irclog2html for #asterisk on 20060429

00:00.21asterboyok, I pointed at the gs site and the thing still won't upgrade.
00:00.23bzbwok, here is the related config:  http://pastebin.ca/52124
00:00.30asterboyWhere is the self destruct button.
00:00.39asterboythis crap is going to the garbage bin.
00:00.46asterboyfucking garbage
00:00.57bzbwI don't understand why it is not working:(
00:01.25ghost99Manxpower: .. when you said that milliwatt thing how does that go ... I type in in at centos promt or CLI and it doesn't like it ....
00:01.30bzbwasterboy: what's the symptom on your upgrade, I happened to have a few GXPs.
00:01.32[TK]D-Fenderexten => _2.,3,Dial(sip/BroadVoice/${EXTEN:1})
00:01.44dlynestainted-, prices have gone up since the last time i looked
00:01.54asterboyit just won't even initate an upgrade
00:01.56bzbwD-Fender: same thing, I change it to that, same story
00:01.59ManxPowerghost99, you REALLY need to read The Book
00:02.04ManxPower~thebook
00:02.06jbothmm... thebook is Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
00:02.06asterboyit's ignoring the files in my tftp root
00:02.22[TK]D-Fenderbzbw : pastebin sip debug of a complete call attempt
00:02.27asterboybzbw, what version do you have installed?
00:02.37bzbwasterboy: u have a trace file(in ethereal)?
00:02.38ManxPowerghost99, you can't run apps from the Asterisk CLI
00:02.54dlynesasterboy, what ftp server are you using?
00:02.59dlynesasterboy, erm tftp i mean
00:03.03CunningPikeasterboy: I upgraded one once when were trialing them, and I think all I did was what you said
00:03.26asterboyha
00:03.33asterboyI mean tftp-ha
00:03.47dlynesasterboy, yeah...you shouldn't have any problems with that one
00:04.16asterboyIs there a way to get more logging out of tftp-ha
00:04.17asterboy?
00:04.25dlynesasterboy, is the tftp server's time synced to the same source as your polycoms?
00:05.11dlynesasterboy, you mean besides -vvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv?
00:05.14asterboyyes
00:05.26asterboyno I mean with tftpd
00:05.37dlynesasterboy, yeah...specify more v's for more verbosity
00:05.37bzbwasterboy: I've heard that they are releasing the newer one soon, maybe next Tuesday:)
00:05.48asterboyI
00:05.56asterboyI'll be kicked out of here by then
00:06.12asterboyBesides it won't matter if I can't upgrade
00:08.21dlynesasterboy, one of the asterisk developers lives in Calgary
00:08.29dlynesasterboy, maybe he'll help you for a hefty fee?
00:09.35bzbwasterboy: call their LA office now, they are still working, I know a couple guys there can help a lot:)
00:09.39dlynesHe used to live in Langley, but he moved to Calgary recently
00:10.07dlynesbzbw, He bought Polycoms as an end user
00:10.50dlynesasterboy, did you not buy the polycoms from a polycom channel partner?
00:10.55asterboy(626) 956 0260
00:11.01asterboynope, ebay
00:11.09dlynesah...suckage
00:11.23asterboylol, they have "PLease DIAL..." on there answer service.
00:11.27asterboysays it all
00:11.33asterboywho the hell DIALS anything
00:12.51[TK]D-Fenderasterboy : Whats the problem with them now?
00:13.02CunningPikeWon't upgrade, apparently
00:13.06asterboyyep
00:13.19[TK]D-FenderPolcom's?
00:13.54CunningPikeYou're about 5th in line [TK]D-Fender lol
00:14.20[TK]D-FenderTFTP doesn't upgrade based on file date IIRC....
00:14.37[TK]D-FenderPart of why I always use FTP personally.
00:14.55dlynes[TK]D-Fender, ah...thought it might because my sipuras autoupgrades were flaky as hell, too
00:15.20dlynesnot to mention sipura autoprovisioning
00:15.34[TK]D-FenderTFTP = slightly dumb.  FTP=better
00:16.30*** join/#asterisk franck (n=franck@tikiwiki/franck)
00:16.30asterboyooooo...Grandstream is giving me the latest firmware!
00:16.35CunningPike[TK]D-Fender: Correct about tftp
00:16.38asterboyand yes ftp is cooler
00:16.54asterboybut no choice with GXP
00:16.55dlynespolycoms don't do https?
00:17.10[TK]D-Fenderdlynes : Certainly they can.
00:17.22dlynesjust ftp is easier to set up?
00:17.59[TK]D-Fenderdlynes : That too.
00:18.20franckHi all
00:19.00franckI'd like to make a script that sends me an e-mail each time a SIP phone register outside a predefined range of IP addresses, How to?
00:22.26VahramAnybody can help me with asterisk development, i whant to patch chan_h232.c to set variable tat will have Q931cause
00:22.53Vahramactyally i need only know the puction that is putting channel vars
00:23.08Vahram*
00:24.04justinugood afternoon #asterisk
00:24.43VahramI guess everybody is sleepin))
00:33.00*** join/#asterisk MikeJ[Laptop] (n=vircuser@64.241.37.140)
00:34.35CunningPikeWell, it's been a slice
00:36.05*** join/#asterisk De_Mon (n=de_mon@fl-69-69-140-189.dyn.sprint-hsd.net)
00:37.01*** join/#asterisk BadPacket (n=BadPacke@unaffiliated/badpacket)
00:39.53bzbwasterboy: did you get the new firmware?:)
00:39.58asterboyyep
00:40.04bzbwwhat was it?
00:40.06asterboylatest greatest and it is far more impressive
00:40.14bzbwwhat version is it?
00:40.40asterboy1.1.0.10
00:41.04*** join/#asterisk Thazza (n=me@229.9.233.220.exetel.com.au)
00:41.05asterboyand Bootloader 1.1.0.1
00:41.12bzbwman, it's almost the latest one, but they said it should be 1.1.0.11:)
00:41.18*** join/#asterisk yxa (n=diablo@58.185.90.101)
00:41.19asterboythe menu is the way it should be now.
00:41.34asterboylet me know if you want it?
00:41.38*** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com)
00:42.17bzbwIs it working?
00:44.37bzbwk, my issue is yet to resolved:(
00:45.02asterboyhaven't tested yet
00:45.12Thazzaanyone got a Sipura SPA-3000 working without echo on the FXO?
00:45.31bzbwman it's Dial(SIP/${EXTEN:1}@BroadVoice), it just did NOT resolve to the right host:(!!!
00:46.59bzbwanyone knows how to step through * for debugging? Into the Context translation level?
00:47.24asterboyyippeee!!!
00:47.35asterboythe phone was causing the transfer problem
00:47.42asterboynew upgrade fixed it.
00:48.09asterboyI have a natural knack for finding those problems no one else finds...drive me nuts
00:50.25DoktorGregYou just need to think more like everyone else:P
00:50.35[TK]D-Fenderasterboy : no, plenty of other people got chumped into buying GS' :D
00:51.25asterboywell, this is the first time I caved in cause the client couldn't afford the Polycom
00:51.31asterboyso now I'
00:51.40asterboyI'm the one who pays in the end.
00:52.23*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
00:52.47paolobHi guys! How do I put a call on hold? is there a special key?
00:53.01[TK]D-Fenderpaolob : depends on your phone.  What do you have?
00:53.05De_Monasterboy so ture
00:53.20paolob[TK]D-Fender, normal phones connected to pap2
00:53.33asterboymore grey hair....just great.
00:53.51[TK]D-Fenderpaolob : then its up to the PAP2 to put the call on "hold"  there may be a flash + *code feature in the ATA for that.
00:53.54asterboythey'll be pilling dirt on my early at this rate
00:54.20paolob[TK]D-Fender, doesn't asterisk use a special key, like # to transfer calls?
00:54.23[TK]D-Fenderpaolob : The SPA series from them has a code for it, but not sure about the PAP2 specifically.
00:55.05[TK]D-Fenderpaolob : # is just a way so that shit phones and analog devices which don't have special signalling capabilities can tell # of its intent.
00:55.09[TK]D-Fender*
00:57.24paolob[TK]D-Fender, explain better, I don't understand...
00:58.30jake1932# also works on cell phones
00:59.05[TK]D-FenderReal SIP phone had hold/transfer/conference buttons, etc.  just like digital PBX phones.  You are plugging a DUMB analog phone int an ATA.  So you need to tell the ATA to put the call on hold with the only tools at your disposal : Flash and DTMF.
00:59.21[TK]D-FenderNow thats assuming the PAP even OFFERS you the capability of telling it to do that.
00:59.28sevardhttp://video.google.com/videoplay?docid=-3857855347623051125
00:59.38[TK]D-Fenderjake1932 : to transfer calls?
00:59.43jake1932yes
00:59.53[TK]D-Fenderjake1932 : I presume you mean coming in on a Zap cahnnel of course...
01:00.06[TK]D-Fenderjake1932 : Either way its an * mechanism.
01:00.09jake1932well sort of - VIOP term to cell phone
01:00.10*** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-51-118.dsl.irvnca.pacbell.net)
01:00.49*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
01:00.57jake1932just wanted to point out that it's more than just crappy phones and analog
01:01.24*** join/#asterisk mbrooks (n=mbrooks@gateway.digium.com)
01:01.36jake1932it's anything except IP phones connected directly to asterisk
01:02.40*** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-51-118.dsl.irvnca.pacbell.net)
01:06.12bzbwasterboy: u can't transfer the call with new firmware?
01:06.32*** part/#asterisk franck (n=franck@tikiwiki/franck)
01:07.05DoktorGreganybody know of a way to get adobe illustrator out of this danged MDI interface?
01:07.14bzbwspeaking of video, anyone knows whether 1.2.7.1 support H.264?
01:07.28*** join/#asterisk zeppelin__ (n=zeppelin@201.66.166.237)
01:08.56skydogsevard: pretty cool stuff
01:09.48jake1932yep - i could be at two places at once finally
01:09.56sevardskydog: speculation that it's what is inside of the nintendo revolution
01:09.59ManxPowerbzbw, Did any other 1.2x support it?
01:10.17skydogcool
01:10.25sevards/it\'s/is/g
01:10.47sevardskydog: speculation that it's what is inside of the nintendo revolution
01:10.51sevards/it's/is/g
01:11.00sevard:|
01:11.03paolobGuys, can I use a variable this way in extensions.conf : exten => ${VAR},1,Dial(${VAR},20,Tt) ?
01:11.13sevardi'm tired.
01:11.32[TK]D-Fenderpaolob : No, you can't use a variable as an EXTEN, only a CONSTANT.
01:11.39skydoganybody running asterisk as non root?
01:11.56jake1932paolob: use Goto
01:12.04sevardskydog: i am
01:13.05skydogI think ive got it...but the start up is simply starting it by - su asterisk -c /usr/sbin/safe_asterisk or dropping into asterisk to start?
01:13.45ManxPowerskydog, read the wiki page on running Asterisk as non-root
01:13.57skydogyep im on it now..
01:14.09bzbwManxPower: Mark told me at one point it does:(
01:14.13sevardskydog: I thought safe_asterisk was a script that also ran asterisk as a different user, i don't know.  i don't use that script.
01:14.28ManxPowerbzbw, 1.2 gets no NEW features.
01:14.35skydogasterisk -U asterisk -G asterisk ? maybe??
01:14.39ManxPowerif it's always been a 1.2 feature, then the latest should have it.
01:14.52sevardskydog: man su
01:15.02ManxPowerskydog, yes, but then you also need to change the logging permissions, the zaptel device permissions, the database permissions.  READ THE WIKI PAGE
01:15.03sevardand man sudo
01:15.19skydoghttp://www.voip-info.org/wiki/index.php?page=Asterisk+non-root  is what im referring to.
01:15.24ManxPoweryou do NOT need to sudo or su.  safe_asterisk can be told to run asterisk as non-root
01:15.39ManxPowerskydog, if those instructions don't work, get back to us.
01:16.01paolob[TK]D-Fender, how do I define a CONSTANT?
01:16.12skydoggood enough..:) ..so far it has been successful minus these parts -
01:16.15paoloband how do I reference it?
01:16.38skydogno such location for /usr/local/share/asterisk ..
01:16.46*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
01:16.48skydogon centos install that is.
01:17.02[TK]D-Fenderpaolob : What are you really trying to DO?
01:17.22skydogand im guessing since ztdummy is installed /dev/zap doesnt exist either...right?
01:17.44*** part/#asterisk poisoner (i=poisoner@tauceti.poisoner.de)
01:18.01paolob[TK]D-Fender, I'm trying to write extensions.conf in a way that permit me change the ext. number without changing all the occurrences of each ext. in extensions.conf
01:18.19[TK]D-Fenderpaolob : And what is this an exten TO?
01:19.07skydogperhaps the use of macros will help? with the extensions ??
01:20.05paolob[TK]D-Fender, for example, I define 701 as my phone number, but I want to have the freedom to change "my" phone to 702, and I want to write extensions.conf in a transparent way
01:20.35paolobin a way that a change 701 to 702 only once, and I reload the confs, and all is ok
01:21.02[TK]D-Fenderpaolob : how big a PBX setup are we talking about?  Thi just doesn't seem worth it....
01:21.21paolob[TK]D-Fender, 30 extensions
01:21.37[TK]D-Fenderpaolob : Not worth it....
01:21.44[TK]D-Fenderseriously
01:22.04*** join/#asterisk miguel3239 (n=chatzill@ns1.nashuacs.com)
01:22.11paolobbut how do I define a constant in extensions.conf ?
01:22.33paolobMYPHONE=701 ?
01:22.45paolobMYPHON E => 701 ?
01:22.53[TK]D-Fenderpaolob : look in the sample and check the WIKI
01:23.04paolob[TK]D-Fender, ok, thank you!
01:23.15[TK]D-Fenderpaolob : but that looks about right.
01:25.46natmltpaolob in the book "Aterisk, The Future of Telephony" that you can download for free, check out Global Variables on page 91
01:25.55jake1932you could do 701,1,Goto(myphone,1)
01:26.08jake1932then have myphone,1,Blah
01:26.16paolobnatmlt, but are constant only predefined, or can i define my own ones?
01:27.18natmltpaolob you can define your own
01:27.19paolobnatmlt, how?
01:27.19natmltpaolob their example JOHN=zap/1 for dialing a user John over zap/1
01:27.19wunderkinmust be experiencing packet loss
01:27.34paolobnatmlt, and how do I reference that constant?
01:27.47natmltPut it in the context [globals]
01:27.57natmltit is reserved strictly for global variables
01:28.15natmltand the will apply to all contexts in extensions.conf
01:28.31natmltYou should download the book
01:28.38natmltit will help a bunch
01:29.03[TK]D-Fenderjake1932 : That sample is HORRIBLY wrong...
01:29.24natmlthttp://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
01:29.30[TK]D-Fendernatmlt : much better sample...
01:30.09[TK]D-Fenderpaolob : per natmlt's sample it'd be - exten => ${JOHN},1,Dial(SIP/myphone,30)
01:30.11jake1932hmm - should work
01:30.19jake1932lemme try
01:30.26skydoganother good write up on configuring * as non-root : http://www.dynx.net/ASTERISK/AMP/INSTALL
01:30.32[TK]D-Fenderjake1932 : Yes it'd work, but thats got nothing to do with "constants"
01:30.53jake1932no - but it solves what he asked originally - trying to alias extensions
01:31.22[TK]D-Fenderjake1932 : not to alias an exten, then he'd have to change the "701" in a ton of places.
01:31.45jake1932not true
01:31.52jake1932only in the goto statement
01:33.28jake1932no?
01:34.13[TK]D-Fenderjak he wants to change the very fact of "701" leading to his phone.  and when he has 10 occurences of 701 ledaing to him, what would he do to change it to 702?
01:34.45jake1932it would be myphone leading to him
01:34.50jake1932701 would be the alias
01:34.56skydogprobably a little complex, but he could put his stuf in a database and change it that way...just a thought..but that would make things just a hell of alot more complicated for 30 callers...
01:35.11jake1932realtime would be overkill
01:35.17skydogagreed
01:35.33skydogbut again a solution of sorts..
01:35.42jake1932yes
01:36.01[TK]D-Fenderjake1932 : he's talking about have many "exten => 701,1," all over his dialplan.  He wants to replace ALL OCCURENCES of 701 to something else.  You are talking about changing that action taken by 701,  he want 701 ITSELF to change.
01:36.38jake1932ok - i'm going to reread what he asked for
01:37.24jake1932<PROTECTED>
01:37.26skydogwell it is flat file and he could simply grep the file out and change the occurences of 701 wih one shot...but im truly confused now as to what the original requirement was...lol
01:37.31[TK]D-Fenderjake1932 : Mind you I can hardly imagine why he'd have more that 2 references to a given "phone extension number".  I always do cascaded contexts which inherit everything...
01:38.23[TK]D-Fenderjake1932 : and the exten # is 701, not the action it should take on being dialed.
01:38.41skydogthanks to peeps for the reference of the non-root install works like butta! ;)
01:39.48*** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com)
01:40.31jake1932regarding more than 1 reference, isn't that what you're doing by defining MYEXTEN anyways?
01:40.39jake1932as a constant
01:42.07jake1932he's probably not even here anymore
01:42.08jake1932:)
01:43.13[TK]D-Fenderbut you only have to change that constant at the top ONCE.
01:43.27[TK]D-FenderAnd all 10 times it appears follow along...
01:43.46jake1932you're saying with my example you'd have to change it more than once?
01:44.37[TK]D-Fenderseet this? <jake1932> you could do 701,1,Goto(myphone,1)
01:44.52jake1932yepp - that's exactly what i have
01:44.57[TK]D-Fenderhe's saying he'll have lots of 701's all over the place.  its a FIXED value there!
01:45.08jake1932oh
01:45.14jake1932i think it follow you
01:45.15[TK]D-Fenderand when he wants to change them ALL to 702?!
01:45.32jake1932he would change them to myexten (not 702)
01:45.47jake1932i'm going to pastebin
01:46.44*** join/#asterisk nortex (n=breeves@adsl-69-149-172-106.dsl.amrltx.swbell.net)
01:47.21[TK]D-Fenderhe wants to do : exten => ${MYPHONE},1,Dial(Zap/1,20) and so on in 50 differnt places and for the MOMENT use 701 and be able to change his mind later and not have to change 50 lines.... just the constant declared at the top.  Getting it now?
