodiemdlynes, thanks
16:22.46*** join/#asterisk Tier_1 (n=Tier@c-24-9-75-234.hsd1.co.comcast.net)
16:23.06CoffeeIV_but some process should put a version number, or at least a date, in version.h when you check into CVS, or something
16:23.59CunningPikebrodiem: You could _try_ ulaw - I'm having success with ulaw on IAX - ymmv
16:24.15*** join/#asterisk SparFux (n=player@e182016205.adsl.alicedsl.de)
16:24.22sevarddoes anyone know off hand how to tell ZAP trunk to give you a dialtone on one line?
16:24.30Hmmhesaysnothing wrong with freepbx
16:24.39Hmmhesayswell.. not nothing, but it is useful
16:25.14sevardlike exten = 400,1,(ZAP/1)
16:27.21brodiemCunningPike, yeah it just seems ulaw has some shakiness sometimes from periodic laginess
16:27.42*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
16:27.49sevardexten = 400,1,DIAL(ZAP/1)
16:27.51brodiemCunningPike where dropped frames cause a bit of broken speech
16:27.51CunningPikeYes - it does sometimes, but I find it quite useable
16:27.53sevardrock
16:28.30CunningPikebrodiem: It depends on your connection - no QoS here :D
16:28.48brodiemCunningPike, I just figured it would be better off to use something that has a bit more overhead but shorter intervals and packet sizes to make up for it
16:29.46CunningPikeWhat UA are you using?
16:30.28brodiemCunningPike I find ulaw quite usable too just picky :)
16:30.36*** join/#asterisk copland (n=stonecol@209.216.65.10)
16:30.38CunningPike:)
16:30.39brodiemjust annoys me when I hear any little breakups
16:30.51*** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin)
16:30.51coplandhello is there a room for asterisk at home on freenode
16:30.59CunningPikeYa - I can live with it.
16:31.03brodiemcopland, look at topic
16:31.24CunningPikebrodiem: I work from home one day a week and use an IAXy with ulaw
16:31.30CunningPikeIt works
16:31.31coplandsorry irrsi burried the topic I see it now
16:31.53CunningPikeIAXy doesn't do gsm, and I've only enabled ulaw and gsm on our sever
16:32.36mutther any way to bridge a call on my pri and free both channels it's using?
16:32.42mutsay someone calls in on 1234
16:32.45muti call out on 4321
16:32.51mutcan i bridge that and drop those channels?
16:33.05mutor will * need to stay in between
16:33.41CunningPikemut: Can't see how - your PRI channels are your connection to the POTS - unless I misunderstand the question
16:33.42jsharpThere's some basic support for that in libpri.  Dunno how well its tested, though.
16:34.14Hmmhesayscarlos mencia is hilarious
16:34.20SparFuxHow can I run a script or something whenever a special user logs in to asterisk via sip?
16:34.48coplandhas anyone managed to get stanaphone service to work with asterisk?
16:35.17wunderkinmut, 2 b channel transfer?
16:35.52Hmmhesaysdoesn't look very complicated copland
16:36.10brodiemSparFux I would think you'd need something listening for manager events to trigger a script, otherwise a system call when a dial plan gets used for that SIP ext
16:36.47SparFuxbrodiem: So far no hook available for that, I suppose? :-(
16:41.22lunkis there a return type command that will send you back to the calling context?
16:41.42ManxPowerYay!  I'm getting 3,000 ft of water pipe for $250!
16:41.46lunka goto from ctxt1 sends you to ctxt2 where you do something and want to return
16:41.51trelane_ManxPower, why?
16:42.00mutwunderkin: yae
16:42.05ManxPowertrelane, conduit to bury cable
16:42.18trelane_ManxPower, a valid use of your time, I approve, carry on.
16:42.19websaeCVC?
16:42.28websae1/8inch?
16:42.56websaewhoops...mean 1'' 1/4
16:43.27mut1/8... YEA!
16:43.51websaehaha...find like maybe a twisted pair in that
16:43.52websaehaha
16:43.56mutfew strands of fiber
16:44.00mutplenty enuf
16:44.10websaehaha
16:45.49*** join/#asterisk key2 (n=key2@gob75-2-81-56-64-17.fbx.proxad.net)
16:46.16techman97_andyhey all - I'm trying to put a "ring" tone across incoming lines - I'm doing "exten => (my phone #),1,PlayTones(ring)", the CLI shows the command executing, but I do not hear the ring sound in the phone - what am I doing wrong?
16:47.05*** join/#asterisk mrtwister|mobile (n=andrius@cable-10-68.cgates.lt)
16:47.27viperdudeuktechman97_andy: you need to answer the channel first
16:47.33techman97_andylemme try that quick
16:47.50pythosanyone care to comment on using Debian for platform to run asterisk?  I am trying to decide what release to use <stable, testing, unstable> ??
16:48.19viperdudeuki use debian stable without a problem
16:48.35pythosand version 1.0.7 then?
16:48.45pythos<asterisk>
16:48.47techman97_andyviperdudeuk:  I did the answer command, but I still do not hear my ringtones...=(
16:49.24*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
16:50.34viperdudeukso you have answer as priority 1 and playtones as priority 2?
16:50.52techman97_andyanswer = 1, wait = 2 (just to test), playtones(ring) = 3
16:51.02*** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net)
16:51.13sevardWhat sort of PRI signaling does * do? Ground/wink/mediate
16:51.14viperdudeukanything after 3?
16:51.19*** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net)
16:51.30sevardor is that card dependant
16:51.33techman97_andyyes, I have a background menu sound play
16:51.35techman97_andythat works -
16:51.42techman97_andypress 1 for...2 for...
16:51.44bkw__sevard, um
16:51.46viperdudeukpythos: I use 1.2.7.1 but all versions I have tried are ok
16:51.49bkw__sevard, those are not PRI
16:51.58sevardbkw__: what are they
16:51.58techman97_andythe call gets picked up...the commands fly by in the CLI, and then my menu plays
16:52.12bkw__sevard, CT1 signalling, Inband
16:52.24viperdudeukso the playtones dont fire because it immediate goes to the background command
16:52.43techman97_andyno, priority 4 is the background command
16:53.06pythosviperdudeuk: 1.2.7.1 is the latest, I think.  Not sure of which Debian release it might be in yet, or if that release would be more/less desireable than stable/with 1.0.7
16:53.28viperdudeukyes i know but the playtones will only work until another command plays audio, the background plays straightway... try putting a wait after the playtones
16:53.31sevardbkw__: I'm looking at getting a PRI and that's what my telco asked me what my * box does
16:53.32techman97_andyviper:  I can put wait commands between the playtones, but still nothing plays.
16:53.48bkw__sevard, no what switch types do they support first
16:53.49bkw__then you can pick one
16:53.51viperdudeukdo you have a ring tone defined?
16:53.55bkw__chances are asterisk does it
16:54.08viperdudeukpythos: i always compile from source
16:54.10bkw__technically its not asterisk its libpri that does it but we aren't splitting hairs here
16:54.21pythoswhat is the GNU GUI for asterixk?
16:54.28*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
16:54.28techman97_andyhow do you define a ring tone?  I thought was just a sound file * played?
16:54.49viperdudeukindications.conf
16:54.51*** join/#asterisk MooingLemur (n=troy@shells200.pinchaser.com)
16:55.08viperdudeukthere is no gui for *
16:57.17DaminIs there an agenda for tommorow's con call yet?
16:57.32lzhangguys, any ideas about what this debug message means: channel.c: Didn't get a frame from channel: SIP/8113-9e81
16:57.35pythoshuh, thought I had read of a bunch...
16:57.42lzhangI'm getting one way audio problems and I think it may be related
16:57.53viperdudeukthre are some web interfaces
16:58.01pythosoh, ok
16:58.41*** join/#asterisk ramo (n=ramo@59.92.137.242)
16:59.02viperdudeuklzhang: firewall / NAT issue?  one way audio is usually one of those
16:59.04pythosviperdudeuk: thus requiring something like apache? Does it call for https: ?
16:59.41viperdudeukpythos: not if you dont require it.... Asterisk @ Home is a full asterisk based distro with a gui
17:00.56pythosviperdudeuk: oh, so, getting debian all set is not required I guess... Wonder why I didn't go that route in the first place, now that I have debian/sarge all configured for my tdm400p
17:01.42viperdudeukpythos: lol ok, not sure if A@H handles the tdm400p, you would need to check first
17:03.17pythosviperdudeuk: the package said digium tdm22b, but it is recognized by my kernel dmesg as tdm400p 2fxs/2fxo
17:03.29*** join/#asterisk MstlyHrmls (n=mh@melbourne.mostly-harmless.ca)
17:03.33[TK]D-Fenderpythos : thats exactly right.
17:04.46CunningPikesevard: Your telco may be talking about ESF/b8zs signalling
17:04.54CunningPikeThat's the norm in NA
17:05.08sevardCunningPike: actually EFF/B8ZF
17:05.25techman97_andyviperdudeuk:  ok - I'm back.  If I dial in on my ZAP line, I get the ring tone.  Calls that come across my SIP peer don't.
17:05.35sevardoh, she said EFF, made her repeat, echo foxtrot foxtrot, it's ESF?
17:05.47justinu|laptopESF is correct
17:05.47CunningPikesevard: OK - ours is what I said :)
17:06.09viperdudeuktechman97_andy: the SIP calls hit the same exten?
17:06.41techman97_andycorrect - the only difference between the two is that the SIP calls come in a seperate context, but I execute a GoTo(s,1) in that context
17:06.50CunningPikesevard: No-one ever went broke underestimating the intelligence of telco customer service agents
17:06.56rajiv|workon my PRI when i call out to my cell phone i see my calling #, but not when i call an analog line that has working caller id. anyone know why?
17:07.09sevardCunningPike: haha
17:07.12CunningPike:D
17:07.18lzhangviperdudeuk: it's not firewall/NAT, no NAT and I turned off the firewall temporarily
17:07.47viperdudeukyou are specifying the context in the goto?
17:07.47lzhangI think it may be a g729 issue, how do I turn this codec off? just commenting out the allow line in sip.conf is not working
17:07.55techman97_andyyes I am - sorry 'bout that.
17:08.17*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
17:08.23viperdudeuklzhang: disallow = all  allow=ulaw or what ever codec you want
17:08.33CunningPikerajiv|work: We are continually amazed by the inconsistency of CID presentation - inbound as well as outbound. Our telco hasn't been able to explain it either
17:08.37viperdudeukdo the sip channels hear the background after?
17:09.00techman97_andyviper:  SIP channels hear the background, yes.
17:09.06lzhangviperdudeuk: yeah that's what I tried but for some reason I'm still getting no compatible codecs with Xlite
17:09.22brodiemany recommendations on an origination provider (sip and/or iax) with good toll free inbound rates?
17:09.25viperdudeuki use ulaw ok with x-lute
17:09.31viperdudeuki use ulaw ok with x-lite even
17:09.31techman97_andyviper:  the only thing that is different to the end user is that SIP calls cannot hear the ring tone.
17:09.32*** join/#asterisk trbldwine (i=trbldwin@vpn163245.vpn.northwestern.edu)
17:09.43viperdudeuktechman97_andy: weird
17:09.45brif8how can I disconnect from the console a call on SCCP which is hung
17:09.49techman97_andytell me about it!  =)
17:10.15viperdudeukbrif8: soft hangup SCCP/callid
17:10.22brif8sccp show lines shows that the phone is connected yet the phone is off
17:10.26poisonerhmmm
17:10.39poisonerbrif8 reload sccp
17:11.33*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
17:11.52viperdudeukno reload on chan_sccp have to unload then load again which cuts off all sccp channels
17:12.23*** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com)
17:12.38ManxPowerperhaps you are using bindaddr?
17:13.07brif8viperdudeuk: I get SCCP/100-0000003 is not a known channel
17:13.13SpaceBassI have a cisco 7940 that is behaving strangely
17:13.26brif8poisoner: sccp reload is not yet implemented
17:13.35brodiemVoicePulse no I'm in the US, but a lot of calls come from canada
17:13.51viperdudeukbrif8: looks like chan_sccp is crashed, need to unload chan_sccp followed by load chan_sccp
17:14.03SpaceBassI'm using a modified POE cable from the wiki and it powers on fine, but says "ethernet disconnected" ... i get the same result if i use a standard able and a cisco power cube...however I have ONE port on an older switch on which it does work
17:14.11SpaceBassand I cannot for the life of me figure out what is different about that port
17:14.16viperdudeukchan_sccp crashes a lot which is why i ditched it
17:14.19brif8unload sccp
17:14.19brif8Unable to unload resource sccp
17:14.23*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
17:14.30*** join/#asterisk trbldwine (n=trbldwin@vpn163245.vpn.northwestern.edu)
17:14.30viperdudeukbrif8: ouch
17:15.23harryvvI have mailed the digium news groups for a question about the messages button on the ip500 and got no responce. Anyone here own a ip500 and have been able to configure the xml part of it to make this feature work?
17:15.32harryvvSame with the conferance button.
17:15.57brif8viperdudeuk: do I have to reload *
17:16.21ManxPowerharryvv, you read the admin buide?
17:16.24CunningPikeharryvv: Polycom IP500?