01:47.38jake1932yes
01:47.56jake1932http://pastebin.ca/52153
01:48.57*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
01:50.20[TK]D-Fenderjake1932 : Thats nice. but he'll have 100 occurances of the FIRST LINE like 701 throughout his plan.  Do you get it?  Each one would have to be changed.
01:50.40jake193230
01:50.53jake1932but i don't think there's any way around that
01:51.03jake1932you need to specify it somewhere
01:51.13[TK]D-Fendernot 30.  FOR 30 people with potentially multiple occurences of each throughout his plan.
01:51.33[TK]D-FenderSo 30 * # of occurences
01:51.51jake1932somewhere, whether you use a constant, or the method i put out there, you need to specify what myexten points to
01:52.59jake1932now - if he goes and changes his phone user ids, vm boxes, etc
01:53.06jake1932then - that's a different issue
01:53.12[TK]D-Fenderjake1932 : Your's has 701 HARD CODED.  Do you get it?  thats not a universally updateable CONSTANT.  if he puts a 701 in 10 contexts he'll have to chage all 10.  if he uses a constant he need only chage the constant declaration up top.
01:53.40ManxPower[TK]D-Fender, so basically he wants to create a complex, hard to understand dialplan just so he doesn't have to put in extensions.
01:54.02[TK]D-FenderManxPower : Quite possibly, but that wasn't my point :)
01:54.33*** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca)
01:54.33ManxPowerDialplans are complex, no matter how hard you try to make them simple.  You soon find that simple does not work in the real world.
01:54.43[TK]D-FenderManxPower : just trying to get across that he wants the EXTENSION modified globally, not the APPLICATION called NOR its PARAMETERS.
01:55.03[TK]D-FenderManxPower : Sure it can work, its just simply "not worth it"
01:55.23ManxPower[TK]D-Fender, I know.  Totally stupid thing to do and may or may not work in the current verison of asterisk but could EASILY not work in future releases.
01:55.34[TK]D-FenderManxPower : My dialplans are DAMN clean things as are the macros and other methods that support it.
01:55.53jake1932mine too
01:55.55ManxPower[TK]D-Fender, mine are CLEAN, but not SIMPLE.
01:55.56[TK]D-FenderManxPower : SHHH!!!! Don't destroy his hopes!  Let him "live for the moment"
01:56.13bzbwhmm, I've been troubleshooting with D-Fender on this, looks like broadvoice now require * to use "sip.broadvoice.com" in From, To and Contact header, if not, it will reject the call:(
01:56.15*** join/#asterisk bjohnson (n=bjohnson@i216-58-59-27.cybersurf.com)
01:56.53[TK]D-Fenderbzbw : Not really, thats copied right from a client using it right now....
01:57.05[TK]D-Fenderbzbw : PM
01:57.21jake1932"if he puts a 701 in 10 contexts he'll have to chage all 10" - nowhere did i see talk about multiple contexts - i can see what you're saying on that
01:57.48[TK]D-Fender<jake1932>  paolob: [TK]D-Fender, I'm trying to write extensions.conf in a way that permit me change the ext. number without changing all the occurrences of each ext. in extensions.conf
01:58.00[TK]D-Fender701 is the ext, not the ACTION it would take.
01:58.20[TK]D-Fendertherefor this statement implies multiple occurances of 701 <-
01:58.51jake1932701,1 701,2 ,etc
01:58.55bzbwD-Fender: I compare the sip registration, which uses all "sip.broadvoice.com" in all 3 headers with the Invite, registration goes through, but invite gotten 604.
01:58.56jake1932maybe i read it wrong
01:59.18[TK]D-Fenderjake1932 : those are PRIORITIES of ext 701 in a given context.....
01:59.42jake1932paolob: are you still here?
01:59.54[TK]D-Fenderbzbw : In what you pastebin'd for me I never saw a reject, it kept RETRYING only.
02:00.03paolobjake1932, yes
02:00.04De_MonI need help using dbg to track down this codec 128 WARNING I keep getting... I've connected gdb to the asterisk process, now what?
02:00.32bzbwD-Fender: I'll take your advise and try more, got to go now, thanks anyway:)
02:00.34jake1932paolob: were you using multiple contexts in which you need the dynamic extensions?
02:01.34jake1932or were all your extensions in a single context?
02:01.36paolobjake1932, yes, and it was for a matter of trasparency: someone that would have to modify asterisk configuration should understand clearly what are those ext. numbers referring to
02:03.30jake1932but that was a fun debate!
02:04.03ManxPowernobody needs dynamic extensions
02:04.10paolobguys, have config files changed their syntax from 1.0.9 to 1.2?
02:04.21Qwellpaolob: some, yes, of course
02:04.45tainted-Qwell!
02:04.52paolobQwell, if I copy a 1.0.9 sip and extensions.conf to a 1.2, what should I change?
02:05.02Qwellpaolob: Everything that the README files says to change
02:05.13Qwelleverything that the CHANGES file says to change
02:05.21paolobQwell, ok, thank
02:05.27QwellEverything that was discussed on the various mailing lists, and forums, and emails, etc
02:05.40jake1932yes - read it all
02:05.43jake1932haha
02:05.55jake1932there will be a wuiz
02:05.58jake1932qiz
02:06.00jake1932ah
02:06.02QwellYou fail
02:06.25jake1932should've paid attention in typing class
02:06.42[TK]D-FenderManxPower : I should remember a timeless addage "Arguing on the internet is like running in the Special Olympics.  Even if you win, you're still a RETARD"
02:07.04ManxPower[TK]D-Fender, hence my extensive /ignore list
02:07.31[TK]D-FenderManxPower : Fortunately I don't get repeat offenders.....
02:07.43jake1932i wasn't trolling
02:08.02[TK]D-Fenderjake1932 : Buy you we're caught ; hook, line, and sinker!
02:08.08dlynesDoes anyone happen to have a better alternative to Text::CSV_XS (Perl) for importing old asterisk csv files?
02:08.19dlynesI've encountered some csv lines iwht binary data in them
02:08.48[TK]D-Fenderjake1932 : Wow I typed that out SO wrong.. hehe
02:08.52[TK]D-Fenderits getting late...
02:08.53*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
02:09.01*** join/#asterisk Splas (n=jwb@brooklyn.paravolve.net)
02:09.01jake1932yes - see it's catchy
02:09.10[TK]D-Fenderadfgafgsdfhgkljhsjhfdkjtew;klnrg
02:09.17[TK]D-Fender*blarg*
02:11.26[TK]D-FenderManxPower : 501 display clearly isn't as nice as the 601, but still better than pretty much everything else except a Cisco 7940+
02:11.46[TK]D-FenderI'm betting its probaby close to tie with Cisco.
02:11.54[TK]D-FenderWish I had one for testing...
02:12.08jake1932is there a backlight on the +?
02:12.25[TK]D-Fenderjake1932 : is no "+" i mean that model and higher
02:12.33jake1932ok
02:12.40[TK]D-Fender7960, etc
02:12.44jake1932right
02:12.55jake1932how come they couldn't fit a backlight on these phones
02:12.56jake1932?
02:13.26jake1932i even have a cheapo 9417 nortel with one
02:14.02[TK]D-Fenderjake1932 : Dunno.... its amongst the most requested features.  Maybe they'll all wake up soon and pay attention.
02:14.13jake1932let's hope
02:14.54[TK]D-Fenderjake1932 : the Aastra 480i has a backlight but a chacter based display, not pixel.  Pretty much ever phone has a clearly counterbalancing factor against it.
02:15.26jake1932how much would a backlight up the phone cost though?
02:15.35*** join/#asterisk Tusker (n=tusker@203.117.94.152)
02:16.17[TK]D-FenderGrandstream GXP-2000 is the PERFECT example of this.  Has back-light, multiple keys for presence PoE and more for $85USD!  great value, right?  NO!  Shit speakerphone and handset.  NASTY echo... ona  SIP PHONE!!  Flakey ass firmware.  FEELS like something fit for Ken & Barbie...
02:16.29jake1932haha
02:16.41[TK]D-Fenderjake1932 : Probably not too much more... they just cut corners thinking the market they were targeting.
02:17.08jake1932the aastra is a business phone though
02:17.19jake1932same market
02:17.33[TK]D-FenderI'd HAPPILY pay a bit more for Full PoE and backlight options on all Polycom phones.
02:17.58jake1932well even with that - i have a 7960 on my desk - because of the general quality of the phone
02:18.00DoktorGregwooo hooo the pvr is done transcoding and commercial skipping doktor who!
02:18.02[TK]D-Fenderjake1932 : Yewah, but they just built that phone off their old analog model..... just changed the board.  not a "new" phone like all the others designed.
02:18.27jake1932right
02:18.51jake1932have you used the aastra phone before?
02:19.29Tuskerheya guys... sorry for such a newby question, but I was wondering if it is possible to call one #, and then dial the requested number, within a dialplan/extension... ie, like when you dial using a normal modem, and you have to put the , (delay) there... imagine a "normal" key or pbx system, which asks "please dial the extension now"
02:20.28[TK]D-Fenderjake1932 : A little, yes.
02:20.56jake1932how does it compare with the polycom?
02:21.01[TK]D-FenderTusker : Sure you can.
02:21.04jake1932or cisco?
02:21.42Tuskerwhat "feature" is this called?  everything I search on, such as extension, refers to asterisk internal extension :)
02:21.55[TK]D-Fenderjake1932 : Only advantage to Aastra is the backlight.  Polycom wins on configurability, LCD usability (backlight aside), and physical.audio quality
02:22.31jake1932can you get polycom support through normal channels without spending a fortune?
02:22.47jake1932with a small number of phones?
02:22.48[TK]D-Fenderjake1932 : never needed support :)
02:23.06jake1932i think they come with SIP loaded - right?
02:23.07[TK]D-Fenderjake1932 : you get it from your reseller typicall, and cost is just fine.
02:23.27wunderkin[TK]D-Fender = polycom support
02:23.51wunderkinand sales
02:23.57[TK]D-Fenderwunderkin : wel I AM planning on getting cert'd :)
02:24.07wunderkini know, what a surprise
02:24.23Tusker[TK]D-Fender: can you share your wisdom on the way to write the "dialplan" ?
02:24.28[TK]D-Fenderand NO, I don't sell them :)  I'm just promoting what I honestly feels is a WORTHY product, and people of repute agree.
02:24.38wunderkinpromoter, sorry
02:24.44[TK]D-FenderTusker : depends what you're trying to acheive.
02:25.47[TK]D-FenderPolycom IP 501 = $170, Aastra 480i = $200+.
02:26.48jake1932with SIP?
02:27.04[TK]D-Fenderjake1932 : Yes, both do.
02:27.11*** join/#asterisk MoutaPT (n=MoutaPT@85.139.183.56)
02:27.21Tuskerwell, it's a sip account... so, I'd like to put it as one of the outgoing connections in asterisk... so if I connect to my asterisk account, and dial say, 00 xxxx xxxx, so asterisk takes that, dials 55,,,xxxx xxxx
02:27.23[TK]D-Fenderjake1932 : They can also be flashed with MGCP (both brands)
02:27.23jake1932"SIP Software  - Certified VoIP Resellers can download from the Polycom Resource Center (PRC)."
02:27.31jake1932that line is a little scary
02:27.38MoutaPTHi does any one could advice me how to have incoming calls pop up with SugarCRM
02:27.48MoutaPTI'm planing an app for call center
02:27.53[TK]D-FenderTusker : no need for pauses in dialing sip #'s
02:28.21Tuskerthe sip service has a redirect call service
02:28.26[TK]D-FenderTusker : you woul dimsple prefix the # and route based on the prefix, filtering it off and adding any other digits as required.
02:28.43Tuskerany number I dial directly, before it answers, goes to the voice prompt
02:28.55Tuskeronce it answers, then I can dial the destination number
02:28.56[TK]D-FenderTusker : hmmm so you want it to use thier timed interface to do changes?
02:29.06jake1932like a LD pin code?
02:29.16[TK]D-FenderTusker : Sounds like a pain in the ass....
02:29.28[TK]D-Fenderjake1932 : Yeah, sounds like that kind of style...
02:29.41Tuskeryeah, it is a bit of a pain... they have their LD service running on the same server it seems
02:30.20jake1932you can't just get the extensions authorized - i wouldn't rely on sending tones afterwards
02:30.46jake1932only plan Z
02:30.46[TK]D-FenderI'm not sure how to do an out-bound macro like that... I know you can do pauses in dialing out an ANALOG interface, but your situation is outside of my experience.
02:31.17Tuskerok, I'll have a look at out-bound macro then, cheers!
02:35.22[TK]D-FenderTusker : Take a look at the m() parameter of Dial.  Might do what you're looking for if you time it right.  Use along with a command to send DTMF.
02:35.25MoutaPTHi does any one could advice me how to have incoming calls pop up with SugarCRM??
02:35.57[TK]D-FenderMoutaPT : First questio to ask yorself : how to identify the caller....
02:36.03*** join/#asterisk Spy000007 (n=Spy007@ool-44c045b0.dyn.optonline.net)
02:36.11[TK]D-Fendercan't type again.. sheesh
02:36.34Tuskerok, cool, cheers
02:36.38jake1932it's pseudo spanish
02:37.02Spy000007anyone use SOAP (XML) and asterisk?
02:37.31MoutaPTbased on RT request ticket id
02:37.34MoutaPTor caller id
02:38.21MoutaPTi may set caller id based on DTM from caller or simply use Request ticket id to forward to SugarCRM and pop up this call, don't know if it is easy...
02:38.33MoutaPTdTM=DTMF
02:39.16MoutaPTNo body has done this before?
02:39.23Spy000007no one?  everyone still using perl to hack together sip.conf?
02:39.38MoutaPTwhat is more usual for call centers? Agents need an app to handle calls...
02:39.39[TK]D-FenderMoutaPT : I'd say have them enter the identfying field with a Read, then have a PC call a URL based on it.
02:39.54MoutaPTlike YACC?
02:40.05[TK]D-FenderMoutaPT : Depends on what you can call to do something usefull.
02:40.16[TK]D-FenderDon;'t know Yacc offhand...
02:40.53MoutaPTYACC throws an URL with some parameters from asterisk call, like caller ID and UNIQUEID
02:41.07MoutaPTi was looking for a complete solution already
02:41.17MoutaPTeven if it is not opensource
02:42.11[TK]D-FenderMoutaPT : Sorry, such a thing should be Google-able
02:42.32Spy000007MoutaPT: doesn't fonality have some sort of crm integration?
02:42.37*** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net)
02:42.48MoutaPTi'm going to look for it
02:43.01MoutaPTi've been googling SIP CRM phone
02:43.06MoutaPTSip crm calls
02:43.11MoutaPTnothing usefull
02:44.35Spy000007MoutaPT: see www.fonality.com -- it mentions crm integration... but says salesforce.com, not sugarcrm
02:44.48MoutaPTthank you
02:44.49Spy000007MoutaPT: it's only an example though, so maybe it supports more than one
02:44.58MoutaPTi'm there already
02:45.00MoutaPT:)
02:45.44asterboygotta give Grandstream an A+ for customer tech support.
02:45.53asterboyBig fat F for Polycom
02:45.55Spy000007actually it seems like there's a guy who's on the board of directors for both sugarcrm and fonality, so i'm sure integration is on the way
02:46.10[TK]D-FenderMoutaPT : What kind of URL could you call with a ticket # or similar to pull up something sueful in a browser?
02:47.09[TK]D-Fenderasterboy : Grandstream greatly supporst a crappy product, Polycom crappily supports a great product.  And thus the Universe's balance is restored!
02:47.17asterboylol
02:47.22asterboythat is exactly it
02:47.24fileuniverse... balanced...
02:47.25filenever!
02:47.30[TK]D-Fenderasterboy : Whats your Polycom issue now?
02:47.50asterboynone, just that when I tried to get tech support, the buck was passed.
02:47.56[TK]D-Fenderfile : My karma ran over your dogma :D
02:47.58MoutaPTi may have a cgi runing with ticket number argument... so all the customer info is there...
02:48.09filemuahahahaha
02:48.12[TK]D-Fenderasterboy : Fine, but do you actually still have a problem right now?
02:48.15asterboytwo rings, 1 push of a button and Grandstream tech support HUMAN was online.
02:48.27asterboynot that I know of.
02:48.33[TK]D-Fenderfile : I've banked so many here you can't faze me :)
02:48.39asterboyThe grandstream upgrade fixed the polycom issue.
02:48.50asterboyerr...stopped screwing up the polycom
02:48.52filethis is true
02:49.03filenext time I'm in Montreal, breakfast/lunch/dinner/whatever's on me!
02:49.16[TK]D-Fenderasterboy : Sounds more like GS had a problem and its fixed until something ELSE breaks :)  More like your Polycom's REPORTED that your GS' suck :)
02:49.18asterboyMontreal is beautiful
02:49.28asterboylol
02:49.33asterboyyep that was it.
02:49.59asterboyIf this upgrade works though, the Grandstream will be a good phone.
02:50.11[TK]D-Fenderfile : just get yourself up here, we'll grab JunK-Y and go for a beer.  No debts between us!
02:50.15jake1932ERR: Cheapo Phone detected on network - Aborting!
02:50.27file:D
02:50.30asterboyMontreal is a great place for beer and food.
02:50.34[TK]D-Fenderasterboy : I preffer the term "less shit" where GS is concerned ;)
02:50.42asterboytrue
02:51.04asterboyI won't be selling them
02:51.49[TK]D-FenderImagine I payed for my own IP 301 & IP 501 for home.  Thats about $330 + tax CDN.  Thats the price of 3 GXP's, and I'm very glad to have payed more.
02:52.12asterboyyep, at home I have IP 300, 500 and 600
02:52.20asterboyno gxps
02:52.40fileI've got an IP600 at my desk... works well
02:52.43[TK]D-Fenderfile : Once I confirm that item we discussed as being constant between 1.2.4 & 1.2.7.1 think you could lend a hand in patching? :)
02:52.52asterboyespecially with the micro browser
02:52.59asterboyI get digg.com feeds and weather
02:53.04file[TK]D-Fender: yessir, I can give you something so you can figure out the flow
02:53.09fileand see what's going kaboom
02:53.12[TK]D-Fenderasterboy : Did I help you with those at one point?
02:53.36[TK]D-Fenderfile : cool.  will let you know once I've got proof :)
02:53.42fileyay proof
02:54.19Beirdo151 proof?