17:16.34SpaceBassharryvv, what does the XML part do?
17:16.35ManxPowerbuide == guide
17:16.45*** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
17:16.45jaigerharryvv, I have one
17:16.50ManxPowerSpaceBass, Polycoms use XML config files
17:16.55TripleFFFFFFFFFFcan one help me with realtime ?
17:17.04TripleFFFFFFFFFFseems it doesnt read it or something
17:17.09TripleFFFFFFFFFFi get 404 from the debug
17:17.18SpaceBassah
17:17.20*** join/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net)
17:17.34harryvvjaiger cool. did you get the conferance a messages button to work?
17:17.35*** join/#asterisk wulfstan1 (i=nclarey@cpc3-cmbg1-0-0-cust887.cmbg.cable.ntl.com)
17:17.54[TK]D-Fenderharryvv : Read the admin guide... they spell out how to do it pretty clearly.
17:17.58jaigerconference works out of the box
17:18.00viperdudeukbrif8: you can try that or you might have to kill asterisk and restart
17:18.02jaigermessages...
17:18.10[TK]D-FenderYup....
17:18.17brif8viperdudeuk: okay
17:18.19harryvvI press conferance button and no reply
17:18.23harryvvno responce that is
17:18.30ManxPowerBTW, I finally got pricing info from my reseller for polycoms.  /msg me if you were looking for that info
17:19.01jaigerharryvv, msg.mwi.1.callBack="*86" msg.mwi.1.subscribe="1" msg.mwi.1.callBackMode="contact"
17:19.01*** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com)
17:19.10ghost99deos the centos ISO have an FTP server ?
17:19.14SpaceBassVoipsupply.com still hasn't shipped my wip330 and they had the nerve to call me about volume pricing
17:19.20[TK]D-Fenderharryvv : to conference > be on a call >  press "conference" > call the 2nd party. > they answer >press conference again > Done
17:19.24jaigerharryvv, where callBack is the extension to call to check voicemail
17:19.32harryvvI see
17:19.40wulfstan1My TDM400P won't detect hangups on my NTL line in the UK. Has anyone had any success getting this to work?
17:19.50SpaceBassghost99,  not sure its enabled by default, but it has ssh so you can use sftp and scp
17:19.51justinu|laptopntl?
17:19.52jaigerharryvv, I also set bypassInstantMessage="1" to skip a menu
17:19.57[TK]D-Fenderharryvv : And the default callbackmode is "Registration", not "contact"
17:19.58wulfstan1NTL is a provider in the UK
17:20.02justinu|laptopah
17:20.02[TK]D-Fenderyup
17:20.04wulfstan1NTL and BT are the two major telcos
17:20.11justinu|laptopbt i know about
17:20.15[TK]D-Fenderghost99 :PM
17:20.18wulfstan1BT apparently works fine
17:20.22wulfstan1NTL does not
17:20.23justinu|laptopdo you know how your telco signals disco?
17:20.24CunningPikeCan someone run slabtop on their * box and see if they have really high numbers for size-32
17:20.30wulfstan1Nope
17:20.36justinu|laptopis it a drop in loop current, or possibly a polarity reversal?
17:20.36wulfstan1How can I find out
17:20.38viperdudeukwulfstan1: is it NTL ISDN?
17:20.41wulfstan1Nope
17:20.44wulfstan1It is NTL analog
17:20.58justinu|laptopif you could get in touch with one of the switch techs perhaps
17:20.59viperdudeukok ok
17:21.05wasimCunningPike: 140k
17:21.14justinu|laptopperhaps call the trouble reporting number, make up some problem to get a knowledgeable tech on the line
17:21.17justinu|laptopthen ask
17:21.24viperdudeuklol you have never tried NTL customer services
17:21.26CunningPikeThanks, wasim - how many objects
17:21.29wulfstan1I could always just use a multimeter
17:21.39fourcheez-awaydon't both with ntl customer services
17:21.40wulfstan1It's probably more likely to tell me what I need to know than NTL customer service
17:21.43justinu|laptopyou might be able too, but it could happen fast enough you won't see it
17:21.49wulfstan1Ahh
17:21.50wulfstan1Right
17:22.01fourcheez-awayyou need an oscilloscope
17:22.05justinu|laptopor they may not be sending any signalling
17:22.05wulfstan1Eeep
17:22.05wasimCunningPike: 113
17:22.05harryvvMan, polycom came up with some kind of wireless communicator that links with skype.
17:22.08viperdudeukwulfstan1: it will have more of a clue than CS
17:22.17justinu|laptopwulfstan1: call someone, then ask them to hangup... what happens?
17:22.19harryvvI wonder if asterisk can link with skype
17:22.23justinu|laptop(use a test phone)
17:22.33wulfstan1Well, I'll try it with my mobile
17:22.34CunningPikewasim: So, I guess 2241365 is a little unusual, then
17:22.34wulfstan1Two secs
17:22.36*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
17:22.37CunningPike:D
17:22.40generalhanWhats up all !!
17:22.41viperdudeukharryvv: i think there are some SIP gateways to skype around
17:22.41wasimCunningPike: indeed
17:22.43sevardO cam
17:22.46techman97_andyhey all:  new issue.  I just installed a GrandStream GXP as a SIP client on my * server.  With the xLite softphone, we can do multi-line conferencing just fine.  If the Grandstream says, "no compatible codecs".  The Grandstream can make and receive calls without any other issues...Any ideas?
17:22.53sevardI can't find this.  What does ESF stand for?
17:23.14fourcheez-awayeat some figs
17:23.28sevardOh, that makes sense.  Yes.  My PRI wants figs!
17:23.28CunningPiketechman97_andy: Using g729?
17:23.34techman97_andyg729 end to end
17:23.47wasimmmmh ... figs and mozarella cheese, baked in the oven
17:24.03justinu|laptopinteresting combo
17:24.18CunningPiketechman97_andy: I believe that * Meetme can't do g729 because * doesn't have the necessary licensing
17:24.21ManxPowersevard, eXTENDED sUPER fRAME
17:24.21sevardwasim: that sounds lime some nasty styff
17:24.42generalhanDoes anyone know of a way to test a SIP channel to see if it is active BEFORE sending a call to it?
17:25.06jsharp* meetme can do g729 IF you have a G729 codec license.
17:25.10[TK]D-Fendergeneralhan : "show application chanisavail"
17:25.12fourcheez-awaysevard: http://www.google.com/search?q=define%3A+esf&start=0&ie=utf-8&oe=utf-8&client=firefox&rls=org.mozilla:en-GB:unofficial
17:25.14wasimsevard: try a search on google, lots of places are now doing it
17:25.20viperdudeukgeneralhan: a sip peer on the * box or in general?
17:25.23wulfstan1Ok
17:25.27generalhan[TK]D-Fender: well ... i mean in extensions.conf
17:25.27brif8has anyone else experienced SCCP hanging  I've even tried sccp reset DEVICEID  it also did not work
17:25.33wulfstan1So I just tried calling my line from my mobile
17:25.40[TK]D-Fendergeneralhan : Thats the command
17:25.43wulfstan1I get nothing for 5 seconds when I hang up
17:25.43sevardwikipedia failed me on the acroynm but wins all typed out
17:25.44[TK]D-Fendergeneralhan : Look it up
17:25.49wulfstan1and then I get a continuous tone
17:25.57viperdudeukbrif8: we had chan_sccp crashing *
17:26.00generalhanIm still trying to find a way to turn off call waiting on these stupid aastra phones. and this is my next solution to try
17:26.08justinu|laptopwulfstan1: ok... you can probably program asterisk to listen for that tone, and detect a hangup
17:26.12brif8viperdudeuk: what was your solution ?
17:26.20justinu|laptopwulfstan1: not the best way to do it, but it maybe the only choice you have
17:26.24viperdudeuki switched from SCCP to SIP lol
17:26.27wulfstan1Yeah, it's like 9 seconds
17:26.31wulfstan1Before it starts the tone
17:26.41wulfstan1Ok, so what do I need to do
17:26.56justinu|laptopthere's something called "callprogress=yes" that might do it
17:27.15wulfstan1I tried that and it didn't seem to do anything
17:27.32wulfstan1Although to be fair it may have just been waiting for the 9 seconds to timeout before it hung up
17:27.33justinu|laptopmaybe your locale needs to be modified
17:27.40wulfstan1I have set it to UK
17:27.41justinu|laptopasterisk needs to know what tone to listen for
17:27.51justinu|laptopmaybe that tone isn't defined in UK
17:27.57wulfstan1Entirely possible
17:28.00justinu|laptopi'm not an expert on this type of disconnect supervision
17:28.14brif8viperdudeuk: but the Cisco 7920 is not SIP capable yet or is it ?
17:28.17wulfstan1So let me get this straight
17:28.22justinu|laptopi usually use ISDN, CAS trunks, SS7 or SIP
17:28.36*** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
17:28.37rajiv|workCunningPike: i wonder if my cell carrier is using ANI and not CID
17:28.37justinu|laptopdisconnect and answer supervision sucks on analog lines
17:28.39wulfstan1Usually a terminal can figure out that the line has hung up because of some voltage condition
17:28.46viperdudeukbrif8: not sure i use 7940's and 7960's which have SIP firmware
17:28.53justinu|laptophere in the US, we use "kewlstart"
17:29.01justinu|laptopwhich is a 500ms drop in loop current
17:29.02wulfstan1But you can fall back on tone detection
17:29.19CunningPikerajiv|work: It's possible..... who is your carrier?
17:29.20wulfstan1Right
17:29.40viperdudeukBT signalling for analog is signalling=fxs_ks
17:29.46justinu|laptopi remember reading that some UK providers use polarity reversal
17:29.48rajiv|workCunningPike: conversent on the PRI, tmobile for cell, RCN for analog phone (all in boston, ma, usa)
17:29.50wulfstan1Yup, I realise that works for BT
17:30.04wulfstan1I was going to experiment with that on a friend's line tonight
17:30.16wulfstan1So if it's polarity reversal, how would I enable that
17:30.53CunningPikerajiv|work: Is your CID consistent on incoming calls? We were told by our telcos that CID handling from one network to another is inconsistent, but I'm not really buying that
17:30.55justinu|laptophttp://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html
17:30.58justinu|laptopcheck that link wulfstan
17:31.23IkarusCunningPike: CID can be inconsistent, with the leading 0 being dropped by some, but not others here
17:31.30rajiv|workCunningPike: incoming to the PRI works fine. incoming to the analog i've never had problems until i started calling from my own pri
17:32.07rajiv|workshould i be sending 10 or 11 digits for CID on the pri ?
17:32.23ManxPowerrajiv|work, In the USA 10 digits.
17:33.26ManxPowerThe leading 1 is NOT part of the callerid or phone number, it's a toll access code.
17:33.26CunningPikeIkarus: We get inconsistencies around whether we get CID name at all or not - sometimes we get name, the majority of the time we don't
17:33.26MooingLemurit's a country code
17:33.26DrDekeManxPower: Although that's the way I see it as well, many ITSPs do send the "1" as part of the caller ID number when the caller is from the US. It's very annoying.
17:33.26ManxPowerCunningPike, put a wait(1) before anything else.
17:33.27IkarusCunningPike: ah
17:33.27CunningPikeWe put a Wait(2) in - no joy
17:33.32IkarusCunningPike: quite different for the issue here
17:33.36ManxPowerDrDeke, caller from us and calling a non-usa number?
17:33.39CunningPikeIkarus: True :)
17:33.44DrDekeManxPower: Caller from US calling a US number
17:33.46rajiv|workManxPower: so my equip should send 10 digits... but what if i am calling non-usa? then dont they need to have 11 ?
17:33.49ManxPowerdr0ck, ah.
17:34.13DrDekeManxPower: yeah, it's a pain that they aren't all consistent. But I do think you're right; you really "should" send 10 digits
17:34.14wulfstan1Right, thanks I will give it a try and see if I have any luck
17:34.16ManxPowerrajiv|work, I think the carrier will add the country code of 1 at the beginning, not sure.
17:34.17wulfstan1Will be back if I don't
17:34.19generalhan[TK]D-Fender: i dont know if that will help me accomplish my goal ... i have 20 phones in a call queue and i cant turn the callwaiting off in the config or on the phone itself (i called the manufacturer) so what i want to do is dial to a context from the queue that will check to see if the person is on a call and only ring to the 15 of the 20 that are not on the phone.
17:34.20wulfstan1:-)
17:34.20CunningPikerajiv|work: Your CID is unlikely to survive the journey anyway
17:34.39ManxPowergeneralhan, what phone.
17:34.40[TK]D-Fendergeneralhan : Yes this command IS what you need.
17:34.42MooingLemurwe get UK callerid from time to time at work
17:34.43rajiv|workCunningPike: really? inbound caller id from germany to my analog lines in boston works
17:34.48generalhanAastra 9112i
17:34.58brif8anyone working on getting a Cisco 7920 to be SIP capable,  or what is the process ?
17:34.59CunningPikerajiv|work: Really - interesting
17:35.00[TK]D-Fendergeneralhan : And I have used it for EXACTLY that reason.
17:35.23IkarusCunningPike: CID in the form of a name never survives internationally with current protocols
17:35.27CunningPikeWe usually get the number for our nearest access point - but I'm in Canada, so it may be different here
17:35.31Ikaruswell, possible exception, USA and Canada
17:35.43generalhan[TK]D-Fender: in the configs it talks about trying one line and if its not avail going to the next line ... but i want it to test several of the lines, and then dial ALL that arent on a call
17:35.45ManxPowerbrif8, you don't "work on it" Cisco either has SIP firmware for the phone or they do not.
17:35.51viperdudeukbrif8: you need the SIP firmware from cisco
17:35.51MooingLemurthe name seems to be a lookup by your local phone company
17:36.21brif8any ideas if or when they may get SIP firmware for the 7920 ?
17:36.37ManxPowerbrif8, Cisco seldom announces such things.
17:36.45brif8ok
17:37.01ManxPowerDidn't you confirm that the phone supported SIP before you bought it?
17:37.02Ikarusrajiv|work, MooingLemur: number based Caller ID is highly reliably internationally in my experience
17:37.18viperdudeukcheck the cisco site... you will probably need a cisco support contract to get the sip image
17:37.21ManxPowerbrif8, MANY Cisco phones never get SIP firmware
17:37.31eric_hillCisco doesn't offer a SIP firmware for the 7920 at the present time.
17:37.44NuggetCisco has difficulty spelling SIP.
17:37.46eric_hillSCCP only, and you have to have crypto enabled.
17:38.05brif8ManxPower: my hope is that the 7920 isn't one
17:38.07viperdudeukcisco want you to by ccm
17:38.10viperdudeukbuy
17:38.31[TK]D-Fendergeneralhan : You don't ring multiple phones on a queue pas normall, jsut an individual agent through "agentcallbacklogin"
17:38.33brif8eric_hill: I realize presently it is only SCCP, I'm hopes is they change as they did with the 7960
17:38.58Qwell[]7960 has pretty much always had sip firmware
17:39.00generalhan[TK]D-Fender: i dont use agents at all ...
17:39.02Qwell[]7970 however...
17:39.26Qwell[]but, what's wrong with sccp?
17:39.41viperdudeukchan_sccp crashes a lo
17:39.42viperdudeuklot
17:39.49Qwell[]yeah, but what
17:39.54Qwell[]'s wrong with sccp itself?
17:40.07generalhan[TK]D-Fender: i just have each SIP line that i want to dial to in the queues.conf file and they ring that way. but if i just put one single member in there i can have that one be a different context in extensions.conf that will test all the lines and them ring them all
17:40.30viperdudeuknothing wrong with it just not stable on *
17:40.31brif8Qwell[]: it would appear mine just crashed.  I started a call and now have the phone off, yet sccp show lines  still shows it connected.  sccp reload doesn't work  nor has sccp reset DEVICEID  or sccp restart DEVICEID
17:40.58Qwell[]yes, chan_sccp is buggy...but that doesn't mean sccp is a bad protocol
17:41.09generalhan[TK]D-Fender: so ChanIsAvail will work if ${AVAILCHAN} can hold multiple lines
17:41.11[TK]D-Fendergeneralhan : Either way it will test a given line and you can chain them up any way you likel
17:41.18viperdudeukthats why i said chan_sccp crashes not sccp crashes
17:41.28brif8Qwell[]: I'm not calling it a good or bad protocol. just SIP seems alot more stable.
17:42.18*** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu)
17:42.43[TK]D-Fendergeneralhan : Read the instructions CLOSER.  You're not paying attention...
17:43.01[TK]D-Fendergeneralhan : There is an option to se if they are on ANY CALL AT ALL.
17:43.26generalhan[TK]D-Fender: lol ... ok ok im still looking into it
17:44.31generalhan[TK]D-Fender: i just get discouraged when i see things like this in the wiki "If you want to use it for limiting simultaneous calls to the peer, it will not work reliably for you.
17:45.06[TK]D-Fender<PROTECTED>
17:45.06[TK]D-Fender<PROTECTED>
17:45.20[TK]D-Fendergeneralhan : Like Nike says "Just Do It"
17:45.36harryvvbtw, how do i get the ip500 to switch over to the caller waiting caller if I am already talking to one party.
17:48.12harryvvI have missed a few calls trying to press the reciver button and ended up disconecting the caller.
17:48.55Hmmhesaysyay, the gf is bringing me to lunch
17:49.10Qwell[]Hmmhesays: would be better if she was bringing you two lunch
17:49.27Hmmhesaysit'd be better if she was bringing me over to the warehouse
17:49.41Hmmhesayscause then I could get some and play guitar
17:50.55*** join/#asterisk DeV-rAd (n=jesse@fl-69-69-130-197.sta.sprint-hsd.net)
17:51.09Hmmhesaysand that my friends would be fantastic
17:51.30DeV-rAdcan someone help, i need help setting up 2 fxo cards
17:51.48Hmmhesays$75/hour
17:51.49[TK]D-Fenderharryvv : depends how you set up your line keys.  if the cascade w/ 1 line/call then you jst need to hit the next line key, otherwise you use the cursor keys to pick the call on the keeys thats ringing
17:52.16[TK]D-FenderHmmhesays : The one who got you into that mess?
17:52.20harryvvtk, this is a single line pstn phone system.
17:52.24Hmmhesays[TK]D-Fender she is long gone
17:52.33Hmmhesaysi met a new one about a month ago
17:52.53Hmmhesays6ft 135lbs
17:52.57Hmmhesaysall kinds of yummy
17:53.02sevardHmmhesays: always cat > /dev/null before cleaning your keyboard
17:53.15Hmmhesayshahahaa
17:53.35*** join/#asterisk IceManRISK (n=kart@200.138.147.142)
17:53.51*** part/#asterisk pythos (i=pythos@unaffiliated/pythos)
17:53.52[TK]D-FenderHmmhesays : Always dodge left on ice-picks.... remember it....
17:53.52IceManRISKhi
17:54.01Hmmhesayslol
17:54.19*** join/#asterisk heison (n=heison@ns.somanetworks.com)
17:54.31brif8which a more stable version of chan_sccp  berlios or sourceforge  and why the two ?
17:54.35sevard[TK]D-Fender: always tip chinese hookers an extra dime if it was worth it
17:55.20*** join/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net)
17:56.25[TK]D-Fendersevard : Not my lifestyle...
17:56.36sevard[TK]D-Fender: i've seen your clsoets
17:56.38sevardclosets*
17:57.01[TK]D-FenderYeah... I have the body of a Chippendale..... its buried in my backyard :D
17:57.59*** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net)
17:58.23lunkany ideas why an internal sip phone can respond to a message by pressing 1, but an externally called person can not?
17:58.56jake1932how are you getting out?
17:59.10*** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net)
17:59.20lunkit's a broadvoice trunk
17:59.27justinu|laptopdtmf modes
17:59.30jake1932check compatible dtmf modes
17:59.46lunkwill do
18:00.39sevardDo Digium PRI cards do ESF, b8zf and NI2?
18:00.43sevardI would assume so.
18:01.15*** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net)
18:01.52Hmmhesayssevard: you could do the unthinkable
18:02.06sevardHmmhesays: superman powers?
18:02.11Hmmhesayslook on the digium website
18:02.14sevardOMG
18:02.18jake1932no!
18:02.27sevardthe devil's handbook, lad.
18:02.47justinu|laptopsevard: yes, they do
18:03.25sevardjustinu|laptop: just wading through all of this tdm crap and trying to soak it up, do you know if the TE205P does ground/wink/mediate
18:03.37Hmmhesaysanyone ever read "saucer" by stephen coonts?
18:04.56bzbwstrange, my * is sending DNS query to a none exist DNS name where part of it is the context name, and I did not define any domain for *, anyone has any clue?
18:05.07*** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
18:06.14justinu|laptopsevard: afaik, the digium cards support all common T1 signalling methods, including FXS_GS, E&M wink/immediate
18:06.26mog_worktrue true
18:06.31justinu|laptopsevard: correction, there may be issues with wink start
18:06.37*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-241-16.red.bezeqint.net)
18:06.39justinu|laptopmyself and others havent been able to make it work right
18:07.14sevardjustinu|laptop: alright, i just found the TE405P signaling page
18:07.28sevardit includes Wink (E&M) but for nubsake i should stray from that?
18:07.34jake1932bzbw: if it's failing, it should should which module is trying the lookup
18:07.41jake1932should say
18:07.50*** join/#asterisk inv_arp[work] (i=junya@adsl-10-153-159.mia.bellsouth.net)
18:08.00justinu|laptopsevard: what do you need to do with it?
18:08.06bzbwI have a BroadVoice context in sip.conf for dialing out using my BV account, and the * box itself has a dns name called asterisk.mydomain.com, and it is sending BroadVoice.mydomain.com DNS query, why?
18:08.21sevardjustinu|laptop: i'm buying pri service from my telco
18:08.36sevardjustinu|laptop: they asked if my card does mediated start, wink start, or ground start
18:08.49bzbwjake1932: it should be sip.conf, right?
18:08.53*** join/#asterisk Mike (n=mike@dsl-201-129-119-118.prod-infinitum.com.mx)
18:08.56*** part/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com)
18:08.56jake1932bzbw: looks like you missed ".com" or ".net"
18:08.56justinu|laptopsevard: ok, they're on crack... tell them that PRI is mutually exclusive from wink/immediate start.
18:09.00sevardit's looking like i should do ground start
18:09.02justinu|laptopsevard: stupid telco morons
18:09.08sevardjustinu|laptop: what do you mean?
18:09.09jake1932bzbw: it's using the default dns suffix
18:09.10justinu|laptopPRI is common channel signalling
18:09.16Mikehow can i make a channel that always gives me all circuits busy? like a sip channel incominglimit=0?
18:09.38justinu|laptopall those other signalling protocols aren't used with PRI... it's either or.
18:10.05bzbwjake1932: how do I get rid of it?  I don't want * to use the suffix for my * box and instead, use broadvoice.com for the suffix.
18:10.35bzbwjake1932: for dialing out using BV accounts.
18:10.39justinu|laptopsevard: it's unfortunate that your order entry people are so clueless, but the provisioners who configure their switches will understand.
18:10.50justinu|laptops/your/your telco's/
18:11.04jake1932bzbw: most likely it's somewhere in your sip.conf file that says "broadvoice" instead of "broadvoice.com"
18:11.13CunningPikejustinu|laptop, sevard: It's frightening how moronic some telcos can be
18:11.24jake1932either in the register line or somewhere in the BV context
18:11.40sevardjustinu|laptop: so PRI is a signaling method and wink/ground/immediate start are not encapsulated by PRI
18:11.44bzbwjake1932: yes, I define the context as BroadVoice instead of BroadVoice.com.
18:11.54justinu|laptopsevard: that's right.
18:12.07justinu|laptopwink/ground/immediate start are NOT applicable on a PRI
18:12.20bzbwJake1932: but inside the context, it defines host=sip.broadvoice.com and fromdomain=sip.broadvoice.com
18:12.32mutimmediate!
18:12.40mutlike a partyline!
18:12.42mutwoohoo
18:12.45jake1932what about the register line?
18:12.55sevardjustinu|laptop: that's where the confusion lies, where are they applicable?
18:12.59bzbwjake1932: and in extensions.conf, I use Dial(SIP/${EXTEN}@BroadVoice)
18:13.10LostFrogParty??? Where?
18:13.28justinu|laptopsevard: those signalling methods apply to a channel associated signalling t1
18:13.32justinu|laptopolder technology
18:13.34bzbwJake1932: I thought it will resolve BroadVoice to the right peer for calling out:)
18:13.46jake1932<PROTECTED>
18:13.51justinu|laptopsevard: those things are commonly used when talking to a channel bank connected to analog phones.
18:14.15bzbwjake1932: the strange thing is, it was working till a week ago:(
18:14.21jake1932<PROTECTED>
18:14.36bzbwjake1932: yes
18:15.33jake1932it clearly thinks broadvoice is a host name and not a peer entry
18:15.45*** join/#asterisk ToTo (n=ToTo@host187-131.pool872.interbusiness.it)
18:15.55bzbwwhy???
18:16.07jake1932can you pastebin both?
18:16.17jake1932(just the relevant portions)
18:16.21blitzragebzbw: that is the wrong formatting -- SIP/BroadVoice/${EXTEN} is what you want
18:16.44bzbwblitzrage: I tried that format, same thing.
18:16.46jake1932blitzrage: that format works for me
18:16.47blitzrageif you use an @, its going to assume a hostname
18:17.07jake1932maybe it's being phased out though
18:17.08blitzragebzbw: what does your sip.conf entry look like? (pastebin)
18:17.22bzbwblitzrage: let me get them pastebin
18:19.01sevardjustinu|laptop: alright, the problem wasn't with the telco but the person who gave me the notes from the telco lady was retarded
18:19.06sevardi called her directly and sorted it out
18:19.08sevardthank you
18:19.15mutanyone know any settings that might help fax over voip?
18:19.18*** join/#asterisk a1fa (n=a1fa@207.210.210.202)
18:19.21mutlike more fxs impedence
18:19.26mutor no echo cancel on the ata
18:19.27mutor something
18:19.37a1fawhat wa s the guys name that owns kneedraggers.com?