02:54.21[TK]D-Fenderfile : Well I know for sure about the intermittint one in 1.2.4, I'll jsut make sure its the same in current as well as the other mode.
02:54.22asterboyyep, pointed me towards the wiki to show what the semantics were
02:54.46asterboyI almost have a calendar ported over to the phone now.
02:54.56[TK]D-Fenderasterboy : picky-ass thing isn't it?  I HATE that it doesn't have tables....
02:55.20asterboyreally picky, and the wiki is horribly insufficient
02:55.34asterboyIt's like programming in old Pascal
02:55.48asterboyone wrongly placed space and boom
02:55.49[TK]D-Fenderasterboy : No... I LIKED Pascal...
02:56.03asterboyold Pascal was way too picky
02:56.07[TK]D-Fenderand Pascal was not white-space sensitive.  Fortran and Cobol often were though...
02:56.46asterboyone missing space before a semicolon and the whole program fwould not run and you got an error message which told you nothing of where the problem was
02:56.49[TK]D-FenderI was a TurboPascal GOD in my day (you know... when swooping pteradactyl's were the greatest threat to man...)
02:57.03asterboymine was
02:57.17[TK]D-Fenderasterboy : I did mine on an IMB 360 and in DOS... never had whitespace issues before...
02:57.57[TK]D-FenderIBM*
02:57.58asterboyI did mine on a TI
02:58.14[TK]D-FenderTI?  No accounting for their flakeyness :)
02:58.25asterboyheh
02:59.08asterboycobol should be wiped from history
02:59.27asterboyFortran was good, but not very fun
02:59.33[TK]D-FenderOne thing I'd really like to know : If only the 60x has the microbrowser, what is the "Services" button on my 501 for?
03:00.11asterboydidn't they get the microbrowser in the latest updates
03:01.43[TK]D-FenderDon't recall that....
03:01.59[TK]D-FenderI'll go DL the release notes...
03:02.18asterboyI had a C64 with a 300 baud modem to upload my fortran code.
03:02.27asterboyto the university computers
03:02.57asterboycan't remember if I was using a fortran emulator or terminal emulator
03:03.33[TK]D-Fenderasterboy : I've had worse :)  not as a modem... but I think I may have mentioned my earliesst 300 baud mode required you to hold a toggle switch to initiate carrier :)
03:03.38asterboyanyway, I remember sitting in my little room till 3am pushing buttons.
03:03.52asterboyoh ya...lol
03:04.29[TK]D-FenderYay, you can use DHCP to set the SIP server address now, cool...
03:09.33*** join/#asterisk cryptnix (n=andrew@64.25.198.126)
03:11.13[TK]D-Fenderasterboy : nope, nothing about the MB for 50x.  Tried adding it into the config, still no activity....
03:14.29Tusker[TK]D-Fender: is D(digits) in asterisk 1.0.7 ?
03:14.47[TK]D-FenderTusker : I believe so..... taht might do it...
03:15.38*** join/#asterisk websae_ (n=websae@CPE-24-167-206-22.wi.res.rr.com)
03:16.54websae_quiet friday night out there
03:20.37websae_jake: how are you doing?
03:20.48jake1932doing well - how bout you?
03:28.13*** join/#asterisk znoG (n=gs@109-130-89-200.fibertel.com.ar)
03:31.11websae_come one...come all
03:33.00*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
03:36.57websae_welcome back cunningpike
03:37.37CunningPikeThank you :D
03:37.49websae_you bet
03:38.16[TK]D-FenderWell.. I'm done... veg night concludes and back to thr grind tomorrow...
03:40.12[TK]D-Fenderater
03:45.05asterboydam that sucks...I thought the 501 could do that
03:45.19asterboyI know the 500 can not
03:45.39asterboyservice call for me.
03:45.49*** join/#asterisk shuri (n=shuri@64.235.209.226)
03:45.52asterboyI just got off work and now I have to go back
03:45.55asterboyyuk
03:49.04websae_why?
03:49.17asterboymoney
03:49.31asterboywhy else does anyone work?
03:49.35websae_ohh
03:49.35websae_ok
03:49.37websae_there you go
03:49.40websae_but are there issues?
03:49.55asterboynothing on the telco side.
03:50.15asterboybbl
03:50.23websae_bye
03:53.04*** join/#asterisk bmg505 (n=leon@dsl-146-51-61.telkomadsl.co.za)
03:53.10*** join/#asterisk pageus (n=FreePBX0@ip70-190-19-6.ph.ph.cox.net)
03:53.48*** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net)
03:56.22GamercjmAnyone know about VirtualHosts in apache, cant seem to get mine to work
03:56.38websae_#Apache
03:57.10CunningPike:)
03:57.24websae_cunningpike: still alive heh?
03:57.36CunningPikeOh yes
03:59.41*** join/#asterisk angom_h (n=angom@red-corp-200.79.134.173.telnor.net)
04:02.25dlynesAnyone play with MixMonitor?
04:02.37*** join/#asterisk jimmy_deanPB (n=jhodapp@cpe-24-166-23-17.indy.res.rr.com)
04:02.37dlynesErm ControlPlayback, I mean?
04:05.52*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
04:11.19dlynesActually, I guess the problem I'm having is with MixMonitor
04:11.25dlynesIt seems to record an empty gsm file
04:11.37dlynesOr empty raw file, or whatever format I'm recording
04:12.32dlynesI'm using MixMonitor(filename.gsm,a)
04:12.37tainted-why not just use Monitor
04:12.51dlynestainted-, because it doesn't do call leg mixing without using an external process
04:13.00tainted-soxmix
04:13.06tainted-what's wrong with that?
04:13.08dlynessoxmix is an external process
04:13.10Qwellwell, try using Monitor, and see if either/both are empty
04:13.18tainted-Qwell!
04:13.20QwellIf they aren't, you know it's likely to be a bug in mixmon
04:13.21tainted-stop trolling
04:13.33dlynesyeah...good idea
04:13.53tainted-Qwell u know much manager api?
04:13.59Qwellnot really
04:14.52*** part/#asterisk angom_h (n=angom@red-corp-200.79.134.173.telnor.net)
04:17.13*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
04:17.45jake1932<PROTECTED>
04:17.55jake1932we use it in a production system
04:18.08tainted-what platform do u use
04:18.13tainted-to connect to it
04:18.21jake1932we use a .NET app
04:18.25tainted-2.0?
04:18.57dlynesYeah...works fine with monitor
04:18.59jake1932.net 2003
04:19.16tainted-are u using the asterisk.net library?
04:19.21jake1932oh
04:19.30jake1932no - we wrote are own routines to parse
04:19.42tainted-oh nice
04:19.47jake1932it was pretty straightfoward
04:20.01tainted-yea i just got into
04:20.02tainted-it
04:21.01jake1932we're going to move over some agi code in a few weeks so everything is using manager
04:21.13tainted-wow
04:21.14jake1932should be cleaner
04:21.15tainted-same here
04:21.41tainted-agi does have good uses though
04:21.56tainted-do u write all your apps in 1.1?
04:22.51jake19321.1 .NET SDK?
04:22.59jake1932yes
04:23.03jake1932all current apps
04:23.37tainted-i wish there was a channel bridge app
04:24.05dlynesThere is
04:24.09dlynesIt's called the chunnel
04:24.37dlynesIt bridges England and France across the channel :)
04:24.45dlynesGood application of modern architecture :)
04:25.08tainted-groan
04:25.12dlyneschannel...bridge...app...
04:25.13tainted-someone spiked your bubble tea
04:25.17jake1932i think you can hack something with meetme
04:25.29tainted-meetme is a bucket of ass
04:25.45jake1932haha
04:25.56tainted-yea u can originate calls and drop into meetme() context
04:26.13jake1932right
04:26.16tainted-meetme(b) is broken
04:27.09jake1932don't think you need that though
04:27.33jake1932as long as you can still retain control of the call using manager
04:27.53*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
04:27.55tainted-is it possible to get meetme attendee list?
04:28.06tainted-from manager api or otherwise?
04:28.11tainted-for kicking/muting etc
04:28.17*** join/#asterisk gursikh (n=guriskh1@adsl-209-30-245-73.dsl.hstntx.swbell.net)
04:28.52jake1932i know you can access the CLI commands
04:29.00jake1932(from manager)
04:29.09jake1932looking if there is one
04:29.17tainted-there's meetmeadmin()
04:29.24tainted-and i think meetmecount()
04:29.28jake1932that gives you a count
04:29.39jake1932but i'm talking about cli cmds
04:29.42jake1932those are apps
04:29.48tainted-hmm
04:29.50tainted-good idea
04:29.56tainted-wonder if there is
04:30.08jake1932yep
04:30.11CunningPikedlynes: You've got mail
04:30.17jake1932http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe
04:30.21jake1932looks like you can
04:30.42dlynesCunningPike, thx
04:30.49CunningPikenp
04:31.08dlynesJust learning a new perl module atm
04:32.21tainted-hmm
04:32.26*** join/#asterisk Assid (n=assid@203.115.64.12)
04:32.37tainted-is it possible to set meetme to use 729 instead of ulaw?
04:33.08russellbmeetme can not mix g729 natively, no.
04:33.25russellbyou can call into it with g729, of course
04:33.33tainted-and it transcodes?
04:33.37russellbyes
04:33.49russellbif you have codec_g729
04:33.52tainted-no wonder
04:34.53tainted-i don't have 'Meetme' in CLI
04:35.27jake1932do you have the app though?
04:35.45tainted-yea
04:35.48tainted-oops
04:35.49tainted-wrong box
04:35.51tainted-lol
04:36.32pageusheya tainted
04:36.52tainted-hey didn't recognize u w/o the capital P
04:37.01pageusROFL
04:37.04Pageuslol
04:37.13tainted-how's it going
04:37.42Pageuswas just playing some metroid prime hunter with my son before i came in and worked on this.. turns out i won't get my cable drop till monday at 8am.. but i did get the specs on the line
04:37.49tainted-is voicepulse still acting weird?
04:38.07tainted-what kind of cable?
04:38.24*** join/#asterisk fjean (n=fjean@201.29.130.118)
04:38.42fjeanhi guys
04:38.46fjeantell me
04:38.47Pageusnope.. that apparently was a misconfiguration on my end.. since i had hunt on and that extension was offline it went to vm
04:38.57Pageusheya dlayn
04:39.38fjeanis there anything special to be done in order to make zaptel 1.2.5 ?  I get  *** No rule to make target `modules'
04:39.41Pageusit's a T1.. esf coding, b8zs framing, 6 2 way DID trunks. the start signaling is E&M Wink signal type is dtmf
04:39.51russellbfjean: just "make ; make install
04:39.53Pageustg direction (??) is 2 way
04:40.05fjeanrusselb, let me try
04:40.19tainted-Pageus how much is that running u per month?
04:40.40fjeanrusselb - nope...
04:41.08Pageus220
04:41.11tainted-fjean u don't need to 'make modules'
04:41.11Pageusplus ld
04:41.20Pageusi have a total of 20 DID's
04:41.26russellbfjean: then you need the kernel headers installed
04:41.53Pageusthis looks like it's all the info i need to configure that line though, right?
04:42.06fjeanrusselb - ok, I installed  kernel-source already, maybe it's missing a link
04:42.46russellbfjean: you need /usr/src/linux-`uname -r`
04:42.54fjeanah
04:43.02russellbor linux-headers-`uname -r`
04:43.03tainted-Pageus u'll need some hardware
04:43.04russellbsomething like that :)
04:43.12fjeanhehe, let me see
04:43.35tainted-6 channels for 220, that's pricey
04:43.49tainted-are the other channels data?
04:44.45russellbfjean: which should also give you /lib/modules/`uname -r`/build/
04:45.27Pageusi have the hardware..
04:45.35Pageusthat isn't an issue
04:45.52Pageusand the 220 is all business line..
04:46.04Pageuswe were paying 350 a month
04:46.07fjeanrusselb - your good, i think I installed the kernel sources for 2.4 instead of 2,6
04:46.24russellbfjean: ;)
04:46.32Pageusbut since it never leaves the facility, we have no gov taxes
04:52.38fjeanrusselb - way to go, working
04:52.49Pageusthe good part is that it's all local calling.. so i don't have to pay the ld fee for outbound unless is really is ld
04:53.10russellbfjean: awesome
04:55.53fjeanby the way, I am no expert at linux but one thing I know is that it's difficult to get the modprobes to be performed at boot time..I thouht make install would do that part..
04:56.08russellbfjean: it's specific to the distribution of linux ...
04:56.17fjeanright..
04:56.28russellbfjean: there are init scripts you can use for most
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04:58.34fjeanrusselb: any script  included in zaptel ?
04:58.54russellbyeah, what distro are you using
04:59.04fjeanmandrake 10.1  :-)
04:59.50CunningPikeDoes mandrake use /etc/init.d?
05:00.02fjeanyes..
05:00.02russellbi have no idea if the included one will work
05:00.23russellbit looks like it's set up to work for debian or redhat.
05:00.33CunningPikeIf it does, the one that gets installed doesn't have paths to modprobe (e.g. /sbin/modprobe), so you need to modify it to make sure
05:00.42fjeanwell i know make install would not work, if that is what its supposed to do..
05:00.55fjeancunning: ok
05:01.00russellbno, make install does not install it
05:01.04russellb"make config" installs the init script
05:01.10fjeanah, right
05:01.14fjeanI forgot
05:01.17fjeanok
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06:42.51tparcinagood morining
06:43.03websae_good morning
06:43.07websae_how are you?
06:44.54tparcinasleapy :)
06:45.11tparcinait's quite...
06:45.40websae_it sure is
06:45.44websae_no activity for hours
06:46.18tparcinaand you are from australia, or are you doing night shift? :))
06:46.31websae_from United States
06:46.37websae_Milwaukee, WI
06:46.38websae_:)
06:47.17tparcinado, what's the time over there?
06:47.24websae_where are you from?
06:47.29websae_it's 1:45AM
06:47.48tparcinacroatia (hr)
06:48.06websae_what do you use asterisk for?
06:48.10*** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com)
06:48.26tparcinaI work in company that makes pbx's
06:48.48tparcinawe curently use one, and we have sell one call centar
06:48.59websae_do you use sip or pri?
06:49.00tparcinanow, we re selling another two...
06:49.05websae_very interesting
06:49.09websae_people terminating over SIP?
06:49.15websae_or PRIs?
06:49.23tparcinaright now pri, and h323
06:49.29websae_sure
06:49.50tparcinabut those two new * will be conected together, probably over sip
06:50.11tparcinau work with *?
06:50.49websae_yes
06:50.51websae_quite a bit
06:51.06websae_here private msg me
06:51.14tparcinahawe you used cisco phones?
06:51.17*** join/#asterisk Kernel_Core (n=I@193.251.135.118)
06:51.39websae_a couple
06:51.41websae_times
06:52.12tparcinahere private msg me - don't get it
06:52.18tparcinawhat phones?
06:52.24tparcinasip or sccp?
06:52.32websae_sip
06:52.35websae_7960
06:52.46*** join/#asterisk MGSsancho (n=user@adsl-67-125-156-130.dsl.irvnca.pacbell.net)
06:52.48websae_do you have msn or yahoo?
06:56.20*** join/#asterisk nigelr (n=nigelr@ninja.nobiscuit.com)
06:57.04nigelranyone got time for a curly question on tranfer capabilities for a PRA?
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06:59.27CunningPikeSorry, websae_ - didn't see your PM until now
06:59.46CunningPikeStupid Colloquy always pops the PM window under the main one
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07:08.32tparcinacisco, sip, hinting (multiple line appereanses) - has anybody done it?
07:08.52tparcinai have check on wiki, but I couldn't find anything about hinting on sip
07:18.47*** part/#asterisk startled (i=startled@d58-105-31-172.dsl.vic.optusnet.com.au)
07:23.33dlynestparcina, I'm working on a page for that, tparcina
07:23.39dlynestparcina, it's using aastra phones
07:23.55tparcinahi dlaynes, how are you today?
07:24.46dlynesgood
07:24.48tparcinadon't get it. what aastra phones have to do with cisco 7960 sip hinging?
07:24.56tainted-dlynes finish the billing system yet? lol
07:24.59dlynesaastra's do sip hinting, too
07:25.09websae_oooo---dirty
07:25.18tainted-websae_ what games do u have?
07:25.37tainted-we could play halo 2 and stuff
07:25.51websae_they cancelled my xbox live aaccount
07:26.29tainted-that's terrible
07:27.02websae_im depressed
07:27.11tparcinadlynes, have you done anything so far? (cisco hinting)
07:27.23dlynesAastra hinting, yes
07:28.41tparcinaand you are sure that hinting is posible on 7960 with sip firmware?
07:28.52dlynesNope
07:28.57tparcinado you have link to any instructions (even partial)
07:29.03dlynesI don't know what the 7960 supports
07:29.07dlynesOne sec
07:32.59dlynestparcina, http://www.voip-info.org/wiki/view/480i+Busy+lamp+field+'BLF'+support
07:33.03{zombie}I didn't think the 7960 supported hinting even via SCCP - you need the 7961 for that
07:34.19{zombie}or the 7914 expansion panel for the 7960
07:34.49dlynestparcina, btw...you did say "but I couldn't find anything about hinting on sip"
07:35.06dlynestparcina, you didn't say you were looking for hinting on sip, specifically for the cisco 7960
07:35.28{zombie}dlynes: he did in the line above
07:35.35{zombie}cisco, sip, hinting
07:35.50dlynesYeah, but not originally, which is what i had replied to
07:36.56*** join/#asterisk poisoner (i=poisoner@tauceti.poisoner.de)
07:37.02tparcinaok, sorry if I have mislead you :)
07:37.07dlynesyeah...hinting isn't supported on the 7960, tparcina
07:37.42tparcinazombie, have you set up hinting on 7970 with sip?
07:37.54dlynesdoesn't look like 7961 supports it either...you need 7971, and maybe 7970
07:38.01websae_does anyone know tasker here?
07:38.08websae_I am looking for tasker
07:38.26{zombie}tparcina: no I only have 7960, 7940 and 7910 cisco phones
07:38.32tparcinadlynes, i though so, but evry while, sombody mentions it, so i though... anyway, too bad
07:38.37{zombie}although I have set up hinting with grandstream gxp2000 and all snom phones
07:39.03tparcinadlynes, i have 7970, but i can't find instructions
07:39.17dlynestparcina, that's what I said...I don't think the 7970 supports it
07:39.22dlynesI think you need the 7971
07:39.30tparcinai hate cisco phones, they have 100 books for ccme but nothing for sip general...
07:39.44dlynesvoip-info isn't clear on that, but they seem to indicate the 7970 doesn't do hinting
07:39.52dlynesthe 7971 does hinting for sure
07:40.37{zombie}7961 does too, via SCCP at least
07:40.49*** join/#asterisk xMOe (n=Blade@62.149.93.73)
07:40.50{zombie}http://www.voip-info.org/wiki-chan_sccp2
07:40.54xMOehola guys, i've call center system based on Asterisk with  Digium TDM400P but calle id dosnt work out fine can anyone tel if its support caller ID  in Saudi Arabia or not , and if yes any online reference for  zapata.conf configuration settings are needed
07:43.35tparcinaxmoe, check ith your telco does it support caller id
07:44.03dlynesxMOe, your telco and/or your line might not support it
07:44.27dlynesit's not a country-specific thing
07:44.50dlynesi can buy analog lines here with caller id, and without caller id
07:45.18dlynesyou generally pay extra for a line if it has caller id
07:45.46dlynestainted-, do you ever go to sleep?
07:46.12tainted-no
07:46.18dlynesthought not
07:46.23tainted-it's gotten so bad, i dream in code
07:46.59dlynes0x22 0x43 0x44 0x45 0x22
07:47.24tainted-not that bad
07:47.28dlyneslol
07:47.39tainted-sometimes i'll solve real life problems in code
07:47.42dlynesor better yet
07:47.59dlynesA9 22 43 44 45 22 A9 C9 D8
07:48.09dlynesRun that, and tell me what you get
07:48.15dlynesIt's 6502 opcode :)
07:48.22tainted-if (takeGarbageOutNow() { ... smellFactor += 1; ) // hmmm
07:48.35tainted-oops missing )
07:48.43tainted-damn it now i'll never get to sleep
07:48.45dlynesIt prints out the string, 'ABC'
07:48.48dlynesheh
07:49.08*** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it)
07:49.10tainted-yea you're pretty cool
07:49.17tainted-010011001011110100101
07:50.49dlynesIs the search feature on voip-info broken?
07:51.37xMOetparcina its support 100%
07:52.01xMOedlynes yes its do and i used to get called id with asterisk and sipura
07:52.11dlynesah
07:52.29xMOebut now with TDM400P i just got " Astrerisk"
07:52.51dlynesYeah...that's what you get when it can't get a caller id
07:53.03dlynesIs this TDM400P a Digium TDM400P, or is it a clone?
07:53.39xMOe<PROTECTED>
07:53.47dlynesone second
07:54.14xMOeok
07:56.02L|NUXcan some one tell me how can i goto another context once i got timeout ?
07:56.32dlynesxMOe, do you get an error like this: messages.7:Apr 27 00:07:33 WARNING[1569] chan_zap.c: CallerID returned with error on channel 'Zap/1-1'?
07:56.40Pageusi hear you on that one tainted.. dreaming in code has written many a program for me
07:58.40xMOedlynes i got no error msg :)
07:59.11websae_anyone know tasker around here?
07:59.18dlynesL|NUX, Look at the following page:  http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
07:59.34dlynesL|NUX, also, check the documentation for the Dial() application
07:59.47dlynesxMOe, wow...no idea then
08:00.05dlynesxMOe, do you have warning level messages disabled?
08:03.32*** join/#asterisk Pageus (n=FreePBX1@ip70-190-19-6.ph.ph.cox.net)
08:03.47xMOedlynes nope..
08:04.11dlynesxMOe, so you double checked logger.conf?
08:06.19*** join/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com)
08:08.18xMOeyes i ddd
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08:31.52*** join/#asterisk frenzy (n=frenzy@196.45.144.41)
08:32.02frenzyhey all
08:32.27frenzyis there an advanced carrier routing tool available for asterisk ?
08:35.01*** join/#asterisk [hC] (n=hardcore@S01060004e21ea953.vc.shawcable.net)
08:36.07*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
08:38.12frenzy?
08:40.49dlynesfrenzy, i think you're looking for a carrier grade linux
08:41.00dlynes?
08:41.26frenzya carrier routing based on mysql
08:41.38frenzybascially to do LCR failover
08:42.05dlynesOh...you mean a dialplan application?
08:42.25dlynesI think there might be something out there already for that
08:42.28frenzysomething like that
08:42.37dlynesbut if you mean a separate tool that runs from the command line then probably not
08:43.08frenzyno
08:43.26frenzywhat kind of applications are availble to do that?
08:43.32dlyneshttp://www.voip-info.org/wiki/view/LCR+tool+for+i4l
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09:02.05boddyhii all local phone that on Meridian 1C when call the sip client which codec it will use ? asterisk and meridian connected via E1
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09:02.43*** part/#asterisk jomo2005 (n=jomo2005@c-24-98-66-60.hsd1.ga.comcast.net)
09:03.14boddy?
09:10.50*** join/#asterisk treobruce (n=jomo2005@c-24-98-66-60.hsd1.ga.comcast.net)
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09:13.05*** part/#asterisk treobruce (n=jomo2005@c-24-98-66-60.hsd1.ga.comcast.net)
09:13.55boddyanbody help me?
09:13.55boddyzzzzz
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09:30.59dlynesboddy: probably whatever its preferred codec is
09:31.41*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
09:33.21*** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it)
09:34.04*** join/#asterisk jeffik (n=Jeff@Crimson-111.085.ADSL.NetSurf.Net)
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09:39.21boddydlynes I didnt understand
09:39.32*** join/#asterisk frk2 (n=kvirc@202.141.251.102)
09:39.44frk2is there any such thing as a cheap IP phone that DOESNT hang?
09:39.53dlynesyour sip client has a preferred codec; the codec it prefers to use over other codecs
09:40.02dlynesthat's the codec it'll probably use
09:40.18frk2i dont understand. I have tested all IP phones and almost ALL of them hang.. the low end ones
09:40.29dlynesfrk2: I haven't had any hanging issues iwth the budgetone 102's
09:41.02frk2dlynes
09:41.08frk2i had high hopes on those two
09:41.13frk2but one hung today only :(
09:41.17dlynesah
09:41.22frk2i upgraded the firmware to 1.0.8.16
09:41.24dlynesYeah...don't go for the ACT phones
09:41.29frk2i hope that didnt screw it up
09:41.29dlynesThey hang regularly
09:41.36dlynesthey suck total dog doo
09:41.40frk2ACT?
09:41.48dlynesAdvantage Century Telecom
09:41.54boddydlynes:so if my sip clients use g729 all conversation is g729
09:41.59boddy?
09:42.04dlynesalso known as GVC, Azatel, ...
09:42.05frk2arent they the pa1688 guys?
09:42.11*** join/#asterisk bmg505 (n=leon@dsl-146-51-61.telkomadsl.co.za)
09:42.12dlynesfrk2: correct
09:42.21frk2dude. funny enough
09:42.24dlynesfrk2: i've never used that particular product though
09:42.27frk2the pa1688 phones hang the LEAST
09:42.32frk2but they have other issues
09:42.34jeffikdlynes: Is there a way to hard reset an ACT phone?
09:42.51frk2dlynes- what firmware on those non hanging 102s?
09:42.59dlynesjeffik: 255*0*0*0, assuming you've got a hardware revision that actually works on
09:43.44dlynesfrk2: 1.0.1.0
09:43.51jeffikdlynes: well mine worked, i loaned it to a client and now that i have it back when i plug in power all lights come on and display shows blocks
09:43.51frk2thats the beta
09:44.02dlynesif it works, it works
09:44.08dlynesI've never upgraded the firmware on it
09:44.10frk2dlynes-- dont you also have the GXP 2000s?
09:44.14dlynesnope
09:44.38dlynesGrandstream BT-102, Azatel IPCall104, Aastra 9133i
09:45.03dlynesAnd a bunch of polycom 500's that will only work with Artisoft Televantage
09:45.09markitgxv-3000 amatorial video from CeBIT 2006: http://video.google.com/videoplay?docid=6420459719167336518&pl=true
09:45.28*** part/#asterisk liran_ (n=liran@80.178.14.98.adsl.012.net.il)
09:45.42dlynesjeffik: i would guess the firmware is fried on it
09:46.06dlynesjeffik: if the display is all scrambled i doubt the keypad reset sequence will work
09:46.47jeffikdlynes: just tried your suggestion 255*0*0*0 and nothing
09:47.02boddydlynes e1 uses ulaw/alaw ?
09:47.42dlynesjeffik: like i said...i didn't think it would work
09:47.56dlynesboddy: no...e1 uses telephone event
09:48.03dlynesboddy: it doesn't use a codec
09:48.15jeffikdlynes: thanks for the suggestion anyway, it was worth a try
09:48.31boddyI confused :(
09:48.36dlynesjeffik: yeah, but at least you know the reset sequence if you have other phones you've lost the password for
09:49.03dlynesjeffik: it works for firmware 2.0.8 and higher
09:49.19dlynesjeffik: it might work for 2.0.7, but definitely not 2.0.6
09:50.20dlynesI'm going to be calling those bastards on Monday to see if I can get some support out of them
09:50.25dlynesAzatel's gone out of business
09:50.27*** join/#asterisk mutante (i=mutante@s23.org)
09:50.56boddydlynes:Please explain to me when local phone calls to sip client will it use any codec ?
09:51.12dlynesboddy: local doesn't use a codec, sip client does
09:51.39frk2dlynes i think the firmware upgrade did it
09:51.42dlynesboddy: asterisk will by default try to use the least processor intensive codec, unless the client overrides it
09:51.46*** join/#asterisk treobruce_ (n=jomo2005@c-24-98-66-60.hsd1.ga.comcast.net)
09:52.07dlynesboddy: so, if your client supports g729, ulaw, alaw, ...
09:52.18stoffell_hso if u use u/alaw on your sip client, the * server does not have to do transcoding..
09:52.21dlynesboddy: and ulaw is your default, it'll use ulaw
09:52.45Assiderr.. anyone have a poly501 ?
09:53.11Assiderr.. 301
09:53.14frk2i wonder if i can downgrade the GS-102
09:53.15stoffell_hAssid, almost everyone has 1 ;)
09:53.28treobruce_hey guys... i'm trying to setup asterisk on debian
09:53.31*** join/#asterisk apardo (n=apardo@87.217.145.245)
09:53.31Assidis it any good?
09:53.33treobruce_can anyone help
09:53.44Assidwhats thedifference between 301 and 501 majorly
09:53.48stoffell_hAssid, i have 501, it's great. (601 has microbrowser)
09:53.58boddyI am wondering this local phones dosent use codec but how session is compatible with g729 asterisk do this ?
09:54.16Assidi was supposed to get a 501.. they sent me a 3013
09:54.20Assid301 rather
09:54.23stoffell_hI believe the 301 has no speaker phone, 501 does, and the sound of the handsfree is great
09:54.34dlynesand the 301 is half duplex speaker phone
09:54.41stoffell_haah, listen only..
09:54.55Assidso cant speak and talk at the same time
09:55.02dlynesAssid: correct...only listen
09:55.03Assidanythuing else?
09:55.35treobruce_any willing helper here to help a newbie with asterisk?
09:55.42treobruce_msg me if u are willing :)
09:55.43treobruce_thanks
09:56.03dlynestreobruce_: just ask your question in the channel
09:56.27treobruce_oh
09:56.36treobruce_sorry, new to freenode too :)
09:56.48treobruce_well i'm trying to learn asterisk
09:56.56treobruce_so i installed debian on my system
09:56.56boddydlynes:do you understand my question
09:57.06Assidplease tell me it atleast supports ftp
09:57.23dlynesboddy: yes, and so did stoffel_h
09:57.23boddyI am trying to learn logic
09:57.30dlynesboddy: we both answered you already
09:57.42iDunnoerm - logic shouldn't need to be learnt. logic should just happen. it's logical!
09:57.44stoffell_hAssid, I think the 301 also supports central provisioning through ftp (but check voip-info / polycom 2 be sure)
09:57.57treobruce_how do i install asterisk on debian
09:58.05stoffell_htreobruce_, you needs docs, i will give them, just a sec..
09:58.08stoffell_h~docs
09:58.11jbotit has been said that docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
09:58.26Assidstoffell_h: whats better? polycom301 or linksys pap2 ?
09:58.36treobruce_wow that's alot
09:58.47stoffell_hhappy reading ;)
09:58.52*** join/#asterisk kore (i=kore@mindwipe.org)
09:58.57boddyIf I use g729 on asterisk then asterisk will transcoding
09:58.58stoffell_hAssid, don't know that linksys..
09:59.11dlynesAssid: You're comparing a phone to an ata
09:59.16dlynesAssid: that does not compute
09:59.26Assidfeature vs/ quality
09:59.34dlynesAssid: no comparison
09:59.41dlynesAssid: polycom wins hands down by a long shot
09:59.50Assidhrmm
09:59.51Assidokay
10:00.14dlynesAssid: the pap2 has more features, but who cares?  all those features are available in asterisk, too
10:00.43dlynesAssid: a polycom is much better than using an analog phone hooked up to a pap2
10:00.44boddydlynes:
10:01.17AssidPAP2  has features?
10:01.20Assidman i didnt know
10:01.33frk2so anybody using the GS - 102 with the newer firmware?
10:01.35stoffell_hhm, Assid, you don't want that linksys thingie, you want that polycom ;)
10:01.38frk21.0.8.16?
10:01.55frk2damnit.. i shouldn't have upgraded!
10:02.00dlynesstoffell_h: the pap2 is the new two port fxs ata from linksys, it's a drop-in replacement for the sipura 2002, and the sipura 2000
10:02.28dlynesstoffell_h: it's the crap that vonage is giving away for free
10:02.30boddydlynes: If I use g729 on asterisk then asterisk will transcoding
10:02.37dlynesboddy: correct
10:02.38boddy?
10:02.38stoffell_hdlynes, thanks.. yeah, we all want native sip phones, not rubbish :)
10:02.41boddyok
10:02.45boddythanks alot
10:02.45dlynesboddy: and your performance will suffer
10:02.47Assidactually i got both next to me
10:02.59Assidi have to decide what i want
10:03.14boddyI will use good machine has p4 cpu
10:03.18dlynesAssid: the sipura 2000 aka linksys pap2 has a whole bunch of vertical service codes predefined
10:03.34frk2dlynes - is there a way to downgrade the GS-102?
10:03.36dlynesother than that, and the ability to do faxing
10:03.46dlynesfrk2: maybe install older firmware? :)
10:03.54stoffell_hboddy, all depends on the amount of calls
10:03.54frk2no on
10:04.04frk2thats supposed to be impossible (according to grandstream)
10:04.16dlynesthat would seem kinda silly
10:04.21boddystoffell_h yes you right 10-15 calls is simultane
10:04.23dlynestry it and see what happens :)
10:04.30dlynesit's only a $60 phone, anyways
10:04.49boddyin same time
10:05.31dlynesAssid: but other than, unless you really need to do faxing, feel like pulling out all your hair, and then get frustrated when faxing is heavily unreliable
10:05.41dlynesAssid: i wouldn't suggest the pap2
10:05.48boddyok thanks all
10:15.40*** join/#asterisk Assid (n=assid@203.115.64.12)
10:16.56frk2so i guess the best low cost phone is the GS-102
10:18.09stoffell_hfrk2, the "lowest" I go is the thomson st2030
10:18.28frk2The lowest I went was the atcom 323 - pa1688 based
10:18.35frk2thomson?
10:18.38frk2where do you get that from?
10:19.00stoffell_hfrk2: http://www.voip-info.org/wiki/view/Thomson+ST2030
10:19.09stoffell_hthey are starting to get pretty popular in europe
10:19.45frk2checking them out
10:19.48frk2how much for a phone?
10:20.21stoffell_hfrk2, check the e-commerce sites on the bottom, but we do them for 125 EUR without vat
10:20.59frk2nice looking phone
10:21.02frk2damn dude
10:21.05frk2thats not 'low end'
10:21.06frk2:)
10:21.24frk2not for a third world country im in
10:21.35stoffell_hwell, i've been using gxp-2000, it's cheaper, but sound quality is not so good
10:21.46frk2gxp 2000 hangs man
10:21.51frk2with too many simultaneous calls
10:22.00stoffell_hit's buggy yes :)
10:22.07frk2put it as the operator phone at one of my clients
10:22.08stoffell_hbut latest firmware makes it a bit more stable
10:22.10frk2had to remove it
10:22.34frk2see if a IP phone HANGS.. its total loss
10:22.41frk2thats the worst thing that can happen to a IP phone
10:22.57stoffell_hthere's also a thomson 2020, it's cheaper, but doesn't have central provisioning
10:23.02stoffell_hyeah, it is :)
10:23.37Assidstoffell_h: do you have a 301 with you at the moment?
10:24.08stoffell_hno Assid, only have used 501
10:25.35*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
10:26.06frk2for a third world country, i need a cHEAPP ass phone with decent voice quality and one that DOES NOT hang
10:26.16frk2cheap meaning in the $40-$60 range
10:26.17Assidfrk: ata
10:26.49frk2even ATAs are more expensive man
10:27.28Dr-Linuxwhat is third world country? :S
10:28.12Assidis it me.. or polycom site slow?
10:29.13Assiderr.. anyone remember that site.. whihc had the polycom resources
10:29.25Assida third party site
10:30.53Assidgot it .. freedomphones
10:33.53Assidhow do i come to know the sip version?
10:33.54*** join/#asterisk _Vile (n=vile@90.b160.bendtel.net)
10:34.39Assidand what version can i upgrade a 301 to?
10:36.39Assidstoffell_h: you there?
10:37.02_Vileask your question again
10:37.25Assidwhat version of sip /bootrom can i upgrade to with 301 ?
10:37.41_Vile301?
10:37.54frk2anybody using pa1688 based phones?