18:19.49a1fa--
18:20.01mutanyone
18:21.35a1fano
18:21.37bzbwhere is the related info, I hide the real configuration:)   http://pastebin.ca/52629
18:22.01justinu|laptopsam
18:22.15bzbwbtw: I'm using 1.2.7.1
18:24.26bzbwdoes network setting in the * box affect the DNS query from *?
18:25.33lunkjake1932: hey man, changing the dtfm mode to rfc2833 solved the problem, thanks a bunch
18:25.39jake1932np
18:26.22bzbwblitzrage: any clue?
18:27.24*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
18:28.17jake1932<PROTECTED>
18:28.19jake1932?
18:28.25blitzragebzbw: need to see what the CLI is doing
18:28.38blitzragebzbw: and get rid of dtmf=inband, thats an invalid option
18:29.40bzbwk.  there is no error, the sip invite is sent to the wrong dns name with BroadVoice.mydomain.com:(
18:31.01SpaceBassI have a cisco 7940 that is behaving strangely
18:31.05SpaceBassI'm using a modified POE cable from the wiki and it powers on fine, but says "ethernet disconnected" ... i get the same result if i use a standard able and a cisco power cube...however I have ONE port on an older switch on which it does work
18:31.10SpaceBassanyone seen anything like that
18:31.21jake1932unless BroadVoice.mydomain.com is a valid host, there should be an error
18:31.24Qwell[]SpaceBass: wrong port?
18:31.39Qwell[]ie; the "working" switch port is auto-sensing?
18:32.01SpaceBassQwell[],  the older swich is retired due to a VERY loud sound it makes constently.... the new one is an unmanaged autosensing switch
18:32.05*** join/#asterisk saftsack (n=saftsack@p54A7FC22.dip.t-dialin.net)
18:32.30justinu|laptopmaybe the speed negotiation isn't working right
18:32.38SpaceBassQwell[],  I checked the old managed switch carefully and could not find a thing different about that port...it was set to full duplex, autosensing, 100mbs...
18:32.58SpaceBassjustinu|laptop, could be....thought about checking the settings in the phone to see if its dialed down to 10mbs or something
18:33.02Qwell[]which port are you using on the phone?
18:33.24SpaceBass10/100sw
18:33.26SpaceBass:)
18:33.42Qwell[]use the middle one...whatever it says :p
18:33.53Qwell[]rs323, lan, switch...something like that
18:34.25SpaceBasssettings on the phone are media type: auto,
18:34.48Qwell[]iirc, the switch port is the far left one...
18:34.52SpaceBassthe odd thing is that its not the crazy modified POE cable I made
18:35.01SpaceBassQwell[],  far left, yeah
18:35.06Qwell[]yeah, use the middle one
18:35.11SpaceBass(well, between aux on the left and the PC port on the right)
18:35.16SpaceBassusing the middle one
18:35.30Qwell[]it can't be far left AND middle :P
18:35.36Strom_Cdid somebody say Cisco?!
18:35.39bzbwjake1932: http://pastebin.ca/52636
18:36.01SpaceBassusing the middle one
18:36.08Qwell[]okay then
18:36.28Qwell[]straight through cable doesn't work?
18:36.40SpaceBassthe Network port 2 device type is set to hub/switch and I can change it to PC
18:36.49SpaceBassnot sure what the diference is...guess slip cable vs straight through
18:37.04SpaceBassQwell[],  yeah, if I use the power supply and a regular straight cable, i still get the error
18:37.12bzbwjake1932: the dns did respond, but the ip is not bv proxy:(
18:37.13Qwell[]then it's busted :p
18:37.20SpaceBassd'oj!
18:37.22SpaceBassd'oh
18:37.27SpaceBassno smartnet
18:37.53Qwell[]make a crazy two-connector crossover/poe cable
18:37.56SpaceBassthat means I have to take the phone out of my master bathroom in the mean time
18:37.56Qwell[]:P
18:38.04SpaceBass:)
18:38.11Qwell[]Or just a regular one...I imagine the pc port can take poe
18:38.51SpaceBassregular what? phone? Don't have any analogue phones or a spare ATA....i took great happiness in smashing my old POS cordless when I went all VoIP
18:38.51SpaceBass:)
18:39.07Qwell[]no, regular crossover/poe cable
18:39.22Strom_Cjeez, and here I still use a rotary phone made in 1948
18:39.53jake1932bzbw: i tried to replicate what you're doing without an error.  every time i put in an invalid host, it comes up with an error.
18:40.00Qwell[]Strom_C: Do you have to crank it?
18:40.03Qwell[]if not...pfft
18:40.12Strom_CQwell[]: I said rotary, not magneto
18:40.17Strom_Csheesh
18:40.32SpaceBassHEY HEY! I changed it to 10mb full duplex and it worked!
18:40.35SpaceBasseurika!
18:40.45SpaceBassnope...false alarm
18:41.00SpaceBassjust pretended to work until I tried to make a call
18:42.11bzbwjake1932: no, the BroadVoice.mydomain.com does resolve to multiple A record, which is pointing to multiple mydomain.com's IP, but they are not valid sip proxy, not BV proxy:(
18:42.13LostFrogDamn faker phones. :)
18:42.29SpaceBasshosting your own BV proxy?
18:42.51jake1932bzbw: maybe backup your sip.conf and create one with just the bv register and peer entry.  the err seems to have to do with not using BroadVoice as a peer entry
18:42.53bzbwjake1932: the issue is, why does it put "mydomain.com" suffix after the context "BroadVoice"?
18:43.29SpaceBasshave a DNS search suffix on that box?
18:43.29Mikecan i create a zap channel with out a zap card? just so i can create a zap channel that always returns busy?
18:43.29jake1932because it doesn't think BroadVoice is a peer entry - it thinks it's a host
18:43.41SpaceBassMike check out zapdummy on the wiki
18:43.43bzbwSpaceBass: I'm just trying to make a peer entry works, which is using BV accont
18:44.00SpaceBassbzbw gotchat...have you tried using the ip or putting an entry in the /etc/host file?
18:44.44bzbwSpaceBass: I did, it does not work either, looks like BV proxy want From, To header with "sip.broadvoice.com" instead of ip address:(
18:45.01bzbwSpaceBass: i mean in the SIP INVITE.
18:45.42SpaceBassbzbw, gotcha.... I have sip.broadvoice.com in my sip.conf and then i have a line in /etc/host for the Washington DC server
18:45.47MikeSpaceBass, no zapdummy just ztdummy
18:45.58SpaceBassMike oops...good catch
18:46.22MikeSpaceBass, thats just a timer
18:46.37MikeSpaceBass, i need a zap channel that always returns all circuits busy to get a 34 message
18:46.43Mikeon my hangucause
18:46.45SpaceBassMike thought that might do it....if you want something that will always return busy, why does it have to be ZAP?
18:46.50SpaceBassah
18:46.52bzbwSpaceBass: I have this: "147.135.20.128  sip.broadvoice.com" in /etc/hosts
18:47.02MikeSpaceBass, hangupcause
18:47.48bzbwjake1932: u r right, why does it think it is NOT a peer:(
18:48.37jake1932maybe your sip.conf may has an issue - check my earlier suggestion
18:48.47jake1932ooh - horible english
18:48.49bzbwjake1932: and funny things is, how does the * knows the suffix of my linux box and attach it to the context name?
18:49.07jake1932it doesn't
18:49.17jake1932it's appending the default suffix
18:49.34bzbwjake1932: where does default suffix defined in *?
18:49.41jake1932it's not
18:49.52jake1932it's in your dns file - resolv.conf (i think)
18:49.54bzbwthen where does it gotten this?
18:50.25bzbwI don't have domain defined in the /etc/resolv.conf.
18:50.34SpaceBassDHCP?
18:51.15jake1932resolver.conf
18:51.25bzbwno static ip
18:51.48SpaceBasshummmm
18:52.23bzbwjake1932: there is no resolver.conf in my linux:(
18:53.12jake1932i had it right the first time
18:53.21jake1932-- /etc/resolv.conf.
18:53.21nextimebzbw : /etc/resolv.conf, not resolver
18:53.55bzbwthat's why I meant, it's just a couple name server with ip address.
18:54.43nextimebzbw : search domain.tld at the first line of your resolv.conf plus a configured zone in your dns server is what you need.
18:55.18jake1932bzbw - did you try backing up your sip.conf and creating a new one with just the broadvoice (and one phone)?
18:55.22nextime( if i understand right what you need )
18:55.56*** join/#asterisk mercestes (n=merceste@69.15.174.114)
18:56.02bzbwnextime: I don't control the DNS server:(, but what do u mean "domain.tld"?
18:56.41bzbwjake1932: this is a production server, I will have to have a sleepless night again to do so.
18:56.45SpaceBassARRRUUGGGG this crazy ass cisco phone just started working...and i didnt change a thing
18:56.54SpaceBassdo I dare move it back to the kitchen where it lives?
18:57.33jake1932bzbw: what did you change before it stopped working?
18:58.42bzbwas far as I know, not a thing, but if I do remember, I might have already solved this puzzle:).
18:59.16jake1932did it create a backup file for you?
18:59.25nextimebzbw : google.com, google is a domain, .com is a tld :)
18:59.26bzbwI did upgraded it to 1.2.7.1, but that's after it was not working, so I don't think 1.2.7.1 is the reason.
18:59.59bzbwnextime: thx.
19:00.00jake1932i believe your problem to be in sip.conf
19:00.16nextimeif you have your own domain to resolve in your network, you will use search domain.tld in resolv.conf, it's simple
19:00.19bzbwbut what u see is what i have:)
19:00.25jake1932well
19:00.29jake1932that's part of it
19:01.09jake1932for instance - i'm using 1.2.7.1 and doing sip lookups just fine
19:01.15*** join/#asterisk tdonahue-laptop (n=tdonahue@www.vonworldwide.com)
19:01.19nextimestartrek enterprise is starting, bye bye
19:02.17jake1932is BV your only peer?
19:02.30jake1932(besides your phones)?
19:02.59bzbwjake1932: I have another peer with is using SER, same thing:(
19:03.31*** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
19:03.40*** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
19:03.47*** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
19:03.53jake1932the plot thickens
19:03.57TripleFFFFFFFFFF_1NXXNXXXXXX
19:04.05TripleFFFFFFFFFFhow i make this with or without the 1
19:04.26TripleFFFFFFFFFFto match 123-123-1234 and 1-123-123-1234
19:04.38*** join/#asterisk TheCops (i=nobody@got.securebinary.com)
19:04.43jake1932_NXXNXXXXXX
19:04.49[TK]D-FenderTripleFFFFFFFFFF : Add another for _NXXNXXXXXX
19:04.59[TK]D-Fenderjake1932 : Ok, you win this one :)
19:05.22TheCopsHello Fender
19:05.22TheCops:)
19:05.33jake19321 for me, 500 for [TK]D-Fender
19:06.26TripleFFFFFFFFFF??
19:06.29TripleFFFFFFFFFFAS IN
19:07.12[TK]D-Fender:D
19:07.17[TK]D-FenderTripleFFFFFFFFFF : ADD ANOTHER PATTERN
19:07.21TripleFFFFFFFFFF<PROTECTED>
19:07.23[TK]D-FenderTheCops : y0
19:07.29brodiemIs there a difference between the open source G729 and the licensed G729 w/ *?
19:07.49bzbwjake1932: is there a cmd that allows me see how * execute dialing for a context?
19:07.49TheCops[TK]D-Fender. how are you
19:07.55[TK]D-FenderTripleFFFFFFFFFF : You DON'T make one that will work with both patterns, you make 1 EACH to account for EACH pattern
19:08.01[TK]D-FenderTheCops : Still breathing.
19:08.08TripleFFFFFFFFFFk
19:08.20TripleFFFFFFFFFFi know i tought we could make as one
19:08.21TripleFFFFFFFFFFsucks
19:08.28jake1932bzbw: set verbose 30
19:08.29*** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
19:08.32TripleFFFFFFFFFFshoul have {!}
19:08.37TripleFFFFFFFFFF[1]
19:08.39TripleFFFFFFFFFFI MEAN
19:08.43TripleFFFFFFFFFFi mean
19:08.43[TK]D-FenderTripleFFFFFFFFFF : nope
19:08.48bzbwjake1932: I set it to more than 37:(
19:10.30*** join/#asterisk willt (i=wt@wifi-napanet-static-206-81-99-68.napanet.net)
19:11.10willtDoes anyone know if I can disable the speakerphone on a cisco 7960 without using callmanager?
19:11.33*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
19:11.39jake1932bzbw: did you try pinging BroadVoice
19:11.42SpaceBasswillt,  I'm not aware of a setting in the sip config to do that
19:11.50jake1932bzbw: does it append the suffix?
19:12.21TripleFFFFFFFFFFhttp://pastebin.ca/52650
19:12.24willtI know callmanager can do it but I can't find anything in the sip config..
19:12.25*** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane)
19:12.26TripleFFFFFFFFFFso this would make sense ?
19:12.33TripleFFFFFFFFFFto make group res have only 4 channels ?
19:12.35bzbwjake1932: from console?