10:37.58Assidpolycom 301
10:38.04_Vilethought so
10:38.19_VileI can't answer that, I play cisco
10:38.53frk2but anybody? ive been using atcom's for a while
10:38.54_Vilebut gimme a sec to search google
10:40.40*** join/#asterisk saftsack (n=saftsack@p54A7F649.dip.t-dialin.net)
10:41.13_Vile3.1
10:41.18Assidi guess i can use 2.6.2 and 1.6.x
10:41.25Assiderr. 3.1 they suggested not to use
10:41.31Assidyou cant downgrade again
10:41.40_Vilesec
10:42.00_Vile2.6.1
10:42.04_Vilego with that
10:42.07_Vileit seems clean
10:42.21Assidyeah thats what i have
10:42.34Assidbut sip is old
10:42.34Assid0104173809|so   |*|00|Application, main: Label=SIP, Version=1.4.1.0040 14-Dec-04 11:49
10:42.37Assid1.4.x
10:42.57_Vile*shrug*, I can't suggest more than is documented
10:43.17Assidany idea how 1.6.2 is?
10:43.42_Vilenone, do you have any problems?
10:44.02Assidhavent got it up
10:44.07Assidbut i need the files anyways.. to provision
10:44.25Assidmy 501's are running on 1.6.2
10:45.19*** part/#asterisk Pageus (n=FreePBX1@ip70-190-19-6.ph.ph.cox.net)
10:45.40*** join/#asterisk zotz (n=zotz@24.231.32.85)
10:45.42_Vile2.6.1 is apparently the newest, bug free thing you're going to find
10:46.18stoffell_hAssid, on my 501's i'm running the latest
10:47.01_Vilestoff, what version is the latest?
10:47.11stoffell_h1.6.5
10:47.24stoffell_hbootrom can be 2.6.x or 3.x
10:48.06Assidany difference between 1.6.5 and 1.6.2?
10:48.25_Vileyep
10:48.50Assidworth going in fro 1.6.5 ?
10:48.50_VileI can't help here, I don't have a phone and am going from undocumentation
10:49.01_Vilestoff has the right info
10:49.07_Vileverified
10:49.11stoffell_hAssid, difference, yes, you can read it in the zip file (of 1.6.5) I hope :)
10:49.36stoffell_hmy poly's were 1.6.2 also, but I wanted to have the latest to be sure of stability.. :)
10:49.50Assidjust wondering if its worth upgrading to 1.6.5
10:49.52stoffell_h_Vile you using cisco's ?
10:49.56Assidsince its a new phone already
10:50.04_Vile7940s 7960s yes
10:50.11stoffell_hAssid, if you don't have a reason, skip the upgrade.. you can always do it later..
10:50.21Assiddont have a reason NOT TO
10:50.36_Vileyes you do
10:50.43_Viledo you have bugs?
10:50.44stoffell_h_Vile, it's correct that the latest 7960 firmware is free downloadable? (I 've downloaded it a few days ago)
10:50.48_Vilethat's a reason not to
10:50.57_Vileif your answer is no
10:51.13_Vilestoff, I dunno, I have a cisco pass :)
10:51.13stoffell_hAssid, like _Vile says, if it's not broke, don't try to fix it ;)
10:51.43Assidnever tried this phone.. dont know if its buggy or anything.. i remember the 501 being buggy till i upgraded it to 1.6.2
10:51.52_Vilewell
10:51.58_Vilefirst step > try it
10:52.10_Vilesecond step > ask
10:52.24_VileI take that back
10:52.29_Vilesecond step > research
10:52.33_Vilethird step > ask
10:53.12stoffell_h:)
10:53.26_Vileis the 7960 fw releases?
10:53.29_Vilereleased?
10:53.52_Vilemaybe cisco is pushing for hardware now, I need to look
10:54.13_Vilethey're big on hardware, software &&& support.. hm
10:54.43stoffell_h_Vile, i thought so (P003-8-2 or something)
10:55.41_Vile<PROTECTED>
10:55.48_Vilemy phones are currently running it
10:55.51_Vilewas internal
10:56.06stoffell_hyeah, that's the one.. I downloaded it last week (without having cisco pass)
10:56.08_Vilehmm, last month anyway
10:56.25_Vileinteresting
10:56.28stoffell_hmaybe it's released then..
10:57.12_Vilecould be a good way for them to push their hardware
10:57.29_Vileand *hah* support contracts
10:57.58stoffell_hhehe, yeah, would make life easier and phones (even) more widespread
10:58.28*** join/#asterisk hads|home (n=hads@203.109.245.87)
10:59.36_VileI wonder if that's a marketing thing, those downloads are usually restricted... someone slashdot it, Cisco, releasing upgrades for their IP phones
10:59.45_Vile*shrug*
10:59.50_VileI'm going to bed
11:00.21stoffell_hindeed... cu l8er ;)
11:06.26asterboybed?
11:06.46asterboyplenty of time to sleep when your DEAD!
11:07.14asterboyyou know....shovles full of dirt being dumped on top of your body.
11:07.26luke-jr_Anyone else use SellVoIP?
11:07.56asterboynope, just BuyVoIP
11:08.16asterboynight
11:12.16*** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au)
11:13.34Assidi think i screwed something up
11:13.42Assidits stuck at Checking application
11:13.44Assid:(
11:14.55Assidftp shows its trying to transfer my <mac>.cfg file
11:15.02stoffell_hdoes the .cfg file exist?
11:15.06Assidyep
11:15.13stoffell_hit could be in the wrong format then..
11:15.25Assidwrong format?
11:15.27stoffell_hjust wait :) you won't brake it (unless you power off during provisioning)
11:15.50Assid401 /home/p301/0004f20265fb.cfg b _ o r p301 ftp 0 * c
11:15.54stoffell_hwell, any typo in the file could cause errors..
11:16.19stoffell_hin that file you define where to get the 'other' .cfg files, do they exist?
11:16.56Assid<APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="0004f20265fb-phone.cfg, 0004f20265fb-sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="log/" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY=""/>
11:17.02*** join/#asterisk coppice (n=chatzill@153.192.17.210.dyn.pacific.net.hk)
11:17.03luke-jr_EWW
11:17.31stoffell_hdo the files exist? (sip.ld, blabla.cfg, etc..)
11:17.36Assidyes
11:18.02Assidokay how do i stop it and make it start up again? kill the ftp server?
11:18.25stoffell_hwell, if the ftp server says it's doing nothing, then you can do that, or restart the phone
11:18.34Assidpower on/off ?
11:18.48stoffell_hyep
11:18.58stoffell_hunless it's reading the sip.ld file ;)
11:19.23Assidwell.. as long as the bootrom is not being updated.. dont see why it could die
11:19.40Assidjust curious.. whats the buttons for hardware rebooting
11:19.45Assidlike you press a few keys right?
11:20.06stoffell_hhm, i only know of the reboot-through-menu (wich you can't right now), don't know any other
11:21.54Assiddamn... sip.ld is soo freaking big
11:22.53Assiddoes it always download the file every reboot etc?
11:23.15stoffell_hno, only if it's changed
11:24.07Assidstill stuck at checking application on the phone :(
11:24.54stoffell_hI guess you better reboot it
11:25.07Assidi did
11:25.10Assid2nd time its stuck
11:26.14stoffell_hdid you follow the procedure on uhm, voip-info? (there's a big article on poly provisioning)
11:26.42Assiderr.. ive provisioned over 10 phones in 501
11:26.47Assidfirst time im doing a 301
11:26.55stoffell_haaaaah, ok:)
11:26.56Assidbut they were all connected to lan
11:27.04Assidim doing it on a slowwwwwwww broadband
11:27.50Assidsip.ld is like 11MB
11:27.59Assidits gonna takes a while to download
11:28.33Assidif i ftp the file locally .. and have it connect.. and then change the ftp back.. will it download sip.ld from that remote ftp again ?
11:29.19*** join/#asterisk Corydon76-home (i=three@pdpc/supporter/sustaining/Corydon76-home)
11:29.28stoffell_hAssid, good question. If the date/time on the file remains the same, I guess it won't
11:30.34Assidhrmm.. maybe i should get snmp or something to check if my router is rx/tx any packets
11:30.36Assidand their speed
11:31.38*** join/#asterisk Corydon76-home (i=purple@pdpc/supporter/sustaining/Corydon76-home)
11:34.52*** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
11:38.22*** join/#asterisk Assid (n=assid@203.115.64.12)
11:38.27Assidsorry
11:38.28Assidgot cut
11:38.45Assidokay its showing me a new screen... downloading new application
11:39.07Assidi hope it doesnt need to do this again and again
11:46.18*** join/#asterisk Lino` (n=Lino@i577BF510.versanet.de)
11:57.00*** join/#asterisk QbY (i=user@cm-12-146-225-117.dhcp.geo-sc.southerncoastalcable.net)
11:57.30QbYAnyone know of documentation that shows how you can fully utilize the display on Polycom phones?
11:59.28stoffell_hQbY, talking about 601?
12:00.59QbYstoffell_h.. yes
12:02.22*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
12:10.27codebreakersomebody know how to connect n boxes together and route calls to any number between them. no matter on wich box the client has registered. without use of dundi. also like a active-active setup. i know there was an easy way but i forgot
12:13.07Assidhrmm
12:13.19Assidhow do put up a ringer of your own
12:13.30Assidi put it in saf.. i see it download
12:13.32jerliquecan anyone suggest the a way to "link" 3 or more asterisk servers together, so that coupled with ser each * server knows about users which have registered on other * servers via the dispatcher module in ser
12:15.56codebreakerjerlique: if you get an answer please also forward it to me or juast give me a hilight
12:16.27Assidi cant get a new ringtone :(
12:16.35jerliquesure ;)
12:16.58saftsackis there anywhere a tutorial to install capi?
12:18.30*** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
12:18.50Assiddoes the 301 even support custom ringtones?
12:23.07*** join/#asterisk xtr (i=Analogyo@S0106000c41ed11e1.vf.shawcable.net)
12:24.09Assiderr.. i keep getting this noise every few mins.. i think its for the registration.. is there a way to stop it from making that noise every time it registers ?
12:25.56*** join/#asterisk stuartcw (n=chatzill@softbank221025056004.bbtec.net)
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12:28.40saftsackhi has anyone of you chan_capi`
12:28.41saftsack?
12:31.46*** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl)
12:32.39stoffell_hcodebreaker, hm, also interested in your "n boxes" question..
12:32.53stoffell_hAssid, it has something to do with the voicemail maybe? (chirping sound)
12:38.41*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
12:39.47Assidnah..not voicemail
12:39.47Assidits around the time i have my timeout/registration
12:39.47Assid240 secs
12:45.29codebreakerstoffell_h: im asking me if nobody have "n boxes " connected together and as exmaple all clints a connecting to voip.mydomain and always get a different ip vi rrd-dns
12:46.51codebreakerthere is so much written about failover trunks with switches etc.. but not how to setup all the boxes to do sharing the knowledge
12:50.00stoffell_hcodebreaker, i'm not sure if it can be done in a good way, without dundi
12:55.07*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
12:56.41codebreakerstoffell_h: i think a had such a the time of asterisk ver 1.0.... and ist was really easy
12:56.54*** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com)
12:57.40*** part/#asterisk Alystair (n=bob@CPE001109c15241-CM00407b8794db.cpe.net.cable.rogers.com)
13:07.22*** part/#asterisk QbY (i=user@cm-12-146-225-117.dhcp.geo-sc.southerncoastalcable.net)
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13:34.42Assidargh..how the hell do you get the last nmbder dialled?
13:35.50[Airwolf]Does anyone has some experience with call forwarding ?
13:36.01[Airwolf]Because I mad this extention:
13:36.02[Airwolf]http://pastebin.com/688553
13:36.06saftsackdoes someone know the fritz isdn card?
13:36.11[Airwolf]And it seems to be working perfectly
13:36.24[Airwolf]But, if I dial the number it just isn't forwarded.
13:37.32[Airwolf]And I can't figure out why it isn't
13:41.37*** join/#asterisk darby_t (i=darby_t@aapn121.neoplus.adsl.tpnet.pl)
13:45.34*** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.233.134.Dial1.SanJose1.Level3.net)
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13:52.31Assidhow do i know if i have a poe cable or no
13:52.46*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
13:54.04austinnichols102Here's the URL for reference: http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx
13:54.41Assidi got this black fat patch cord
13:54.55austinnichols102They're saying that 8.2 works it just that the CID displays IP addy
13:55.17austinnichols102damn - wrong window
13:56.05Assidand some extra thing.. which seems that it should be clipped onto a a cable
13:59.24*** join/#asterisk Lino` (n=Lino@i577BD976.versanet.de)
14:00.54Assidoh.. i need a special kind of switch to use PoE ?
14:03.45coppicewell, power doesn't appear by magic, or the oil companies would be out of business :-)
14:05.15filecoppice: it doesn't?!? someone lied to me!!!
14:05.47*** part/#asterisk darby_t (i=darby_t@aapn121.neoplus.adsl.tpnet.pl)
14:06.36luke-jr_coppice: oil needs to be made somehow, so oil has its source too ;)
14:06.46luke-jr_coppice: eventually, it all goes back to the Sun, which might as well be magic
14:07.25coppicethe oil companies just steal the oil from the dinosaurs it really belongs to
14:07.43luke-jr_pfft, oil isn't from dinosaurs =p
14:08.10luke-jr_that's fiction
14:08.36luke-jr_not that I could tell you where it *is* from in reality o.o
14:09.05ManxPowerMy car gets 358 dead dinos to the mile!
14:09.13russellbluke-jr_: magic
14:09.21luke-jr_russellb: could be
14:09.26luke-jr_for certain definitions of magic
14:09.45russellbwhere magic is defined as the explanation for things you don't understand
14:10.19luke-jr_or potentially where it's defined as things which are outside the realm of science
14:10.34russellboil is not outside of the realm of science
14:11.34luke-jr_its origin could be, but I don't know
14:11.43luke-jr_could be part of the original creation
14:12.04russellbasterisk is powered by oil, you know ...
14:12.12luke-jr_cool
14:12.40luke-jr_it runs fine w/o oil for me tho
14:13.48coppiceit creaks a bit without lubrication, though
14:14.19*** join/#asterisk mutilator (i=WebChat@65.111.201.122)
14:14.35ManxPowerYay!  The ditch witch works!
14:14.44mutilatormornin everyone
14:14.50mutilatorhow goes saturday
14:14.53ManxPowerAlmost too well, actually, within 2 hrs of getting it they cut one of the water lines.
14:15.20luke-jr_hm, wonder if a digital camera can take pics w/o internal storage via Wifi...
14:15.33Qwellluke-jr_: Sure, why not?
14:15.38*** join/#asterisk VeNoMouS_ (n=jj@202.162.177.196)
14:15.39QwellJust don't expect it to be very fast
14:15.41*** join/#asterisk ApEtc (i=apetc@ip70-162-216-7.ph.ph.cox.net)
14:15.46VeNoMouS_asterisk in the news in nz again
14:15.48VeNoMouS_http://computerworld.co.nz/news.nsf/news/3E42037A86813A51CC25715D00105063
14:16.02VeNoMouS_tsk tsk nawty phisers using asterisk
14:16.05ManxPowerVeNoMouS_, Great, more newbies.
14:16.16VeNoMouS_VoIP services are appealing because they allow customers to set up numbers anywhere in the globe. And because they can be combined with telephone software like the open-source Asterisk PBX (Private Branch Exchange) product, it can be inexpensive for thieves to set up a professional-sounding line.
14:17.20*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
14:17.31mutilatori dun think newbs do that sort of thing..
14:17.41mutilatornewbs can;t even make a dial plan
14:17.52VeNoMouS_lol
14:18.15shido6help them out.
14:18.28luke-jr_Qwell: my actual goal is to do GPS-capable photos but have the GPS usable from a laptop ;)
14:18.33VeNoMouS_heh man, ive pretty much finshed most of my managment frontend for our cdr shit
14:18.35Qwellshido6: I think in this case, helping them would be a bad idea. :p
14:18.36Qwell<VeNoMouS_> http://computerworld.co.nz/news.nsf/news/3E42037A86813A51CC25715D00105063
14:18.37Qwell<VeNoMouS_> tsk tsk nawty phisers using asterisk
14:18.55shido6thanks qwell
14:18.58shido6where's my foot
14:19.05Qwellin your mouth, I imagine. ;)
14:19.09shido6:)
14:19.09VeNoMouS_write mixmonitor to file, transcode to mp3, insert mp3 into mysql, stream mp3 on the fly from mysql via php
14:20.10VeNoMouS_heh so i can list all the calls for the day, and u can play back each call
14:20.41VeNoMouS_Qwell wanna see?
14:20.55Qwellsee what?
14:21.00Qwelloh
14:21.11Qwellnah, too early
14:21.15VeNoMouS_lol
14:21.16VeNoMouS_Sun Apr 30 02:20:47 NZST 2006
14:21.19VeNoMouS_its late for me
14:24.21VeNoMouS_wow its so talkie in here!
14:25.37*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
14:26.13*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
14:29.59*** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage)
14:34.25Assiderr.. i keep having this irritating sound .. every few mins.. i think its of the phone registering every 240 seconds
14:34.30blitzrageanyone have experience with polycom phones? I'm wondering if it's possible to block the call forwarding feature on the phone itself -- or, alternatively -- to detect a 302 redirect in asterisk (1.2.x)
14:34.32Assidbut can i have it disable that noise
14:37.11*** join/#asterisk ComputerWarm (n=donc@HS196-230-97.nt.net)
14:37.23shido6my spa blinks at me everytime it registers
14:37.27shido6so i set it for 3600
14:37.32ComputerWarmHello everyone
14:37.40Assiderr.. okay its ever 3 odd minds
14:37.54Assidwhich still isnt 240 seconds
14:38.25ComputerWarmI have a question I installed h323 that comes with asterisk, i got no errors while i was compiling but its still not installed.... I did as the instructions said
14:38.37ComputerWarmBut when i try to load it. there is no apps for it
14:38.48shido6then you didnt follow the directions
14:38.50shido6:)
14:39.09ComputerWarmya i did everything the Readme and install file said
14:39.10Assid<PROTECTED>
14:39.23shido6most ppl say that and miss a line or two :)
14:39.41ComputerWarmlet me start from the beginning again
14:39.46shido6blow away all remnance of h323 and read the README again
14:40.18nrwblitzrage: do you want to disable the ability to do the redirect, or force the phone to take the call even with call forwarding on
14:40.49Assidwhats lineSeize for?
14:41.29ManxPowerAssid, are you sure it's not the MWI indication ring?
14:41.50Assidhow do i know?
14:42.12ManxPowerwell, does the SIPura only do a partial ring when you have voicemail?
14:42.21Assidits polycom 301
14:42.33ManxPowerAssid, so what is the problem?
14:42.44*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
14:42.48Assidit keeps beeping every minute or so
14:42.54ManxPowerdoes the annoying sound only happen when the MWI light is blinking?
14:43.33Assidyou mean the red light on top
14:43.40Assidi thought thats power light
14:43.59ManxPowernoname32, that's the MWI light.
14:44.05ManxPower<PROTECTED>
14:44.07*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
14:44.14Assiddamn
14:44.14Assidyes
14:44.15ManxPowerthat "annoying sound" is to tell you that you have voicemail.
14:44.18Assidits cause of that
14:44.26Assidjust curious tho
14:44.47ManxPowerI managed to get rid of the annoying sound, but it took 2 days and reading the Admin guide a zillion times.
14:45.04Assidwhere do i program a button for voicemail..
14:45.19ManxPowerAssid, read the admin guide.
14:45.31ManxPowerand the button for voicemail is only for CALLING voicemail.
14:46.02ManxPowerIt's for people too lazy to dial the voicemail extension on your Asterisk system.
14:46.30Assidyeah i know
14:46.53Assidit works for a 501 i remotely configured
14:46.57*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
14:47.00Assidjust dont know what to do on my 301
14:47.13Assidnever seen a 501 in hand .. was configuring my friends box
14:47.15ManxPowerIN FACT, programming the voicemail button was discussed on -users within the past 14 days.  I guess you must have missed that thread.
14:48.06Zeeekgentlemen...
14:48.31ManxPowerZeeek, someone in Paris was looking for an Asterisk consultant.
14:48.38Zeeekyes, we know him
14:48.52Zeeeknot that I'm a "consultant" anyway ;)
14:49.03ManxPowerAh.  OK.
14:49.11Zeeekwe have an asterisk group now meeting every month
14:49.15Zeeekbut thanks!
14:49.46ZeeekI'm having tollfree DID woes at the moment
14:50.08Zeeekhow's thing with you, Manx ?
14:51.10*** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg)
14:51.10ManxPowerZeeek, Pretty good.  Living on a mountian now. 8-)
14:51.19Zeeekheh, understandable
14:51.27ManxPowerNo more fleeing Storms of Doom
14:51.40Zeeeknow you'll be struck by lightning !
14:51.40ManxPowerI hope to have my cabin within 2 months.
14:51.52Assidhrmm.. 301 doesnt have a seperate button for messages
14:52.13coppice"No more fleeing Storms of Doom" sounds like you stopped playing some addictive computer game
14:52.36ManxPowercoppice, we started calling Katrina "Katrina: Storm of Doom" after a while.
14:52.49nrwassid: no it does not
14:52.57nrwyou can assign the second line key to it if you so wish
14:53.18coppiceManxPower: typically american. in the UK it would have been referred to as "the breeze"
14:53.35Zeeekchin wot
14:53.43ManxPowerUh uh.
14:54.12Zeeekin new jersey it might have been know as Da Breeze
14:54.24ManxPowerI'll bet they play sports without padding and eat rare meat too!
14:54.52Assidi wonder whats the price difference between 301/501
14:55.05ManxPowerAssid, that would depend on where you get them from.
14:55.19nrwassid: about 150 bucks
14:55.21Splashrm... my 7960 seems to append the domain of the SIP server to the callerid num of inbound calls...
14:55.28nrwactually about 120
14:55.45[TK]D-Fendernrw : 120 WHAT?
14:55.52coppicecome on. the UK calls the atlantic ocean the pond, and some dislike that kind of overstatement. storm of doom would only be good for laugh
14:55.55Assid120 bucks for polycom301?
14:56.05nrwv<Assid> i wonder whats the price difference between 301/501
14:56.11nrw<nrw> assid: about 150 bucks
14:56.18nrw<nrw> actually about 120
14:56.22[TK]D-Fendernrw : The difference isn't 120$ in USD or CAD....
14:56.27[TK]D-Fendernot by a long shot
14:56.27ManxPowercoppice, well, until you live thru one.
14:56.39Assidso its $80 for a 301 ?
14:56.51[TK]D-FenderIP 301 = $115 USD, IP 501 = $170 USD
14:56.55nrwwow
14:57.01ManxPowerAssid, call Gauston at Avenue Computer Supply,
14:57.01coppicethe US will probably live through some more this year
14:57.03nrwwhere the heck are they those prices?
14:57.18Assidhrmm
14:57.19[TK]D-Fendernrw : Leanr to shop around.... www.atacomm.com
14:57.20ManxPowercoppice, yup.
14:57.26ManxPowerLet me find his contact info.
14:57.28Assidthey shoulda bought me a 501 :(
14:57.42[TK]D-FenderI got mine for $135 CAD and $200 CAD respectively.
14:58.13Assidnow the only thing i gotta get working is this ringtone
14:58.19Assidi wanna get the CTU ringtone in there
14:58.20*** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
14:58.31Assidit just doesnt work :(
14:59.00nrwmy 5012 cost 250 bucks
14:59.04nrw501s even
14:59.07nrwsigh
14:59.23ManxPowerAssid, call Gaston Gureau at Avenue Computer Supply, 225-615-7297  Tell him "Eric" sent you.  I do not get a comission, but he is a friend.
14:59.41ManxPowerI *THINK* he has decent prices on Polycoms, you should shop around.
14:59.58ManxPowernrw, got them on the greymarket, eh?
15:00.07Assidnice.. will keep it in mind
15:00.14Assidim actually in india..
15:00.40ManxPowerI doubt he'll ship outside the USA
15:01.07*** join/#asterisk Kernel_Core (n=I@193.251.135.118)
15:01.08Assidyeah
15:01.13Assidthats wheree alot of my time went
15:01.16Assidi was waiting for the phone
15:01.23nrwmanxpower: im not sure where things really come from. we have a purchasing guy that handles all that stuff. i just have to count the cost against my budget
15:01.24Assidfor someone to get it
15:01.40nrwand i could have gotten a few more phones if i knew they were 80 bucks less than we got them for
15:01.58ManxPowernrw, Avenue has decent prices on a lot of stuff.
15:02.24ManxPowerbut for the really cheap stuff, best to go with someone else.
15:02.33*** join/#asterisk Abydos313 (i=abydos31@ppp-71-133-210-73.dsl.irvnca.pacbell.net)
15:03.12Assidhrmm.. gonna keep that avenues in mind
15:03.17Assidfor the guys i work for
15:03.22Assidthey always need more phones
15:03.26ManxPowerAs I said, I doubt he will ship outside the USA
15:03.44Assidyeah.. those guys are in the US
15:04.07ManxPowerWe get special pricing from Avenue, so I can't tell you what is standard price is for polycoms
15:04.55ManxPowerI doubt they will provide support, other than to maybe provide firmware updates on request.
15:05.33Abydos313anyone know how to setup/check voicemail with telasip?
15:13.39ZeeekManxpower didn't voipsupply once have good prices on Poly?
15:14.00Zeeekthey have good support and do ship internationally
15:14.20[TK]D-FenderZeeek : compared to others "worse" prices, yes.  www.antonline.com has the lowest I've seen to date.
15:14.42[TK]D-FenderVoIP Supply is good for the all-around service though.
15:14.59ZeeekI can testify that they did a good job for us on a return
15:15.11Zeeekand make firware available to customers
15:15.13*** join/#asterisk heka (n=heka@82.114.68.123)
15:15.19Zeeekkinky, that firware
15:15.29Zeeekwooden trousers and all that
15:16.28Zeeekdoes anyone have tollfree DID and if so who is providing them?
15:17.20ManxPowerZeeek, no idea.
15:17.27hekadidx had something similar! but Im not so sure.
15:17.35Zeeekdidx is a broker
15:17.43ManxPowerZeeek, teliax for me
15:18.01ManxPowerOf course, all ITSPs suck.  Teliax seems to suck less than most.
15:18.19ZeeekI had an account at one time. They do tollfree DID?
15:18.40Zeeek<going to look>
15:19.00*** join/#asterisk telenieko (n=marc@167.Red-80-35-144.staticIP.rima-tde.net)
15:19.09Zeeekyeah they do
15:19.14teleniekoanybody here knows how does the bristuff code work ? :o)
15:21.06jake1932ManxPower: that's misleading to say unlimited on https://www.teliax.com/newaccount/?r=1&cp=default
15:21.49ManxPowerjake1932, they disclose their softlimits.  Like ALL ITSPs, unlimited is just a marketing gimick
15:23.11jake1932i can understand an "Unlimited" Plan - but they're showing a table of what is included - and the softcaps below - seems pretty cheezy
15:24.10*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
15:26.01*** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
15:26.05*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
15:26.42jake1932here's a new marketing idea - advertise what you actually sell - and stop with the bait and switch
15:27.23jake1932(not taking away from the service - which from what you said is good
15:28.07jake1932)
15:33.32*** join/#asterisk Flosoft (n=admin@213.219.160.100)
15:33.33Flosofthey
15:33.45FlosoftI am looking for an Asterisk Frontend
15:33.52Flosoftwhich I can install on a Plesk Server
15:34.06QwellI hear vi is a good frontend
15:34.12[TK]D-FenderTry vi, emcs, pico, nano, mc,
15:34.24Flosoftwell ... maybe something which is nicer ;)
15:34.31Flosoftlike: Iritgo
15:34.41[TK]D-Fendergedit, kedit, OOo Writer?
15:34.47jake1932ultraedit - on windows
15:34.57Flosofta Non-texteditor interface
15:34.59QwellMS Word?
15:35.03QwellExcel?
15:35.05[TK]D-Fenderhehehe, ok, I'm outta here
15:35.11[TK]D-FenderQwell : No... Word sucks :)
15:35.23[TK]D-Fenderlater!
15:35.36Flosoftany constructive help?
15:36.06jake1932AMP - try #freepbx
15:36.20Flosoftwell ... it is difficult on plesk I was told
15:36.33jake1932it's a PIA on anything
15:36.41FlosoftPIA?
15:36.48jake1932easiest to just edit the files you need with a text editor
15:36.59QwellPITA?
15:37.10jake1932~PITA
15:37.11jbotsomebody said pita was pain in the ass
15:37.38Zeeekwrong! It's bread
15:38.03Flosofthehe
15:38.13jake1932~PIA
15:38.15jbotpia is, like, ask me about pita
15:38.17Qwelljbot: pita is also a bread-like food
15:38.19jbotQwell: okay
15:38.42*** join/#asterisk Samoied (n=Samoied@201-35-214-13.fnsce703.dsl.brasiltelecom.net.br)
15:39.07Flosoftwell ... I really like Freepbx
15:39.19Flosoftbut to install it on Plesk is quite difficult it seems
15:39.22coppiceQwell: isn't that pitta?
15:39.29Qwellcoppice: no, don't think so
15:39.32Qwelljbot: define pita
15:39.45Qwellat least, not in English
15:39.47FlosoftTalima / Iritgo seems to be perfect ... but I can't find it anymore
15:40.26QwellPita (also called pitta or pita bread) is a round flat wheat bread made with yeast. It is traditional in many Middle Eastern and Mediterranean cuisines and is believed to have originated in Ancient Greece.
15:40.42Qwellwell, I guess both are acceptable...the former is the preferred spelling though
15:40.47*** join/#asterisk pardove (n=pardove@217.219.250.24)
15:41.00pardoveis this command valid in extensions.conf: switch => SIP/user:secret@server/context
15:41.02jake1932flosoft - you probably won't get any help in here on that - check http://www.voip-info.org/wiki-Asterisk+GUI
15:41.18Qwellpardove: I don't think so, no
15:41.20Flosoftthx... i'll take a look
15:41.32coppiceQwell: a google search seems to conflict with you. pitta bread gets massive hits, like www.nutrition.org.uk/upload/bread%20pitta.pdf
15:41.32Qwellpretty sure you can't do SIP switch...
15:42.02Qwellhttp://googlefight.com/index.php?lang=en_GB&word1=pita&word2=pitta
15:42.07Qwellsorry, pita wins :P
15:42.09pardovecan i have video calls when using iax trunks?
15:43.19coppiceQwell: you didn't qualify it with bread. you got all the GWBush references with pita :-)
15:43.25*** join/#asterisk Laureano (n=Laureano@host172046.metrored.net.ar)
15:43.28Qwell:P
15:43.29LaureanoHello
15:43.57Qwellpita bread still wins, by a significant amount
15:44.05pardoveQwell: can i have video calls when using iax trunks?
15:44.17LaureanoDid anyone uses VoipTalk with IAX?
15:44.22Zeeekyes
15:44.23Qwellpardove: I don't know.  and if I did, I would have answered the first time you asked, less than 2 minutes ago
15:44.35ZeeekLaureano might be off today
15:44.42Zeeekthere was a recent change
15:44.42pardoveQwell: sorry ;-)
15:44.58LaureanoI supose that.
15:44.59*** join/#asterisk gr0mit (n=guest@extrt.txrx.org.uk)
15:45.07Zeeekyou have problems?
15:45.17LaureanoI see in the support page that we have to change the server to iax5.voiptalk.org
15:45.28ZeeekI chenged it and now nothing works
15:45.43ZeeekI'm guessing they are doing some work now or something
15:45.49LaureanoOhhh
15:45.56ZeeekJust a guess
15:46.04LaureanoI guess the same.
15:46.15Zeeekwell now we know there are 2 of us :)
15:46.26LaureanoBecause yesterday, for some moments, I can't do outgoing calls.
15:46.30Zeeekaha
15:46.42ZeeekI don't use them much, only for europe
15:46.48LaureanoNow it works. But the incoming calls are broken.
15:47.00Zeeekit works since when?
15:47.08LaureanoWhen I try to pass audio in the call, I get an "Invalid Call" from IAX.
15:47.47LaureanoI guess that the outgoing calls are working since last hours of Yesterday.
15:47.59Laureano(Please note that I'm in Argentina)
15:48.02Zeeekok
15:48.16LaureanoBut, its good to see that I'm no the only one with problems.
15:48.22LaureanoThank you Zeeek
15:48.34Zeeekwell if it continues, I'll have to contact the
15:48.35hekaanybody can help me modify an perl agi?
15:48.46ZeeekLaureano no problem
15:48.55*** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl)
15:49.01LaureanoI try to contact them, but apparently they work Mon-Fri,9:00-18:00
15:49.17LaureanoA least the service number that I can dial from here.
15:49.33LaureanoMaybe the UK customers can call right now, but not me.
15:49.45ZeeekI've never tried to call them
15:50.05LaureanoWell, that was the first time I tried.
15:50.29LaureanoBRB
15:52.31LaureanoBack
15:52.44*** join/#asterisk darby_t (i=darby_t@aapc207.neoplus.adsl.tpnet.pl)
15:52.59LaureanoWell, I think that its just a matter of time to be working again. Lets wait.
15:53.20ZeeekLaureano is outgoing working in IAX right now for you?
15:54.13LaureanoYes.
15:54.31LaureanoI use the VoipTalk account to make calls to UK, and works fine.
15:54.37Zeeekok
15:54.44ZeeekI'll check later
15:54.54LaureanoAren't working for you?
15:55.09jake1932heka - although i probably can't help you with that, usually the guys on here respond better if you ask a very specfic question about your perl agi - like, what you're trying to do with it
15:55.17Zeeekit was not working a few hours ago. I can't check now because the phones are in use
15:56.20LaureanoBut... you can't check it because you can dial to VoipTalk?
15:56.35*** join/#asterisk vittogio (n=vittogio@host173-40.pool8259.interbusiness.it)
15:56.50vittogiohi all
15:57.14vittogioi have a critical issue on my asterisk server, it is there someone that could hep me?
15:57.26*** join/#asterisk salviadud (n=ralfalfa@dsl-200-78-64-10.prod-infinitum.com.mx)
15:57.36hekajake1932: did it by myself. thank you.
15:58.29LaureanoTell us about your problem I we will see if we can help you vittogio
15:58.51LaureanoZeeek, Why don't you use the Asterisk Auto Dialer?
15:59.28vittogiook, i have 2 asterisk server connected via the PRI port, very often this link get a reset and all calls are dropped
15:59.49jake1932vittogio: did you swap the cable yet?
16:00.21vittogioyes, i did and i checked the signalling, master/slave also
16:00.36jake1932irq issues maybe?
16:01.04LaureanoWhen the servers drop the calls. Do you see something in the Asterisk console or in the full log?
16:01.27vittogioi made a check and "unfortunately" there were no irq issue
16:01.45vittogiothere is RELEASED
16:02.18LaureanoThe span goes down and then up?
16:02.25vittogioit is like the D-channel has some issue, also because it seems that the calls move to full duplex to half duplex audio
16:02.36vittogioyes the span do down and then up
16:02.52vittogioi have no IP trunk on the system
16:03.17LaureanoOk. I think that I know what can be, but I can't find the link with the solution. Give me 5 minutes.
16:03.27vittogiook
16:04.31jake1932vittogio - while he's looking - how come you used two PRI cards to link the servers, and not IP?
16:05.29vittogiowe have a point to point connection between 2 sites via a Clear channel
16:05.43jake1932ok
16:06.08mutilatordoes echo can settings in zapata reload when you reload
16:06.11mutilatoror does it need a restart?
16:06.37vittogioin the zapata the echocancel is set to no
16:07.40Laureanomutilator, You need to restart.
16:08.20vittogioi.e. i have no problem with the connection to the PSTN network, i mean no random restarts
16:10.01Laureanovittogio, Do you use the "pri intese debug" command?
16:10.10LaureanoThat always help.
16:10.12*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
16:10.16vittogioyes
16:11.34LaureanoAnd no clues with that?
16:12.11vittogioi mean you can see that a reset occurred by the remote party
16:12.37vittogiowe saw the involved timer, the T308
16:13.11vittogiobut if i connect 2 legaycy pbx over the same link, everythink is working fine
16:14.23vittogioi have also connected 1 asterisk server to an Alcatel PBX usingthe same link and wha i saw is that the system has been able to locate channel over the 30s regularly used by an E1
16:15.01vittogioit seems to be something related to the software hdlc implementation
16:16.46LaureanoDo you see any HDLC abort errors?
16:16.56vittogioyes
16:17.14vittogioHDLC Frame(8)
16:17.21vittogiosorry FCS(8)
16:17.28*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
16:17.47coppicewhat makes you think it is anything to do with the software HDLC?