19:12.38*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
19:13.01TripleFFFFFFFFFFso in extensions i would macro res.. then dial whereever
19:13.24brodiemis anyone using the intel based g729 implemntation?
19:13.25jake1932from a cmd line
19:13.49[TK]D-FenderTripleFFFFFFFFFF : res?
19:13.56TripleFFFFFFFFFFresidential
19:13.57TripleFFFFFFFFFFlol
19:14.00TripleFFFFFFFFFFlike a service
19:14.11bzbwjake1932: hey, u are good, it did responded, just as I described, it tried BroadVoice.mydomain.com and the server responded
19:14.11*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
19:14.19[TK]D-FenderTripleFFFFFFFFFF : I would basically manipulate the # as required to format it so the dial doesn't have to do the thinking.
19:14.21TripleFFFFFFFFFFso my users before lettimg them receive the call i would do that
19:14.31bzbwjake1932: how do I get rid of this?
19:14.47bzbwI did not specify anywhere for this
19:15.24TripleFFFFFFFFFFso in extensions.. => context incoming-user ... prio 1 would be set account =blah , prio 2 , macro res, prio 3 dial wherever
19:16.47jake1932bzbw: try pinging something else
19:17.02jake1932sounds like 2 issues
19:17.30jake1932how come BroadVoice.mydomain.com is responding?
19:17.46TripleFFFFFFFFFFwould it work or not
19:17.49TripleFFFFFFFFFFhow would one do it ?
19:18.02bzbwjake1932: I found something, let me delete that file first, it is in networking/profiles/default/resolv.conf:
19:18.57generalhan[TK]D-Fender: ok so i got it to work with one channel for the ChanIsAvail cmd. now my question is can i have the queue, call on that entry that i made in extensions.conf ?
19:19.06*** join/#asterisk Weezey (n=ohno@lo20.loit.ca)
19:19.31bzbwjake1932: does not work. I try to delete that file:(
19:19.34[TK]D-Fendergeneralhan : I would have though you already had it doing that... how is it calling them NOW?
19:19.38*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
19:19.53*** part/#asterisk terr_ (n=terr_@dsl-cap-66-18-218-43-cgy.nucleus.com)
19:20.31jake1932bzbw: not sure if you have to restart networking to reread resolv.conf
19:20.56TripleFFFFFFFFFFcro_exec: Context 'macro-res' for macro 'res' lacks 's' extension, priority 1
19:21.01TripleFFFFFFFFFFoh well
19:21.11generalhan[TK]D-Fender: well i just set up a context in extensions.conf and called it to make sure that it worked... and i got it to work so that if i call an extension it goes to a macro with that stuff in it to decide if they are avail. or not. now i want to set that context to the queue so that all of the members that are dialed have to do through that macro to see if they are available
19:21.50Weezeyanyone use realtime?  I added an IAX2 peer and it answers with congestion when I dial IAX2/user@context/${EXTEN}  I had this problem before with another one and I can't remember how I ended up fixing it.
19:22.08WeezeyIf I dial user:secret@host it goes just fine.
19:22.19Weezeybut I'd like to keep the secret out of the CDR
19:24.02bzbwjake1932: I did restart the networking many times, not working.
19:24.22jake1932my asterisk experience is better than my linux
19:24.34jake1932you gots some DNS issues
19:25.35bzbwjake1932: yes, why does this damn thing automatically attach a domain, where I can not find it:((
19:26.01Qwell[]Did you do what blitzrage suggested, and change the Dial string?
19:26.16Qwell[]Dial(SIP/BroadVoice/${EXTEN})
19:26.30*** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com)
19:26.43*** part/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
19:27.44Hmmhesayswell that was a pleasant lunch
19:28.00Drukenwas it?
19:28.08Hmmhesaysyes
19:34.02bzbwjake1932: Thx for the help, I somehow work around the problem and it is now working:)
19:34.43hwtare there any good asterisk stress testing tools out there?
19:34.47bzbwjake1932: your tip on pinging BroadVoice from cmd helped me:)
19:34.58jake1932very good
19:34.58TripleFFFFFFFFFFoh god
19:35.02hwtor SIP initiators that i can use to stress test.
19:35.06bzbwbye now
19:35.08*** part/#asterisk bzbw (i=bwz@ip67-153-142-109.z142-153-67.customer.algx.net)
19:35.13TripleFFFFFFFFFFhttp://pastebin.ca/52662
19:35.18TripleFFFFFFFFFFdoesnt work
19:35.26trelane_hwt, astertest
19:35.27TripleFFFFFFFFFFjust passes by it even when i called from 3 phones at same time
19:35.32TripleFFFFFFFFFFi see the noop etc
19:35.38TripleFFFFFFFFFFand it rigns my local cisco
19:35.56hwttrelane_: it doesn't seem to be working properly on 1.2.x.
19:36.19hwtnoop() is just for debugging, right?
19:36.38Qwell[]hwt: sipp
19:37.12TripleFFFFFFFFFFy
19:37.28trelane_hwt, works for me?
19:37.41trelane_hwt, I use astertest here at work
19:37.48TripleFFFFFFFFFFgod
19:37.56trelane_TripleFFFFFFFFFF, yes?
19:38.02*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
19:38.02TripleFFFFFFFFFFanyway to limit # of channel incoming + outgoing for a user part of a group ?
19:38.05trelane_though that might be overly presumptious of me, there may be others
19:38.07eric_hilltrelane_: lol!
19:38.15TripleFFFFFFFFFFlike .. DUMMIES could have all 2 channels each
19:38.20TripleFFFFFFFFFFand GURUS 100
19:38.33TripleFFFFFFFFFFbut you could have if we want 500 GURUS.. each with 100 channels
19:38.41hwtQwell[]: intriguing, but i would prefer something that also establishes RTP-channels.
19:38.46Qwell[]yeah, sipp
19:38.48TripleFFFFFFFFFFi DONT mean 100 total .. but 100 for that user
19:38.54TripleFFFFFFFFFF??
19:39.13trelane_Qwell, sipp took effort to understand, I had astertest up in minutes and it draws nifty graphs
19:39.13TripleFFFFFFFFFFso lets say we had 100 dummies thats a max of 200 channels
19:39.14hwtQwell[]: ah, "SIPp can also send media (RTP) traffic through RTP echo and RTP / pcap replay."
19:39.18hwtQwell[]: missed a line.
19:39.40hwttrelane_: well, you have to recompile asterisk. that kinda sucks.
19:40.02trelane_hwt, what makes you recompile asterisk for astertest?
19:40.44*** join/#asterisk SuperLag (n=aaron@gentoo/developer/SuperLag)
19:40.52hwttrelane_: i just looked at this one: http://www.asteriskguru.com/tutorials/astertest.html
19:41.08trelane_hwt, you know how to use managers.conf?
19:41.50TripleFFFFFFFFFFdarn
19:41.52trelane_you dont' *NEED* those modules
19:41.56TripleFFFFFFFFFFok i need group = account number lol
19:42.33hwttrelane_: yeah?
19:42.50hwttrelane_: really? what are they for, then?
19:42.57hwttrelane_: i should probably rtfm, though. :)
19:44.55trelane_hwt: they can return cpu/memory load metrics from the two asterisk systems
19:45.10*** join/#asterisk mtaht3 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
19:46.48TripleFFFFFFFFFFworks
19:47.00Zodiacalanyone know if theres a way to make the button press tones quieter?
19:47.12hwttrelane_: ah, but i feel that's a pretty nice feature.
19:47.46TripleFFFFFFFFFFok thanks for chatting lol actually put my mind aside and helped me out
19:48.06HmmhesaysZodiacal: ear muffs
19:48.13Zodiacal:P
19:48.31Zodiacalmight not be a bad idea, the a/c here is allways on full blast.
19:49.15Zodiacali guess i would have to increase the volume of the audio and lower the handsets volume...
19:49.26Zodiacalbut if i increase the gain's will that increase the tones too?
19:49.57Hmmhesayson what hardware?
19:50.17Zodiacalcisco 7960 phones with digim tdm400p fxo's
19:50.25Hmmhesayswhich tones are loud?
19:50.38Zodiacalwhen i press a button on the phone, i.e. 1,2,3, etc
19:51.03Zodiacalit hurts when calling automated systems that require button presses
19:51.11Hmmhesayson the 7960?
19:51.16Zodiacalyep
19:51.22Hmmhesaysthat is really odd
19:51.24Hmmhesaysmine doesn't do that
19:51.29Zodiacalsccp
19:51.32Hmmhesaysoh
19:51.56Zodiacalya sip was softer
19:52.06Zodiacali remember now
19:52.14Zodiacalsmall draw back
19:52.54*** join/#asterisk IceManRISK (n=kart@200.138.147.142)
19:53.01Hmmhesayshe's on teh dance floor yelling freebird
19:56.55TripleFFFFFFFFFFat least not yelling freeewilly
19:57.08noname32lol
19:57.10Hmmhesaysthat comes later in the night
19:57.38TripleFFFFFFFFFFmy 7960 firmware is fucked i think
19:57.49TripleFFFFFFFFFFeach time i go adimin on it ti fizzles ,m wiggles and reboots on me
19:58.02TripleFFFFFFFFFFand htat with only 1 finger
19:58.22TripleFFFFFFFFFFmight be that the phone heard too much crazy shit and went nuts
19:58.31TripleFFFFFFFFFFbah cisco not wht they used to be
19:58.35Hmmhesayshow do I use place holders in a bash script
19:58.44TripleFFFFFFFFFFplace holders ?
19:58.47TripleFFFFFFFFFFas in ?\
19:58.47C4T3lPAGING MERCESTES
19:58.49TripleFFFFFFFFFFARGS ?
19:58.52Hmmhesaysyes
19:58.58TripleFFFFFFFFFF${1}
19:59.25Hmmhesays./my_script boobies weee     ${1} is boobies and ${2} is weeee?
19:59.42Hmmhesaysminus the typo
19:59.55TripleFFFFFFFFFF./test.sh 1
19:59.55TripleFFFFFFFFFF1
20:00.07TripleFFFFFFFFFFecho  "${1}";
20:01.00TripleFFFFFFFFFFso echo "${1} is my first arg and ${2} is my second" my third is in a manual , and my whole is where ?
20:01.10Qwell[]$0
20:01.13TripleFFFFFFFFFFtry
20:01.31Hmmhesaysgot it
20:01.33Hmmhesaysdanke
20:02.04SpaceBass$0 would be the command (script name) itself, right?
20:02.08TripleFFFFFFFFFFecho "${1}ead{$2}he${3}ucking${4}anual"
20:02.16TripleFFFFFFFFFFand parse it Rtfm
20:02.16TripleFFFFFFFFFFlol
20:02.17Qwell[]maybe
20:02.20TripleFFFFFFFFFFno prob
20:02.23TripleFFFFFFFFFFjust having fun with you
20:02.27*** join/#asterisk darylp (n=daryl_ju@63-208-162-59.digitalrealm.net)
20:02.33TripleFFFFFFFFFF${0 is fielname
20:02.57SpaceBassi gotta get around to writing a few bash scripts later today....change some firewall settings
20:02.59*** join/#asterisk Samoied (n=Samoied@200-193-14-53.fnsce7003.dsl.brasiltelecom.net.br)
20:03.15SpaceBassI have one I wrote to control iTunes through asterisk....never got around to installing it on my current * box though
20:03.24*** join/#asterisk nagl (n=nagl@86.59.54.237)
20:03.39SpaceBassand it had no real point, except that I could call my DISA while I was traveling and crank Guns n Roses at 2 am to scare the shit out of my wife
20:04.18darylpI have a slightly off topic phone related question regarding fwd
20:05.02TripleFFFFFFFFFFlol
20:05.08TripleFFFFFFFFFFsapcebass too neat
20:05.23TripleFFFFFFFFFFhad a guy fuck me over 10k
20:05.28SpaceBassim gonna blog the script soon if I get time
20:05.49TripleFFFFFFFFFFi called his GF at 4 am till 5 am randomly every night and him the same but with each others numbers on callerid
20:05.49TripleFFFFFFFFFFlol
20:06.23TripleFFFFFFFFFFwhat ? who you think you are ? yeah you did call me .. i got your digitts on da phone lol
20:06.28TripleFFFFFFFFFFand vice verca
20:06.29SpaceBasslol!
20:06.37*** join/#asterisk PakiPenguin (i=uppal@linuxpakistan/admin/pakipenguin)
20:06.38SpaceBasscid spoofing is the greatest thing since sliced bread
20:06.53pigpenDoes Asterisk support Faxes?  <<<< Ha Ha!  just kidding...
20:06.54TripleFFFFFFFFFFso i guess im now only -9960.89$
20:07.13TripleFFFFFFFFFFi guess 1 monh of fon costes me like ...39$ for around 5000 calls
20:07.14TripleFFFFFFFFFFnot bad
20:07.30TripleFFFFFFFFFFnow what do do fo r9960$
20:07.40Qwell[]TripleFFFFFFFFFF: What's his car worth? :P
20:07.44[TK]D-FenderSpaceBass : Sorry, I've already claimed that one :)
20:07.57TripleFFFFFFFFFFqwell not sure
20:08.05TripleFFFFFFFFFFunpaid
20:08.07TripleFFFFFFFFFFwhy
20:08.07TripleFFFFFFFFFFlol
20:09.08SpaceBassarrrruuuuggg nerdvittles always locks up firefox on os x
20:09.21Qwell[]SpaceBass: Don't do that then
20:09.39darylpOval telecom wants to know the sip or iax destination of my uk phone number, how can I send that into a fwd number?