16:19.00Laureanovittogio, please check this: http://lists.digium.com/pipermail/asterisk-users/2005-September/118021.html
16:21.18vittogiocurrently the resetinterval =never
16:21.39vittogioas i saw that the range of the value starts from 60s
16:21.53vittogiodo you think that "0" is accepted?
16:22.50vittogiocoppice: i tried many setting over the zapata and zaptel, so i thought that should be something related to the software
16:23.41coppiceyou have a problem which everyone else doesn't, so the software must be wrong. well, that seems a logical deduction. :-)
16:24.44vittogiowhen you have issue you try to figure out "all" the possible causes".....
16:25.38coppiceits quicker if you start with the likely ones. commonest reason for link errors is frame slips. are you slaving to the clock from the PBX?
16:26.12vittogioi tried 2 different situation
16:26.16*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
16:26.20Laureanovittogio, This link mayba also can help: http://www.ctunion.com/node/95
16:26.42vittogiothe first is to connect asterisk to a legacy pbx and hget the clock from the last one
16:26.54vittogiothe other is having 2 asterisk server connected together
16:27.01vittogiothe issue is the same
16:27.27Flosofthas anyone got experience with voiceone
16:27.29Flosoft???
16:27.39coppicewhat did you put in zaptel.conf?
16:27.40LaureanoFlosoft, No, sorry.
16:29.18vittogioChange the server... not so easy...
16:29.58vittogioi read the article in the past, i was was also thinking the same, but is something i cannot perform any action
16:30.05vittogioin the zaptel?
16:30.29LaureanoI know that isn't easy to change a server. But, did you test with a loopback cable, for example?
16:31.31vittogiowith a loopback cable you get that: you are master and also the other party ... or slave
16:31.41*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
16:31.46vittogioi made the following
16:32.00vittogioi ran some calls using the outgoing
16:32.04LaureanoI know that. But you didn't see any error?
16:32.50vittogiothe error is occurring only when the channel is fully busy and other calls are trying to access to it
16:33.05LaureanoAha
16:33.32vittogioi tried to replicate the errors on local making auto calls using the .call file
16:33.38vittogiowell no issue
16:34.18LaureanoSo. If you have all the 30 channels of the trunk in use, and then you try to place a new call; the trunk goes down and up?
16:34.24vittogiothe span in the master server is set: span=1,0,2,ccs,hdb3
16:34.39vittogioafter 15/20 minutes
16:35.41vittogioon the debug what is missing is the time: when the error occurs....
16:35.56LaureanoAnd this didn't happen when you connect * to the Legacy PBX?
16:36.44vittogiothis is happening only with 2 asterisk server connected each other or when as asterisk server is connected to a legacy pbx
16:37.13LaureanoAha
16:37.16vittogiois i connect 2 legacy pbx together (Alcatel and Ericsson) no disconnection
16:37.42vittogioi mean it is very very strange
16:37.47Qwellsounds like your timing is pretty wrong
16:37.56vittogioyeah, it is
16:38.12LaureanoIf you were only using *2* connection, I would suggest you to control the used channels with CheckGroup(). But...
16:38.26coppicevittogio: you are whining a lot, but not providing any information.
16:38.57vittogioinformation like?
16:39.21coppicewhat is in your zaptel.conf. you only erplied with part of the answer
16:39.27vittogiook
16:40.57vittogiospan=1,1,2,ccs,hdb3 - span=2,1,2,ccs,hdb3 - span=3,1,2,ccs,hdb3 - span=4,0,2,ccs,hdb3
16:41.23vittogio#1 and # 2 connected to the PSTN
16:41.36ZeeekLaureano are you using the new server for voiptalk? It still isn't working for me
16:41.37vittogio#3 cnnected to Alcatel (that is master)
16:41.52*** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
16:41.52coppiceso, you have 3 spans all set to be the primary clock source. that doesn't make much sense
16:42.07coppiceyou don't have crc4 set, which is normal for ISDN E1s
16:42.29vittogiocrc4 is not required by the provider, so i cannot set it up
16:42.37LaureanoZeeek, Yes, I'm using iax5.voiptalk.org. But, the 2 domains point to the same IP address.
16:42.56vittogioyou see 1 in al the spans as Digium told me to set them in this way
16:43.00Zeeekinteresting, I didn't even check that :)
16:43.03coppicewell, E1s without CRC4 are used in many places, and they are quirky. CRC4 was added to the E1 spec for a reason
16:43.27*** part/#asterisk ComputerWarm (n=donc@HS196-230-97.nt.net)
16:43.28vittogioi know but the provider is not using it
16:44.42coppicechange your zaptel.conf so you don't have multiple E1 set to the same priority. I don't know if matters now, but it used to. quirky things used to happen
16:45.43vittogiothis is what i knew but i got a mail from digium saying to use 1 (slave) and 0 for master, avoiding the use of 2..3 and so on
16:46.17vittogioyou are suggesting to use 2 for the span#3, right?
16:46.35wunderkinvittogio, who at digium told you to do that?
16:46.44vittogiothe support
16:46.45coppiceof course not. you want that to be your clock source, so it needs to be 1
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16:47.20vittogiothe clock source should come from the provider
16:47.26wunderkinheh, man this is rough, who at support
16:47.52coppiceI thought you said 3 was to be your clock master?
16:48.04vittogiosolution but not the man, it is not correct
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16:49.28vittogiothe clock source is the alcatel, but it is preferred to use the PSTN as clock source
16:49.46vittogioso do you think that i should better mange the clock between the PBX?
16:51.07vittogiothe problem is coming from the fact that at 1 side i have the PSTN and from the other i have the other PBX
16:51.32*** join/#asterisk relateness (n=relatene@81.52.161.78)
16:52.01coppiceyou need something like:
16:52.03coppicespan=1,1,0,ccs,hdb3
16:52.04coppicespan=2,2,0,ccs,hdb3
16:52.06coppicespan=3,3,0,ccs,hdb3
16:52.08coppicespan=4,0,0,ccs,hdb3
16:52.37relatenessHi, can any one help me to find a good introduction to asterisk
16:52.43Qwell~docs
16:52.44jbotmethinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
16:52.53vittogiosame issue with that it was my first configuration
16:53.31coppiceis the PBX set to slave to you?
16:53.53vittogiono
16:54.17coppiceis the PBX locked to the PSTN?
16:54.29vittogioyes
16:54.35*** join/#asterisk apardo (n=apardo@87.217.145.245)
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16:55.09relatenessthank you !
16:56.03vittogiobefore having asterisk connected, i had  2 pbx connected, i have repliacted the same timing setting but no results
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16:57.28gaupehow is the current (not beta) firmware on the GXP-2000 now?
16:57.54vittogioi think that i am bothering boring you with this case
16:58.01gaupeI've got a installation with 16 phones, just considering if I should replace them or not
16:58.05asterboygaupe, awesome
16:58.18QwellIf you have to ask that question, it's time to switch phones
16:58.30gaupeasterboy: compared to previous versions, or compared to other phones?
16:58.40asterboyboth
16:59.11gaupeQwell: I haven't taken over the installation yet, but I'm going to - and since it's not more than 16 phones - they can be replaced quite easily
16:59.13vittogiocoppice,Laureano, thanks for your suggestions. if i find out a solution i will inform
16:59.15asterboythe new menu and entire retrofit gives the phone what it should have had from the start
16:59.45gaupeasterboy: I looked at the software when in beta-stages, and it looked a lot better :)
17:00.07asterboyits a MUST upgrade
17:00.12gaupe:)
17:01.06gaupeI've got 10 Thomson ST2030 in at my office, considering using them - since I've already got a autoprovision setup ready for them
17:01.12a1fahello hello
17:01.13a1fahello
17:01.32a1fahey, i need to make a extension for i event, if it goes to i 3 times, to dial a specific extension
17:01.39a1facan i just use vars and gotoif
17:01.41a1fa?
17:01.45vittogiobye
17:02.36a1fahow can i add +1 to a variable?
17:03.47codebreakera1fa: of course you can do this. if somebody tries 3 times to select the rigth ivr and fails let call the operator
17:04.21a1facodebreaker : yeah.. do you have the code allready so i dont have to reinvent the wheel?
17:05.12codebreakerone moment i have read tonight something about this in the asterisk book. and i also know there was a extension on my system. i will poste du pastebin few minutes
17:05.28a1fasweet
17:05.29a1fathanks man
17:05.33QwellSet(VAR = $[${VAR} + 1])
17:05.40a1fai was goint to use set(var...
17:05.44a1facool
17:06.16a1faexten => s,1,Set(SOMEVAR=${MATH(${SOMEVAR}+1)}) ; increment
17:06.24a1faso you dont have to use math?
17:06.31Qwellno...
17:06.43codebreakerhttp://pastebin.com/688858
17:06.47a1facodebreaker : thanks
17:06.49Zeeekdoesn't the [] do the evaluation
17:06.52QwellZeeek: yes
17:07.09Zeeekhence the non need for math
17:07.41a1fasweet
17:07.54a1facodebreaker : thanks man.. i am going to simplify this  and repaste it
17:08.27codebreakera1fa: http://www.asteriskdocs.org/modules/news/  read there the asterisktfot book. really good
17:09.19Flosofthas anyone got experience with voiceone?
17:10.59a1fadone
17:11.27a1fahere is my version
17:11.35a1fa"basterdized" version of timeout
17:11.40a1faand invalid re-entry
17:11.50a1fahttp://pastebin.com/688866
17:12.47a1faworks great
17:14.21a1faany objections?
17:15.11codebreakeryup. often i only need a starting point
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17:22.20Flosoftcan anyone help me installing an Webinterface?
17:22.48a1fano
17:22.55a1faFlosoft : apm? you mean?
17:22.57a1faapm sucks
17:23.13Flosoftwell I am looking for one that I can run on my Plesk Server
17:23.16*** join/#asterisk xtr (n=94752345@S0106000c41ed11e1.vf.shawcable.net)
17:23.26FlosoftI thought of acami or voiceone
17:23.47*** join/#asterisk inv_arp[work] (i=junya@adsl-10-153-159.mia.bellsouth.net)
17:23.59Flosoftfreepbx is nice .. but I can't use it afaik
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17:25.49asterboyFor a great GUI * ....
17:25.49asterboy...use...
17:25.49Flosoftyes .........
17:25.49asterboyvim and X Windows
17:25.49salviadudyeah
17:25.49Flosoftalready know that joke
17:25.49salviadudthe cli rocks!
17:25.50asterboyyep
17:25.50salviadudair guitar
17:25.50FlosoftI really need one
17:25.57a1fa:P
17:26.19Flosoftplease ... I really need some help
17:26.50*** join/#asterisk ManxPower (n=ewieling@dpc67142183150.direcpc.com)
17:27.40a1faaww. poor baby
17:27.45a1fawhy do you want to use a gui?
17:27.50a1faisnt there a readme or install?
17:27.51FlosoftI want an interface to manage asterisk on a shared webhost
17:28.02a1fawhy not use vi or pico
17:28.02Flosoftthe problem is plesk
17:28.10a1fai dunno what plesk is man
17:28.10FlosoftI can't fuck up the Webserver configs
17:28.15ManxPowerFlosoft, Let us know when you write the interface.
17:28.20FlosoftPlesk = cPanel
17:28.28a1fadude
17:28.31a1fajust ssh to the box
17:28.37Flosoftwell I can
17:28.38a1fai once wanted to use APM
17:28.41a1fabiggest mistake ever
17:28.41Flosoftit is a dedicated server
17:28.48a1fait takes control over your configs
17:28.49ManxPowera1fa, It's some silly WEB/GUI thing for people that can't or don't know how to admin a box.
17:28.51a1faand it needs mysql
17:29.07a1famysql takes too much memory and cpu
17:29.09a1fa;)
17:29.10coppicededicated? I like hard working servers :-)
17:29.14*** join/#asterisk gaiadg (n=gaiadg@cpe-74-64-42-87.nyc.res.rr.com)
17:29.19Flosofthehe
17:29.35Flosoftwell ... Freepbx looks nice ... but I don't know how to install it.
17:29.57FlosoftI would need some help setting it up
17:30.17a1fahehe
17:30.21a1fathis is * not freepbx
17:30.21*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
17:30.27Flosoftwell
17:30.32FlosoftI found Voiceone
17:30.34a1fawhy dont you get the source
17:30.35salviadudfreepbx is a nightmare
17:30.36Flosoftor Acami
17:30.49salviadudyou see, with *
17:30.53gaiadghello, quick question. I have a context in sip.conf for three extenions. What is the best way to setup outbound calleid to be the cid of the sip account?
17:30.54salviadudYOU can be the artist
17:31.06salviadudi see a extensions.conf file made in freepbx
17:31.18salviadudit's a jungle of macros and nonsense
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17:31.51salviadudi prefer raw asterisk, for i can bake it the way i want it to
17:32.00salviadudmedium rare
17:32.06salviadudor... sizzling hot
17:32.11Flosoftwell ... I don't know how to configure asterisk without Webui
17:32.14salviadudyou get the idea
17:32.22salviadudflosoft, how old are you?
17:32.25Flosoft16
17:32.28*** join/#asterisk ast_freak (n=jesse@12.28.106.2)
17:32.46salviaduddude...
17:32.55salviadudi remember when i was 16
17:33.09salviadudback then... the internet pr0n was traded, not leeched
17:33.19Flosoftlol
17:33.53salviadudmy recommendation is that you read the asterisk handbook
17:34.13salviadudanother question
17:34.17salviadudwhat distro are you using?
17:34.22FlosoftDebian 3.1
17:34.49Flosoftplease ... I really just want to setup an Webinterface that doesn't conflict with Plesk
17:35.06salviadudi honestly don't know man
17:35.11*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net)
17:35.26salviadudi doubt you'll find someone that would want to help you out with that, it is very off topic
17:35.35*** join/#asterisk n4y (n=tmalkut@host-ip2-24.crowley.pl)
17:35.54Flosoftwhere could I find some then?
17:35.58Flosoft*someone
17:36.17*** join/#asterisk darby_t (i=darby_t@aapa56.neoplus.adsl.tpnet.pl)
17:36.47*** join/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
17:37.14ManxPowerFlosoft, Try the asterisk-biz mailing list.
17:37.30ManxPowermost of here have no interest whatsoever in doing custom CGI programming for interfacing to Asterisk.
17:37.45Flosoftwell not custom
17:38.06FlosoftI just need help installing a Webinterface
17:38.13ManxPowerIt's hard to do, complex, and not very rewarding.  Also since the customer seldom actually knows what they want, poor project management frequently makes such projects a total disaster.
17:38.23ManxPowerFlosoft, which web interface?
17:38.47ManxPowerAsterisk does not come with a web interface.
17:39.39FlosoftVoiceone, Acami ... maybe even freepbx if it can work without changing the config of the webservers
17:40.07ManxPowerI've never heard of the first two, the last one is supported in #FreePBX.  NONE of these come with Asterisk.
17:40.24Flosoftwell there is none that comes with asterisk
17:40.25*** part/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
17:40.47ManxPowerTry #FreePBX.
17:41.12Flosoftjust did
17:42.39salviadudthe guys at #freepbx are there, sometimes
17:42.44salviadudi can understand your grief
17:42.55salviadudstill. we encourage you to read and stay in school
17:43.07salviadudthat way, you'll really learn
17:44.48Flosofthehe
17:45.57coppicestill. we encourage you to read and stay in school, as it keeps down the length of the dole queue :-)
17:49.22codebreakersomebody know how to connect n boxes together and route calls to any number between them. no matter on wich box the client has registered. without use of dundi. also like a active-active setup. i know there was an easy way but i forgot
17:50.05ManxPowerFlosoft, Where are you located?
17:50.23ManxPowercodebreaker, use Dundi or ENUM
17:51.51codebreakerManxPower: how high is the load for every request on the dundi/dnsserver? all this servers are connected to the same network i only ave them for loadbalancing/failover etc
17:53.20ManxPowercodebreaker, I have no idea, but no matter what you're going to have to have some method of querying extensions.  Read up on the dundi website.  Dundi is designed for this sort of stuff, that is it's advantage.  ENUM is based on DNS so people know what issues to expect in DNS, that is it's advantage.
17:53.45ManxPowercodebreaker, there is also a Wiki page on fail over, loadbalancing, etc.
17:55.02codebreakerManxPower: i know. but the all use some ser in front of it. but nowhere(or i dont found it) it is described how to have more servers with the same iax.con extensions.conf etc..
17:55.38ManxPowernone of the clients I have are willing to spend the money on failover, and none of them are large enough for load balancing.  I've used ENUM to reduce the amount of work to manage the servers.
17:56.05ManxPowercodebreaker, look into Realtime support in Asterisk.
17:56.13*** part/#asterisk alephcom (n=alephcom@host75.net14.mcsnet.ca)
17:57.15codebreakerManxPower: there is one way i think to go. when a clint registeres then this asteriskserver is updating the bind and all other will then use this setting until a new update is triggered
17:58.44codebreakerManxPower: for me it is not a question of money. for me it is the question how to do it myself with opensource not for money. just to learn something(or lern more) :)
17:59.07ManxPowercodebreaker, My point is I cannot help yu further.
17:59.33codebreakeri understand. this was only to clarify my point :)
18:05.13*** join/#asterisk tiCo89 (i=mario@debian.uid0.ch)
18:05.18tiCo89hello
18:06.30tiCo89i would like to setup asterisk on my gateway, it should make a connection to a sip-provider. if there are clients in my lan (e.g kphone) it should ring on all these phones, if not it should go on a mailbox, has somebody a good howto?
18:10.07*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
18:12.20asterboystart here:
18:12.22asterboy~docs
18:12.23jbotdocs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
18:12.33asterboylots of howto
18:13.18codebreakertiCo89: http://pastebin.com/688975
18:13.29ManxPower~thebook
18:13.30jbotsomebody said thebook was Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online!
18:13.31ManxPower~mailinglist
18:13.32jbotSearch Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm
18:15.27*** join/#asterisk timscott (n=a@d198-53-19-216.abhsia.telus.net)
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18:23.48Flosofthow easily is asterisk configurable?
18:24.50russellbit's so easy that it configures itself!
18:24.55russellbta-da!
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18:26.08*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
18:26.24Flosoftwell .. .what about IVR and and ringroups etc?
18:26.48russellbthat's all very common configuration, there are many examples available
18:26.50russellb~docs
18:26.51jbot[docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
18:30.28fileyay nice quiet #asterisk weekend
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18:49.26Hmmhesaysheh
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18:59.09Flosoftcan anyone help me with a VoiceOne installation?