20:09.41SpaceBasswell I wanted to read about nvfaxdetect in @home 2.8 :(
20:09.54*** join/#asterisk MacDome (n=eseidel@A17-255-100-181.apple.com)
20:10.16TripleFFFFFFFFFFisnt fax in addons yet ?
20:10.20TripleFFFFFFFFFFwth they waiting if not
20:10.43darylpbasically I'm trying to understand how one specifies an iax desination
20:10.48darylpdestination
20:11.23TripleFFFFFFFFFFuser:pass@ip
20:12.58drfoomod2has anyone mucked around w/ sipX?
20:13.27*** part/#asterisk lzhang (n=lewiszha@67.95.13.46)
20:13.31*** join/#asterisk lzhang (n=lewiszha@67.95.13.46)
20:13.45*** part/#asterisk ringe (n=runar@ti531210a080-6380.bb.online.no)
20:13.54a1faanybody know where to download agi scripts, weather, wake-up? etc/
20:14.12lzhanghi guys, so I'm having this problem with Polycom phones and Asterisk, I'm using sip to dial out and the calls turn into one way sound after the third ring
20:14.14TripleFFFFFFFFFFmy brain has alot of them.. just need to compile them
20:14.27TripleFFFFFFFFFFprob i got no ports left
20:14.42TripleFFFFFFFFFFlzhang NAT
20:14.46TripleFFFFFFFFFFrtp
20:14.52TripleFFFFFFFFFFconnect dircet and test
20:14.57TripleFFFFFFFFFFyou are double natted
20:15.09TripleFFFFFFFFFFSERVER -> NAT -> NET -> NAT -> PHONE
20:15.22TripleFFFFFFFFFFmake sure nat=yes in sip.conf peer
20:15.24lzhangthe asterisk box is the gateway and has a public ip, and the phones are on an internal network getting dhcp from the asterisk box
20:15.37TripleFFFFFFFFFFmake sure nat=yes in sip.conf peer
20:15.55TripleFFFFFFFFFFand insecure=port,invite
20:16.07darylpI know, for example, that ipkall doesn't have my password, only my fwd number. How do they route incoming calls to fwd?
20:16.17TripleFFFFFFFFFFlook in rtp.conf an dforward that rang e from your router to your phone as udp
20:16.20TripleFFFFFFFFFFalso
20:16.33TripleFFFFFFFFFFip ?
20:16.44lzhangTripleFFFFFFFFFF: are you sure this is a NAT situation? ip is 66.76.53.31
20:17.17TripleFFFFFFFFFFcheckign oyu
20:17.51TripleFFFFFFFFFFhm,mm
20:17.56TripleFFFFFFFFFF5060 closed from outside
20:18.27lzhangtry again
20:19.02TripleFFFFFFFFFFnope
20:19.03TripleFFFFFFFFFFsorry
20:19.07lzhanghmm
20:19.18TripleFFFFFFFFFFok what peer name ?
20:19.28TripleFFFFFFFFFFCLI> debug peer pernamehere
20:19.33TripleFFFFFFFFFFand see what it does..
20:19.46TripleFFFFFFFFFFyoull see hwen it stops the rtp stream its because its gonna use internal ips
20:19.50TripleFFFFFFFFFFlike 192.168.1.x
20:20.06TripleFFFFFFFFFFon you r asterisk.. now obviousely it cant forwards that to your phone location
20:20.09TripleFFFFFFFFFFthats NAT
20:22.51lzhangk hmmm
20:23.08*** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154)
20:23.10gbodemantvhey all
20:23.30gbodemantvanypne ever get May  1 13:16:32 WARNING[14610]: format_wav.c:247 update_header: Unable to find our position
20:23.35gbodemantvrepeating over and over
20:23.55*** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-63-236.dsl.irvnca.pacbell.net)
20:24.16Qwell[]gbodemantv: What version of *?
20:24.18TripleFFFFFFFFFFno
20:24.21lzhangTripleFFFFFFFFFF: how do you check the port from the outside
20:24.22gbodemantv1.2.4
20:24.25Qwell[]upgrade
20:24.35Qwell[]lzhang: nmap
20:24.36TripleFFFFFFFFFFhehehe
20:24.38TripleFFFFFFFFFFnmap
20:26.39Drukenwhat do i want for dinner?
20:26.47*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
20:27.30*** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be)
20:27.38*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
20:28.25*** part/#asterisk a1fa (n=a1fa@207.210.210.202)
20:30.26WeezeyDruken: Beer?
20:30.37Drukennah....
20:30.43WeezeyWine.
20:30.58Drukeni was thinking kfc or something :)
20:31.04Drukennot sure tho...
20:31.06Weezeyeww
20:31.25Weezeytoo much fast food has made me hate it all.
20:31.44Drukeni should have hit the point, yet i havent...
20:32.04Drukenlast none fast food meal i had was like a week ago?
20:32.14Drukener.. non-fast food
20:33.19*** join/#asterisk BrianR___ (i=brianr@setient-sucks.978.org)
20:33.56WeezeyYou live near Oakville?
20:34.03WeezeyPanago makes a mean pie.
20:34.11Drukenbarrie
20:34.21Weezeyhmm, not close enough to Oakville.
20:34.30*** part/#asterisk BrianR___ (i=brianr@setient-sucks.978.org)
20:34.31Drukennope....
20:34.50WeezeyWho's providing dial tone in Barrie these days/
20:34.51Weezey?
20:34.58Drukenbell ?
20:35.08Weezeyvia IAX or SIP.
20:35.30Drukensprint/rogers has ilec, but bell is still the clec
20:35.51Drukenoh... who's providing did's.... not many
20:35.59*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
20:36.45*** join/#asterisk dlynes (n=dlynes@216.251.149.66)
20:38.27DrukenWeezey: actually.... i had that backwards, bell is the ILEC, rogers has CLEC
20:39.01Drukenbell is the incumbent carrier, rogers is the competitive carrier
20:39.56*** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net)
20:40.54pigpenIn regards to faxing, Fax - Digium FXS--(Asterisk)--Digium FXO/PRI---PSTN works pretty good.....
20:41.01pigpenbut it gets screwy when sip or iax is in the middle somewhere....
20:41.31Hmmhesaysok my GET VARIABLE agi isn't working
20:41.41pigpenor can faxing get screwed even in the first one?
20:42.08poisonerhmm... can anyone tell me a good GUI for asterisk which is free?
20:42.21pigpenfor config or stats?
20:42.51poisonerboth, if possible
20:42.52SpaceBassanyone know if Broadvoice or Telasip will do a real hard bill rather than charging a credit card?
20:43.05pigpenpoisoner, good luck.  I wrote my own.
20:43.40pigpenI'm sure lots of people would not like it...but getting me happy was #1.....
20:43.55pigpenpoisoner, new to asterisk?
20:45.08SpaceBasspoisoner, check out www.archatechs.com and then www.nerdvittles.com
20:45.41SpaceBassstart there and then you;ll get a sense of the big idea and the gui stuff is mentioned
20:47.08pigpenit was told to me "start command line" then experiment with gui's after you have an idea what is happening under the hood.
20:47.34pigpenif you are anal regarding "doing it your way" then you will end up back at the beginning.
20:48.37poisonerpigpen: At this moment I'm writing config files by hand. This is good enough for me, I think.
20:48.50poisonerBut some GUI will come along with nice scripts
20:49.13pigpensure...I understand...
20:49.27poisonerAnd perhaps I will manage a coworker to exchange Ciscos CM against *
20:49.59poisonerBut He seems to be one which has Problems with a cli or writing Cisco Configs with vi...
20:50.14poisonerSo I want too test some things
20:54.12noname32vi :<
20:54.18noname32emacs is the best
20:54.19SpaceBassthere is a GUI with some nice scripts :D
20:54.23SpaceBassvim
20:54.37noname32hehe waits for riot
20:54.38robin_szquick, lets start an editor war!
20:54.40SpaceBassln -s vi vim
20:54.47noname32i like freepbx
20:54.49drfoomod2nano
20:54.52drfoomod2be-atch!
20:54.55noname32nano is just pico ;p
20:55.07SpaceBassand I like to save them in word's native format...you know, since its so interoperable
20:55.15TripleFFFFFFFFFFman i type shitdown so often i symlinked to shitdown
20:55.17noname32i am die hard pico but for coding emacs is so much nicer
20:55.23mutfaxing over voip, i used to have people connected via fax machine -> sip -> asterisk -> sip -> cisco as5350 -> pri
20:55.24mutnow i have
20:55.41mutfax machine -> sip -> asterisk -> pri via sangoma a104d
20:55.46mutpeople can'
20:55.48mutt fax anymore
20:55.51mutideas why?
20:55.55drfoomod2nano -w
20:56.02drfoomod2gotta go wide
20:56.22noname32poisoner, check out #freepbx see if there gui will do what you want
20:56.26robin_szemacs is without doubt the best mp3 player I have tried
20:56.31noname32lol
20:56.32robin_szapparently, it can edit as well, but thats just a rumour
20:56.43mutoh
20:56.46sevarddoes anyone know how to use ChanSpy
20:56.49mutfax machine -> spa-2002 ata -> sip
20:57.08noname32some one needs to make a java aplet that will alow editing dynamic pages stored in db so i can code in emacs on the web :)
20:57.47brookshiremut: i can tell you one thing.. the pci card has nothing to do with it
20:58.06*** join/#asterisk mikefoo (n=mikefoo@64.124.169.2)
20:58.07mutthats all that changed
20:58.15noname32fax + voip = troublesome i though
20:58.16mutso how could it have nothing to do with it
20:58.16noname32lol
20:58.24mutthese people have been working for a year
20:58.25mutor more
20:58.26*** join/#asterisk abatista (n=Ariel@70.46.87.158)
20:58.47brookshiremut: fax over sip causes problems
20:58.48mutok
20:58.51brookshireend of story
20:58.52muti realize this
20:58.58brookshirethat's where you're problem is
20:58.58mutbut i'm telling you
20:59.00mutit's worked for a year
20:59.05mikefooAnyone NOT in the USA, and is able to perform a test call to me in the US?
20:59.07MikeJ[Laptop]still work to the 5300?
20:59.08muti put the cisco back in
20:59.11mutand it works fine again
20:59.12*** join/#asterisk bkw_ (n=brian@adsl-70-143-63-171.dsl.tul2ok.sbcglobal.net)
20:59.14mikefooI need it to originate from outside of the US
20:59.20coplandhas anyone used stanaphone with asterisk and had it work
20:59.30Drukenmikefoo: yeah
20:59.39mutit's not just because it's 'troublesome'
20:59.57brookshireit's because asterisk does not support t38
21:00.05mutneither does the cisco
21:00.10mutit's just sending it via sip to the cisco
21:00.17mutno special encodings
21:00.21mutjust ulaw
21:00.32poisonermikefoo: give number
21:00.38Drukenmight help.. hehe
21:00.54brookshiremut: the problem still exists with voip
21:01.05*** join/#asterisk wunderkin (i=kev@69.26.192.234)
21:01.05mutheh
21:01.06drfoomod2robin_sz: that;s like the old joke the emacs is a great OS, it just lacks a good editor
21:01.11muti'm deoubling the voip and it works better
21:01.14mutdoubling*
21:01.24mutif i remove one of the VoIP hops then the problem starts
21:01.36muthow can 'voip/sip is troublesome
21:01.39mut<PROTECTED>
21:01.44Drukenmut: doesn't the need for a foip machine? :) i'd like to see a sip/t38 compliant fax machine
21:02.04brookshiremut: jitter messes up faxes
21:02.19mutso theres more jitter by taking out a voip hop?
21:02.25justinu|laptoppacket loss messes up faxes
21:02.33justinu|laptopsince there's no PLC
21:02.34mutso theres more packet loss by taking out a voip hop?
21:02.51MikeJ[Laptop]is sip jb in tree now?
21:02.56brookshiremut: with sip, unless you are doing a passthrough, it's just going to talk directly endpoint to endpoint
21:03.19mutasterisk stays in the call
21:03.27mutit doesn't bridge it
21:04.02Drukenasterisk doesn't handle sip handoff very well, imo
21:04.19mutor does it and somehow
21:04.24*** join/#asterisk Disgrntld (n=ahahah@CPE-65-30-153-8.wi.res.rr.com)
21:04.28muthm
21:04.54*** part/#asterisk mikefoo (n=mikefoo@64.124.169.2)
21:04.59Disgrntldim tryng to make a call from extension 5000 and i get a recording saying that extension 5000 is unavailable? what could be wrong?
21:06.02VoicePulsemut: Why has the PCI card been ruled out as the cause of the problem?
21:06.20muti didn't
21:06.21mut[16:57:47] <brookshire> mut: i can tell you one thing.. the pci card has nothing to do with it
21:07.03mutVoicePulse: i also used to have a te405 with a tellab echo can
21:07.06mutand it had the same problem
21:07.14sevardDisgrntld: Sounds like you're trying to call yourself.
21:07.15mutbut that setup sucked period,
21:07.26Disgrntldsevard: i know its odd
21:07.35Disgrntldsevard: i swear i am not though =)
21:07.42*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
21:07.48DrukenDisgrntld: check your phone is in the correct context?
21:08.03*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
21:08.11Disgrntlddruken: ok
21:08.22VoicePulsemut: So is echo cancellation on or off right now?
21:08.36mutit's on
21:08.59Drukenecho can should be off for faxing
21:09.02VoicePulsemut: Try turning it off
21:09.03mutyea
21:09.10mutzaptel should disable it no?
21:09.13muton fax tone
21:09.18*** join/#asterisk jsaunders (n=root@216.86.121.58)
21:09.24*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
21:09.24brookshireheh..
21:09.27VoicePulsemut: Who knows... that may be the problem.
21:09.42mutwell i can't turn it off, thats the whole reason for buying the card
21:09.55MRH2hi anyway to continue monitoring a call in a single file post call transfer (SIP)
21:09.57jsaundersneed urgent help:  Can anyone tell me how to get around SIP "407 Proxy Authentication Required" problem w/ inbound calls w/ asterisk?
21:10.41Drukentry authenticating to the poxy ?
21:10.44*** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
21:10.46justinu|laptoptry setting insecure=very in your peer entry
21:10.51muti dunno
21:10.53jsaundersTesting....
21:10.59VoicePulsemut: You can also try messing with the volume/gain settings, especially output gain.  But echo cancellation has to be off.
21:11.44jsaundersinsecure=very does not help  :(
21:11.57jsaundersCould it have anything to do w/ the fact that pbx is behind router and set to dmz?
21:12.02muti've been trying to mess with gains
21:12.08jsaundersie, switching address.  I am thinking of this because it says "proxy"
21:12.12mutbut * needs restarted after i do that doesn't it?
21:12.18Drukenjsaunders: did you restart asterisk ?
21:12.26jsaundersDruken:  I did a "reload".
21:12.39Drukentry a restart, just incase :)
21:12.46jsaundersDruken:  I wish.  Live system.
21:12.57Drukenrestart when convient
21:13.01jsaundersA reload should suffice in this instance, only context changes.
21:13.18jsaundersugh
21:13.27jsaundersI am completely baffled by this one.
21:13.28mercestesIf you changed addy's your NAT/ARP translations are screwed.
21:13.37VoicePulsemut: Yes, and the drivers might have to be unloaded/reloaded too.
21:13.47mercestesrestart fixes it.
21:14.00muti'm going to have to take a fax machine to my house or something
21:14.05muttest it at 5am
21:14.06jsaundersThe call's making it to the pbx properly as it is set as dmz in router.  All ports makes it fine.
21:14.09justinu|laptopyour asterisk system is behind a NAT?
21:14.13*** part/#asterisk BadPacket (n=root@unaffiliated/badpacket)
21:14.25jsaundersThe SIP INVITE is hitting pbx just fine, pbx is returning first a SIP 407 followed by a SIP 403 to provider.
21:14.26justinu|laptopdoes asterisk use a non-routable ip?
21:14.28mutcuz i can dialup thru my voip in the office
21:14.33mut31.2k connections every time
21:14.42mutlast all day lon
21:15.25*** join/#asterisk darylp (n=daryl_ju@63-208-162-59.digitalrealm.net)
21:17.19jsaundersjustinu: Yes, behind router (as dmz).  I know.  :(  Temporary.  Outbound works great.  First attempt at receive sip inbound, failing miserably.
21:19.38*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
21:20.21justinu|laptopjsaunders: did you set "externip" and "localnet" in sip.conf?
21:20.50Drukeni don't think that is his problem....
21:21.09justinu|laptopprobably not, since it works one way
21:21.12Drukenit's an auth issue, which is usually insecure=very, but he already did that
21:21.29justinu|laptopthen his sip peer entry must not be "just right"
21:21.33jsaundersyessir, insecure=very
21:21.51Drukenjsaunders: do you register to the proxy ?
21:21.58jsaundersFor outbound, yes.
21:22.11jsaundersErr, I register for outbound to provider.
21:22.11Drukenwhat about inbound?
21:22.20jsaundersInbound, the provider does not register w/ us, no.
21:22.29jsaundersNor should he have to, as far as I know.
21:22.42justinu|laptopyou have to register for inbound calls usually
21:22.46jsaunderscontext=from-trunk
21:22.48Drukengenerally.. hehe
21:22.50poisoneranyone here who has some experinces with chan_sccp ?
21:22.51jsaundersfreepbx (amportal)
21:22.55justinu|laptopthat's how the ITSP knows whre to find you
21:23.08Druken~amp
21:23.09jbotamp is probably "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)"
21:23.28jsaundersDruken:  Heheh.  amp is just a shiney wrap around for *.
21:23.31jsaundersSame config files.
21:24.04Drukenyeah, amp is the condom for asterisk, i know, but i still like trojan
21:24.20jsaundersHeh
21:24.32Hmmhesaysthey fit better then durex
21:24.39Druken#freepbx supports it
21:25.15DrukenHmmhesays: wouldn't know... haven't had to use one in 4 years...
21:25.28generalhanis there a way to have one of my queues directed to a different context in extensions.conf? i want to specify some rules for the members that are being dialed in this queue, and i cant do it in queues.conf
21:25.30Hmmhesaysyou shouldn't be barebacking
21:25.47Hmmhesayssend them into a different queue?
21:25.55DrukenHmmhesays: when she's living with me, and fixed... there was no need :)
21:26.21*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
21:26.25generalhanHmmhesays: well ... what i mean is ... i want to use ChanIsAvail before ringing the phones in the queue, and i cant do that once im in there, so i need to find a new way
21:26.41*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
21:26.47Hmmhesayswhy do you want to do such a thing?
21:26.53sevardHmmhesays: do you know jack about custom apps in musiconhold.conf
21:27.01generalhanHmmhesays: lol, [TK]D-Fender was asking me the same thing !
21:27.14Hmmhesaysdefine custom apps
21:27.20sevardmplayer => custom:/usr/local/bin/mplayer -dumpfile - -dumpaudio - http://64.236.34.97:80/stream/1005
21:27.27sevardthat's what i'm attempting
21:27.49generalhancause i cant turn the call waiting off on the aastra phones that i have; and my reps are bitching at me about the beeping in their ear. so im trying to get around that by using the chanisavail cmd
21:27.53Drukenstreaming audio for moh is a bad idea....
21:28.02sevardDruken: blah blah blah
21:28.29Drukengeneralhan: so limit the phones to 1 incoming channels
21:30.43generalhanDruken: i WAS using incominglimit=1 ... but its depriciated
21:30.44[TK]D-Fendergeneralhan : Start setting you agents as loca/1234@context, and set up a macro accordingly.
21:30.44Drukenas far as i'm concerned, office phones shouldn't be able to receive call waiting.. that's what voicemail is for
21:30.44dlynesIs there any way to tell whether asterisk dropped a call, or if the pri dropped the call?
21:30.44generalhanbut i dont even use agents
21:30.46Drukenwhen was it depricated?
21:30.52generalhan1.2.6 i think
21:31.03justinu|laptopdlynes: without looking at the protocol traces, i don't think so
21:31.20dlynesjustinu|laptop: ah...so no post-mortem method then, eh?
21:31.29[TK]D-Fendergeneralhan : You do now if you want to use dial-plan logic to limit your calls...
21:31.42justinu|laptopdlynes: unfrotunately, not with the stock CDRs
21:32.04generalhan[TK]D-Fender: great ... does that also mean that i need to have my users "login" and "logout" everyday ?
21:32.22dlynesjustinu|laptop: no way to tell by looking at the log, either?
21:32.29justinu|laptopdlynes: full log?
21:32.35[TK]D-Fendergeneralhan : Not necessarily.  Try using Local to define them statically, not as Agent/
21:32.45dlynesjustinu|laptop: mostly full log, yeah
21:32.58justinu|laptopdlynes: iirc, no... it won't log that info
21:32.59dlynesjustinu|laptop: I don't have all the debug crap turned on, but other than htat
21:33.08justinu|laptopyeah, you need the pri debug or sip debug
21:33.12dlynesah
21:33.14dlynessuckage
21:33.21justinu|laptopcan you code? it would be pretty easy to put that in, i think
21:33.52dlynesjustinu|laptop: Yeah, I can code
21:34.04justinu|laptopactually, you might be able to tell if asterisk hit a "hangup" in the dialplan, but that wouldn't tell you who released if a call was bridged with dial
21:34.10dlynesI wouldn't have a clue about where to start, though
21:34.24justinu|laptopchan_zap.so for the pri side
21:35.18dlynesOk, so what would I be looking for to signal a drop on the telco's part?
21:35.39justinu|laptopyou'd receive a q931 DISCONNECT from telco
21:35.55justinu|laptopif you drop the call, you SEND a DISCONNECT to telco
21:36.09dlynesah...and that's an error message I'd be able to detect easily?  i.e. asterisk has a predefine for that?
21:36.24justinu|laptoplemme take a quick look
21:36.32dlynesI've only glanced at the code...never actually gone in depth on it
21:37.19*** join/#asterisk GolobTGG (n=GolobTGG@BSN-77-78-87.dsl.siol.net)
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21:42.14justinu|laptopchrist it's a mess
21:42.21jsaundersSo does the SIP 407 being sent back mean Asterisk is challenging the INVITE?  I found somone saying a permit= should solve this but alas, it did not help.
21:42.41vader--is there any gui type switchboard software that works with asterisk?
21:42.41justinu|laptopcheck like 8812 in chan_zap.so
21:42.41dlynesjustinu|laptop: exactly why i haven't spent much time looking at it :)
21:42.56justinu|laptopthat's where it reads the PRI events off the dchannel
21:43.03vader--im looking for something our secretaries can use to transfer calls and see what lines are in use and all that jazz
21:43.13vader--also to see if someone is on a phone call?
21:43.23*** join/#asterisk linlin (n=linlin@c-67-184-230-198.hsd1.il.comcast.net)
21:43.24brodiemvader--, try FOP but I personally don't like it
21:43.27Hmmhesaysfop can do it, kind of clunky but it works
21:43.28vader--fop?
21:43.32Hmmhesaysflash operator panel
21:43.36justinu|laptopjsaunders: i think asterisk isn't looking at the correct peer entry for inbound calls
21:43.37brodiemvader--, asternic.org
21:43.56vader--how does it tie in?
21:44.07jsaundersThanks justinu, I'll look at that closer.
21:44.10Hmmhesayslook at the demo on asternic.org
21:44.12brodiemvader--, it uses the manager API
21:44.29justinu|laptopjsaunders: turn on sip debug, capture an inbound call and pastebin it if you want
21:44.49justinu|laptopjsaunders: strike that
21:44.56dlynesjustinu|laptop: ah...looks like if it got hung up, it'll say "got hangup"; if asterisk hangs it up, it'll say "got hangup request"
21:45.05justinu|laptopjsaunders: turn on sip debug /and/ full logging, paste that full log
21:45.06vader--so you guys don't like fop?
21:45.19brodiemvader-- I find it too buggy
21:45.24justinu|laptopdlynes: cool, is that in your log?
21:45.31vader--he just released a new version the site said
21:45.35vader--have you tried that?
21:46.02brodiemvader-- notice it's dated 3/13.. but yeah I've used .25
21:46.13vader--gotcha
21:46.29vader--ya seems like it would be kinda hard to use for a larger organization
21:48.48*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
21:49.31*** part/#asterisk Ixthod (n=Ixthod@intellop.static.iaxs.net)
21:49.34brodiemvader-- it has some cool features but to me it just seems kind of sloppy
21:50.32brodiemvader-- I definitely wouldn't rely on it for a business to redirect calls though
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21:53.01Drukenbrodiem: it's nice to watch your extensions tho... i find it works good for that
21:53.28brodiemDruken yeah I agree, but it seems to sometimes show ghost calls
21:53.47mercestesor die for no known reason and require a restart
21:54.00brodiemand when it does you can't disconnect the imaginary call from fop
21:54.02Drukencan't say i've seen any of those... but i have seen it not see a call terminate...
21:54.20brodiemthen you need to restart it, and it loses all of its idle times, DND states, etc
21:54.21dlynesjustinu|laptop: It's in chan_zap.c, line 8847, 8912
21:54.29mercestesthat would be a ghsot call, Druken.
21:54.38jsaundersjustinu: will do, uno momento
21:55.12Drukenmercestes: yeah i guess....
21:56.00brodiemi wish there was something better without having to buy an entire pbx from a vendor who developed their own
21:56.15Hmmhesaysbetter?
21:56.17Hmmhesaysthan what?
21:56.22brodiemthan fop
21:56.30Hmmhesayswrite something op
21:56.32Hmmhesays*up
21:56.49brodiemit's about the only option..
21:56.51Hmmhesayshell you could do something in *.net easily
21:57.00justinu|laptopdlynes: so there ya go...
21:57.05Hmmhesaysyou can mash your face on the keyboard and come out with a working program with visual studio
21:57.13brodiemlol
21:57.38Drukenprintf "hello world";
21:57.47brodiemi don't use any win machines and don't ever plan to:)
21:57.56Hmmhesaysyeah but I bet your secretary does
21:57.59dlynesjustinu|laptop: would you happen to know if you can determine if the remote end hung up, or if the telco's switching eq screwed up, though?
21:58.02brodiemyup
21:58.13Hmmhesaysbrodiem: exactly
21:58.25Hmmhesayswasn't there something awhile back some dude wrote in vb
21:58.36brodiemip switchboard?>
21:58.59dlynesjustinu|laptop: sorry to be a pest...I know pretty much zero about pri signalling
21:59.29justinu|laptopdlynes: the best you can do is look for an abnormal cause code
21:59.36justinu|laptopdlynes: meaning something other than 16
22:00.07Drukendlynes: seems like your looking for a way to detect if the telco's equipment screws up, do you think they will give you a break on the cost if a screw up happens once in a while ?? hehe
22:00.48Hmmhesaysbrodiem yeah that was it
22:01.33brodiemHmmhesays, seems to run better but with the features you'd may as well just run astman
22:02.59*** join/#asterisk denon (i=denon@synapse.subneural.net)
22:02.59*** mode/#asterisk [+o denon] by ChanServ
22:03.44*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
22:07.00distortionis there a good way to hunt down the cause of "*** glibc detected ***" while using the safe startup script?
22:07.51dlynesDruken: No, but I would like to determine if the cause is the remote end hanging up, or Group Telecom screwing up
22:08.09dlynesDruken: if it's Group Telecom, I can threaten to move my business to Telus
22:08.25dlynesDruken: Or Allstream, for that matter
22:10.56*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
22:11.17dlynesdistortion: Are you using any of the gtk or kde modules for asterisk?
22:11.43dlynesdistortion: i'm guessing you're using the gtk module
22:12.06distortiongtk? no i dont htink so, my modules.conf is failly stripped down
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22:19.45distortionhmm, lcdial module seems like its the culprit  guess it cant handle it when you're sending ~8-10 calls a second.
22:20.26websaedistortion: do you know anything about predicitve dialers for asterisk?
22:20.55distortionno
22:21.42*** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk)
22:21.46dlynesdistortion: reason i was asking is because gtk applications will typically output that to the x console when they start up
22:22.35dlynesdistortion: it's not an error; it's just merely a notice to let you know it was able to find your glib library (general utility routines)
22:22.40*** join/#asterisk CpuID (n=none@gentoo/contributor/cpuid)
22:22.41distortionthat was the error im getting when i put asterisk under a heavy load while using lcdial.
22:22.52distortionwell, the asterisk app dies when i see that error
22:23.06distortionmagically stops with no .core file
22:23.27distortionand hundreds of calls get dropped :(
22:24.10dlynesdistortion: maybe libmysqlclient.so uses glib
22:24.16*** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com)
22:24.17dlynesdistortion: lcdial uses that shared object
22:24.51g__Question: is it safe to use '+' as a dialplan extension?
22:26.06g__ie, exten => +16135551234,1,Dial(Zap/g1/${EXTEN:2})
22:26.11tainted-distortion dump lcdial and get a real lcr
22:26.19tainted-jk
22:27.13CunningPikeg__: Try it and see
22:27.25dlynesbtw, does anyone know how to dial a couple of digits, do a pause, and then dial some more digits on a sip channel?
22:27.26g__CunningPike: it appears to work.
22:27.33CunningPikeThere you are, then
22:27.36CunningPike:)
22:27.42tainted-dlynes live channel?
22:28.11dlynestainted-: Say like Dial(Zap/g1/*70w6041234567), but on a sip channel
22:28.29tainted-yea
22:28.40g__CunningPike:I was kinda hoping someone would say "oh, we do it all the time and it's fine" or "don't go there!  There are bugs all over the place!  Beware!  Spoooky!"
22:28.42tainted-it's i think the D() option in dial
22:28.54dlynestainted-: that only works after the call's been answered, though
22:29.02tainted-oh u need before?
22:29.05dlynestainted-: i need this to happen after i get a second dial tone
22:29.19CunningPikeg__: I understand - but, if you were to wait for that all the time, you'd never get anything done ;)
22:29.20g__Is that a sip provider you have?
22:29.32generalhan[TK]D-Fender: you still here ?
22:29.33tainted-well technically the channel is answered at that point
22:29.38fourcheez-awaydlynes It must be answered to get a dial tone
22:29.43g__CunningPike: that's a good point.  So.. anyone know the answer? :)
22:29.45dlynesfourcheez-away: oh
22:30.00dlynesfourcheez-away: ok, I figured someone talking on the other end was considered to be an answer
22:30.15tainted-no
22:30.18fourcheez-awayno, ther'es a very presise definition of answered in SIP
22:30.21CunningPikeg__: afaik, '+' isn't any kind of reserved character for an extension name, so I think you're good
22:30.35g__That's not always the case.. but if you don't have to worry about that consider yourself lucky, dlynes.
22:30.36distortiontainted: real lcr?? ;)
22:30.58dlynesah
22:31.08fourcheez-awaydlynes, think about it - there's no real ringing going on is there?
22:31.10tainted-distortion yea .. something mysql driven
22:31.17distortionlcdial is mysql
22:31.21tainted-oh
22:31.25distortionyup.
22:31.33dlynesfourcheez-away: nope...it's just a voltage issue, and a q931 response sent back
22:31.42g__CunningPike: I suppose to test this theory, I should try it out and run a large test case that somehow involes a multi-play network game.. it would certainly waste enough time, right?
22:31.51CunningPikelol - sure would
22:31.55CunningPikeI like your style
22:32.00tainted-dlynes try the D(435454545wwwwfafe) which each 'w' representing a wait 1 second
22:32.05tainted-u'll like it
22:32.14g__system-administration-by-doom
22:32.24fourcheez-awaydlynes, no in sip there isn't even a voltage
22:32.35g__unless you lick the wires
22:32.40fourcheez-awayonly the fact that the client has said that it's answered
22:32.42tainted-nope
22:32.45tainted-not even if u lick wires
22:33.04g__I'm sure it could happen..
22:33.10g__you can't be too careful.
22:33.14tainted-nope
22:33.15tainted-it can't
22:33.35anthmi think w is 1/2 second
22:34.27tainted-really?
22:34.33tainted-i wish it was documented someplace
22:34.41tainted-voip-info's got nothing
22:34.48anthmyah i only know cos i made that function
22:35.11tainted-the 'send dtmf' in live sip channel?
22:35.14tainted-or dial()
22:35.44anthmit's in app.c as a public func called ast_dtmf_stream
22:35.56tainted-wow
22:36.03anthmthen used in mods like dial and senddigits
22:36.05tainted-you've really contributed a lot
22:36.12g__Someone have time to add it to voip-info?
22:37.44g__Nevermind, I'll add it.
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22:40.07g__done.
22:41.53*** part/#asterisk CpuID (n=none@gentoo/contributor/cpuid)
22:44.04*** part/#asterisk jake1932 (n=Administ@68.236.22.143)
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22:53.05dlynesfourcheez-away: yep
22:53.35dlyneson another note, I'm having the following problem when trying to load wcfxo or ztdummy:
22:53.38dlynesZT_CHANCONFIG failed on channel 1: No such device or address (6)
22:53.38dlynesFATAL: Error running install command for ztdummy
22:53.40RoyK<PROTECTED>
22:54.05dlynesI've got a cheapo x100p card in the machine
22:54.39dlynesAnd the same error when I try to modprobe wcfxo:
22:54.39Qwell[]So why are you using ztdummy?
22:54.47dlynesZT_CHANCONFIG failed on channel 1: No such device or address (6)
22:54.47dlynesFATAL: Error running install command for wcfxo
22:54.51Dr-Linuxhey dlynes
22:54.53dlynesBecause wcfxo isn't working, either
22:54.55Qwell[]check dmesg
22:55.49dlyneswaiting for pastebin to become available
22:57.42dlyneshttp://pastebin.com/693270
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22:59.43dlynesIt's got some definite errors there, but I have no idea what the real error is
23:01.26dlynesCould it just be having an issue because it's sharing an irq?
23:01.39ManipuraI have a IAX phone and a DID.. But no asterisk server anymore.. Are there any services out there that allow you to point a DID to them and route it to your phone?
23:01.49tainted-probably
23:01.54tainted-dlynes what kind of card
23:01.56*** join/#asterisk c4t3l (n=robert@cpe-24-175-57-117.houston.res.rr.com)
23:01.57tainted-digium?
23:01.58dlynestainted-: x100p.com
23:02.05tainted-gag
23:02.27dlynestainted-: well, seeing as how digium doesn't make the x100p card anymore, i didn't have a choice, but to buy a clone
23:02.55tainted-it's basically an analog modem
23:03.00dlynesyep
23:03.05dlynesand it's a pci device
23:03.12tainted-i had great luck with a clone
23:03.13dlyneswhich, technically should be capable of sharing interrupts
23:03.16tainted-forget which one
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23:04.21generalhanCan some one please take a look at my recording setup and advise me on a way to get this to work properly ?? http://generalhan.pastebin.ca/52716
23:04.57generalhanthe calls are not recorded i think becuase * is fighting over where to save the recording ... so i need to figure out a bitter way to do this to make it work correctly
23:06.04[TK]D-Fendergeneralhan : I don't think you can ask * to make a full pasth like that on demand.
23:06.21[TK]D-Fendergeneralhan : May all your time/date vars part of the FILENAME, not the PATH
23:06.47[TK]D-Fendergeneralhan : Its bad enough it hopes to have a folder ready by SIP account
23:06.55generalhan[TK]D-Fender: well i do it on an idividual basis cause thats the same macro i use for regular calls. i just need to know how to make it work for multiple calls
23:08.02generalhan[TK]D-Fender: i got the queue to dial out to the people and do the ChanIsAvail first ... so thats taken care of .. if i can get it to record correctly (or record AT ALL i guess) then ill be almost done
23:08.03[TK]D-Fendergeneralhan : Follow my ealier advice about removing all other vars from the path than the EXT
23:08.15generalhan[TK]D-Fender: ok ill try that
23:10.54generalhan[TK]D-Fender: still no dice ... on the CLI i see it come up and say that its starting the recording for all the extensions that it is dialing... and i think that is where my problem lies. its like * is fighting over which directory to store this in and it just gives up
23:13.16[TK]D-Fendergeneralhan :PAstbin your new macro
23:13.27[TK]D-FenderAnd VERIFY your path for your last attempt.
23:14.08generalhan[TK]D-Fender: i tried it this way too
23:14.09generalhanhttp://generalhan.pastebin.ca/52717
23:14.24generalhanto specify only one place to record it to ... still no go on that either
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23:15.30[TK]D-Fendergeneralhan : Stop using the date for the folder!
23:15.37generalhan[TK]D-Fender: i cant
23:15.48[TK]D-Fenderyes you can, and you WILL.
23:15.59[TK]D-FenderAre you going to premake EVERY FUTURE VALUE right now?!
23:16.31generalhan[TK]D-Fender: i work in a Law Office and it is the attorney's policy to record all calls to be EASILY referenced .... so they need to be in folders by user and by date
23:17.10*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
23:17.30[TK]D-Fendergeneralhan : Parse it out late.  use a script, whatever, but nothin in there is guranteein your path!
23:18.06generalhan[TK]D-Fender: http://generalhan.pastebin.ca/52718      that is how i have been doing it for regular calls and all incoming calls have been recorded just fine ( IF they are called directly from the autoattendant or transferred from inside the office)
23:18.38generalhani also have one for outgoing calls as well ... all that works perfectly ... until i try to use it the same way to record out of the queues
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23:19.52[TK]D-FenderI dunno...
23:20.17generalhan[TK]D-Fender: when i specify the date as the folder, as soon as the call starts it automagically creates that folder for me to put the call into
23:21.03generalhanthat way when my boss says that an attorney says we DID NOT call him on this date .. i can go right to the date pull up his phone number and play back the conversation for him on the spot
23:21.08[TK]D-Fendergeneralhan : Never seena  system that would invent a whole path jsut to support a new file.
23:21.15[TK]D-Fendergeneralhan : But hey, whatever
23:21.19tainted-does anyone have the SNL clip of the IVR lady
23:21.20generalhan[TK]D-Fender: really ?
23:21.32websaetainted- that was hilarious from SNL haha
23:21.35[TK]D-Fendergeneralhan : Just try to seperate HOW the system works, from the fact of it working.
23:21.47generalhan[TK]D-Fender: what do you mean ?
23:21.55[TK]D-Fendergeneralhan : You could put it all in the name for all it matters as log as its standardized
23:22.32generalhanwell if im using a macro for all incoming and outgoing calls i would say thats standardized
23:22.46generalhannothing gets to an extension without passing though those rules
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23:23.11generalhanits just that when the queue is trying to save 1 call to 15 different places it doesnt seem to want to work.
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23:54.40philippelCorydon: are you there?
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23:56.17MRH2hi just upgraded to latest svn1.2 stable and it ain't compiling??
23:58.03MRH2any known issues atm?
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