19:00.01salviadudis there a black man in the chan?
19:00.20salviadudi am looking for an african-american, maybe even an african-south-american
19:02.04*** join/#asterisk stoffell_h (n=stoffell@81.83.249.224)
19:02.35sevardwhatchu need bro
19:02.36Flosoftcan anyone help me with a VoiceOne installation?
19:03.40salviadudi need some pointers
19:03.43QwellFlosoft: keep asking every 2 minutes
19:04.04salviadudon how to pretend i'm black
19:04.06salviadudat least on the phone
19:04.18Qwellsalviadud: Just talk normally.
19:04.21sevardsay stuff like shit dawg, and That's Wack
19:04.25Flosoftwell ... I am stuck with this for a week now
19:04.37salviadudi'm gonnna have fun with this
19:04.40Qwellflorz: So ask voiceone
19:05.31salviadudsevard, are you authentically black, or are you just giving me some generic advice man?
19:05.42sevardsalviadud: everyone on the internet is black.
19:06.05salviadudo_O
19:06.25salviadudthat's wack dawg
19:06.32timscottMay I ask why you would want to "sound black" on the phone?
19:06.44sevardthere you go.
19:06.51stoffell_hlol
19:06.58timscottwhiird dawg why u needin' that shit on the tele', yo?
19:07.29sevardtimscott: black people don't belong on the telephone?
19:07.32salviadudi want some hoes :)
19:07.37timscottoh man
19:08.03timscottget off the internet, and go make friends. money plays you'll find more "hoes" that way.
19:08.10salviadudim gonna call all those laquishas
19:08.24sevardsalviadud: what about the La Fan Das
19:08.34sevarderm, Duh
19:08.40timscottla fon da
19:08.41salviadudthey got all kinds of cookie names
19:08.46Qwellenough, now
19:08.58Qwellthis conversation ends immediately
19:09.18salviadudqwell, what's wrong with the hoes?
19:09.32salviadudyou like hoes too right?
19:09.52sevardalright, i'm irish and i love irish jokes, i also love potatoes (does the bride come with potatoes?!) why is it when ever somebody makes fun of me it's cool but when i start picking on black people it's like OH SHIT MAN THE NAZIS ARE HERE.
19:10.13timscottBecause you're white.
19:10.24timscottYou're the prverbial "man."
19:10.27sevardthat's wack dawg
19:10.45timscottI like asterisk.
19:10.51timscottWhat about them thurr asterisks, huh?
19:10.55salviadudi'm mexican
19:11.25salviadudyeah, those asteriks got a big booty
19:19.29*** join/#asterisk cmx_DK (n=dp@cpe.atm2-0-1051183.0x503f8576.kd4nxx11.customer.tele.dk)
19:19.31*** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no)
19:20.06cmx_DKgood afternoon guys :) (in denmark its 9:21pm) :)
19:20.36timscottGood afternoon. :)
19:20.40timscottIt's 1:23 here. :D
19:21.57cmx_DKhehe :) - thats the beauty of the internet :)...
19:22.19cmx_DKi have a  problem with double dtmf tones with sip trunks.
19:22.36timscottWhat do you mean?
19:22.41timscottDTMF is getting sent twice?
19:23.43cmx_DKyup - its not just a standalone problem - we have 100+ asterisk serveres all with the same problem, and not the same hardware, and different asterisk versions
19:24.08timscottyou have 100 asterisk servers?
19:24.12timscottoof
19:24.19cmx_DKi was just wandering if any one else has that problem.
19:24.20timscottwhat distribution?
19:24.31timscottno, i've never had the problem
19:24.37timscottare you using a virtual asterisk setup?
19:24.43timscottlike with Xen or VServers?
19:24.45coppicepoisson or gaussian
19:24.57cmx_DKwe uses CentOS - not the aah - but our homemade iso.
19:25.07cmx_DKnope - not virtual.
19:25.15timscott:/
19:25.26Flosoftcan anyone help me with a VoiceOne installation?
19:25.35fileFlosoft: over and over and over you ask...
19:25.49cmx_DKthe problem is with DISA - atleast the problem is very consistent with DISA.
19:27.06jake1932is VoiceOne an ITSP?
19:28.06Flosoftno
19:28.16FlosoftVoiceOne is an GUI for Asterisk
19:28.16jake1932what is it?
19:28.19jake1932oh
19:28.26*** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-69-209-162-62.dsl.sfldmi.ameritech.net)
19:28.34jake1932i don't think anyone on here supports GUIs for asterisk
19:28.49cmx_DKanyone else that can help regarding the DTMF problem?
19:29.19jake1932does VoiceOne  have a mailing list or forum?
19:30.26Flosoftthey are both quite bad
19:30.29Flosoftthe only thing I have
19:30.30Flosofthttp://www.voiceone.it/download/
19:31.18jake1932the forum seems to have some activity
19:31.29*** join/#asterisk Whisk (n=whisk@82-40-184-22.cable.ubr04.croy.blueyonder.co.uk)
19:32.03Flosoftthe thing is I don't know too much bout asterisk or linux
19:32.08jake1932haha
19:32.12Flosoftso I am a bit lost in the installation procedure
19:32.34jake1932why not start with asterisk - you may find you don't need this GUI
19:32.34Flosoftespecially... Debian isn't a sudo system
19:33.00Flosoftso I would need to change the beginning of the tutorial they have there
19:33.06*** join/#asterisk kaz0358 (n=kurtzogl@ES-189.telecom.ksu.edu)
19:33.10jake1932asterisk itself is very well supported here (including the installation with Debian)
19:33.11Flosoftand I don't know what debian packages I would ned
19:33.31jake1932there's even a page on the wiki on how to install asterisk + debian
19:33.32kaz0358does anyone know if iaxtel is down?
19:34.12VoicePulseFlosoft: How many users is your system intended for?
19:34.40Flosoft20 - 40
19:35.01Flosoftdon't know yet really :S
19:35.36VoicePulseThen I suggest you go with a proven plug-and-play product like Fonality or SwitchVox, it will take at least 6 months to learn enough to get things up and running reliably.
19:35.59jake1932VP - i take issue with that
19:36.19VoicePulseor pay jake1932 to set it up for you :)
19:36.23jake1932at least a few years!
19:36.25jake1932haha
19:36.55jake1932actually - rapid is pretty fast - your results may differ
19:37.05jake1932it's prebuilt debian + asterisk
19:37.24kaz0358could someone that has iax setup dial the echo test number as see if it works? 1-700-999-9613
19:37.31kaz0358err iaxtel
19:37.48russellbkaz0358: it probably doesn't work
19:37.56jake1932http://xorcom.com/rapid/
19:37.57fileiaxtel goes up and down, a lot
19:38.27*** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net)
19:39.06kaz0358russellb and file, ahh okay.. well that might explain why i'm not able to do a iax2 register or dial any 700 numbers even when going out as user guest
19:39.34jake1932Flosoft - although rapid runs on an older asterisk version, it's probably good enough for you to get your hands dirty a bit
19:39.39filewe're mucking with it right now
19:40.02Flosoftthe point is I need to integrate it into an running system
19:40.03paryldoes anyone here use a tdm2400p?  i assumed my zaptel.conf and zapata.conf should be similar to my machine with a tdm400p, but it doesn't seem to be liking my config :\
19:40.08FlosoftI can't change Distro
19:40.21russellbkaz0358: try again now
19:40.34jake1932<PROTECTED>
19:40.37Flosoftyes
19:40.38kaz0358russellb, okay.. trying to re-register
19:41.30russellbthis machine is so weak
19:41.39jake1932Flosoft: ok, try http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Debian
19:41.45paryli get "ERROR[3776]: chan_zap.c:6878 mkintf: Unable to open channel 1: No such device"
19:42.08parylbut ztcfg -vvvv shows 4 channels configured properly
19:42.10kaz0358russellb, okay.. cool. registeration was successful. and i was able to dial the echo test number. thanks! hehe. i was about ready to pull my hair out.
19:42.30filedon't gauge a working Asterisk on calling iaxtel
19:43.08*** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no)
19:43.18kaz0358russellb, there is only *one* machine running the iaxtel stuff? what order of users are normally registered over iax2?
19:43.35file788 right now
19:43.44fileit's 2.4GHz P4
19:43.51russellb512 MB ram
19:43.53filewith 512MB of RAM
19:44.09russellbprobably an emachines or something, lol
19:44.12bsdfreakheya file
19:44.13kaz0358file, thanks.. maybe there should be a short addendum to voip-info.org wiki on that :)
19:44.30filehellllllllo
19:47.55russellbit's still fun to poke at it
19:48.02russellbjust don't expect it to work :)
19:48.10*** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org)
19:48.15russellbit's actually a very useful load test for us
19:48.20parylcan anyone help me configure my tdm2400p?
19:48.28parylit's just not working
19:48.35fileit's better then it was, but there's still one last problem to track down
19:48.36Ariel_russellb, it was great 2 or 3 years ago. for testing asterisk iax2 connections.
19:48.50russellbyeah, we need a new server for it
19:49.19filerussellb: we should put in a request for a new one
19:49.25russellbfile: yes, we should
19:49.33russellbfile: why don't you get on that
19:49.35*** mode/#asterisk [+o file] by russellb
19:49.45fileyou have seniority!
19:49.55russellbbut you're the iaxtel maintainer :-p
19:50.12Qwellooo, bribe
19:50.15fileminor detail
19:50.18Qwellnow, time for blackmail :p
19:50.19russellbzing!
19:50.20jake1932who has the janitor job?
19:50.30russellbjake1932: file
19:50.32jake1932haha
19:50.34filepfft
19:50.38russellbhe has to clean our office every  night
19:50.44Qwellremotely!
19:50.56russellbit's a virtual office ...
19:51.05Qwellyeah
19:51.16QwellYou don't telecommute to your virtual office?
19:51.19russellbthere's a chair on the ceiling, feel free to have a seat.
19:51.29*** join/#asterisk gr0mit (n=guest@extrt.txrx.org.uk)
19:51.49*** join/#asterisk Laureano (n=mdelia@host172046.metrored.net.ar)
19:52.03filewe are not responsible if you fall and hurt yourself
19:52.04LaureanoHello
19:52.22Ariel_but it would be fun to watch
19:53.17kaz0358any idea on the number of extensions accessible on the public dundi context? i'm not for sure if that is the correct wording for it
19:53.54russellbkaz0358: a lot ... but as a percentage of the e164 number space, probably very little
19:55.06kaz0358russellb, i thought that might be the case.
19:55.39russellbi'm not really sure, though
19:55.46russellbi'm not a member anymore
19:56.21kaz0358if you want to participate as a tier 3, does it take much active bandwidth?
19:56.37kaz0358thanks for the info russellb
19:58.27parylhate to ask again... but is there anyone that can help me with my tdm2400p?
19:58.39parylno one is answering any of my questions :\
19:58.46kaz0358paryl, what kind of problem are you having?
19:58.51gr0mitparyl what is the problem?
19:59.25parylwell, i've configured zaptel.conf pretty much identical to my machine with a tdm400p in it... fxsks=1-4
19:59.37paryland ztcfg -vvvv shows 4 channels configured
19:59.45gr0mitmkay....
19:59.52parylbut asterisk gives me "ERROR[3843]: chan_zap.c:6878 mkintf: Unable to open channel 1: No such device"
19:59.55paryland dies
20:00.09parylzapata is configured the same way that it is on the other machine too...
20:00.39parylnone of the docs on digium's site have anything specific to the 2400p, so i wasn't sure if i'm configuring it wrong or not
20:01.06*** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
20:01.08surfduehello!
20:01.17surfduei want a pbx i think this is the best right?
20:01.26surfdue:)
20:01.35luke-jr_...
20:01.38kaz0358surfdue, the best is a high subjective thing :)
20:01.57surfduewell I am looking for a pbx with a web config
20:02.02surfdueim thinking asterisk with amp?
20:02.03luke-jr_kaz0358: not at all; just depends on what you need the best *for*
20:02.08surfdueor do you have a diffrent suggestion for me?
20:02.15luke-jr_surfdue: see the topic
20:02.19surfdueI see it
20:02.27luke-jr_surfdue: specifically regarding AMP
20:02.29parylalso, "WARNING[3885]: chan_zap.c:923 zt_open: Unable to specify channel 1: No such device" and " ERROR[3885]: chan_zap.c:10314 setup_zap: Unable to register channel '1-4'"
20:02.33surfdueare you telling me to not use asterisk?
20:02.42surfduei dont use amp.
20:02.43luke-jr_surfdue: telling you not to use AMP, sure
20:02.45surfdueand i dont know what that is?
20:02.50luke-jr_...
20:02.56surfdueis there a web config utility?
20:02.59luke-jr_you just fscking said "im thinking asterisk with amp"
20:03.03surfduein astirisk
20:03.08luke-jr_no
20:03.15surfduei was reading the topic when i read that sorry
20:03.16kaz0358paryl, do a zap show status
20:03.22surfduei meant i need asterisk with web config
20:03.28Qwell"need"?
20:03.43parylkaz: i can't, asterisk dies on that error
20:03.47surfdue:|
20:03.48gr0mitkaz0358 i dont think paryl  can get it running
20:03.49luke-jr_well, I don't think anyone here wants to play with people using web configs
20:03.52kaz0358surfdue, no i'm not telling you not to use it. i've never used any external web interface to configure or setup asterisk.. so i can't really comment on that aspect of it
20:04.10surfduekaz0358, how should i configure it
20:04.20gr0mitsurfdue : you are asking for trouble trying to use a web interface
20:04.21*** join/#asterisk ToTo (n=ToTo@host38-109.pool8258.interbusiness.it)
20:04.23luke-jr_surfdue: configuration files
20:04.31surfdueis it easy to write them?
20:04.46kaz0358paryl, ahh.. well i haven't played with either the 4 or 24 cards, but i have messed around with the 2 port t1/e1 card and i ran into the same problem for a while. it turned out it was a problem with the configuration files
20:04.49*** join/#asterisk Assid (n=assid@203.115.64.12)
20:04.49jake1932not at first
20:05.00jake1932but you'll get it - they're plenty of samples
20:05.01luke-jr_surfdue: sure
20:05.06gr0mitsurfdue you can get a call in and out with about 5 lines in extensions.conf
20:05.11Assidokay something wrong with my phone.. it keeps losing ticks for the time
20:05.16parylkaz: what was the problem with the config files, exactly?
20:05.19paryldo you remmeber?
20:05.21luke-jr_gr0mit: extensions.conf is crufty ;)
20:05.36gr0mitstart with something simple!
20:05.40gr0mitcrufty?
20:05.49jake1932oohg - new word!
20:05.59gr0mitAEL ????
20:06.04jake1932crufty
20:06.14Assidit just slows down
20:06.23Assidanyone know how ot fix it?
20:06.26kaz0358paryl, well.. there were several problems.. but one in particular was that i was specifying a logical port on the spanmap in the zapata.conf file. i never did figure out what the deal was with that, but simply removing the 3rd option made things work.
20:07.04luke-jr_gr0mit: 2000 => { Dial(SIP/2000); switch(${DIALSTATUS}) { case BUSY: Busy(); break; default: Playback(wtf); Hangup(); }; };
20:08.05gr0mitwhat is this, luke-jr_ ???
20:08.15kaz0358paryl, but that didn't stop asterisk from firing up. not specifying how the channels should be setup did cause problems.. or trying to specify something conflicting for the channels in both zaptel.conf and zapta.conf
20:08.23luke-jr_gr0mit: Asterisk Extension Language
20:08.31luke-jr_gr0mit: extensions.ael
20:08.35gr0mithmmmm.  is this something recent?
20:08.39luke-jr_1.2
20:08.43gr0mitmkay.
20:08.56luke-jr_tho buggy until next major release
20:09.05luke-jr_cuz russellb doesn't want to fix bugs in 1.2's implementation
20:09.23surfdueis there a fedora core 4 install guide for asterisk?
20:09.26gr0mitvery low WAF
20:09.36luke-jr_well, just make sure your switch()s have a default: and fallthrus have at least one statement before the fall
20:09.48luke-jr_the other bugs are mostly gone in latest 1.2
20:09.50snitt\o/
20:09.56luke-jr_oh, and don't use 'continue;'
20:10.05luke-jr_at least not in a for loop
20:10.25parylok... if you don't have anything in zapata.conf, should asterisk still recognize the zap card?
20:10.40gr0mitparyl, no!
20:11.03parylbut it does!  i mean, it doesn't configure the channels, but 'zap show status' shows me the card
20:11.08gr0mityou need to define the channels in zapata.conf
20:11.23gr0mityes but you will not be able to make any calls
20:11.30luke-jr_gr0mit: http://www.voip-info.org/wiki-Asterisk+AEL
20:11.46gr0mittvm, luke_jr_
20:12.20luke-jr_tvm?
20:12.27parylok, with just:
20:12.28paryl[channels]
20:12.30parylsignalling = fxs_ks
20:12.31parylchannel => 1-4
20:12.33paryli get the error
20:12.43parylthat's all that's in zapata.conf
20:13.17gr0mitluke-jr_ tvm = Thanks Very Much or Ta very much if, like me, you are in the UK
20:13.44luke-jr_ah ok
20:13.45luke-jr_np
20:14.16gr0mitparyl, you need a bit more than this in zapata.conf, i think
20:14.27luke-jr_append a '2' to the wiki link to get info on the new version in HEAD
20:14.41parylgr0mit: well, yeah, but should that be enough to kill asterisk?
20:15.01gr0mitbut i speak from a position of ignorance as I have only used the ultra-crappy X100p and lots of BRI and E1 cards
20:15.31parylis there anything else i can use to verify the card is working properly?
20:15.36gr0mitdumb quesstion paryl but you did compile and install zaptel....?
20:15.50parylyes, just compiled it fresh
20:15.53*** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
20:16.28gr0mitplease paste your zaptel.conf and zapata.conf into www.pastebin.ca
20:17.29parylit's not big, three lines:  fxsks=1-4, defaultzone=us, loadzone=us
20:17.38gr0mitalso paryl please paste the output of ztcfg -vv
20:17.44*** join/#asterisk MikeJ[Laptop] (n=vircuser@adsl-69-209-183-91.dsl.sfldmi.ameritech.net)
20:18.30sevardhe got sick of your shit.
20:18.42*** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net)
20:18.50parylgrr... got kicked out
20:20.03gr0mitok paryl paste paste paste
20:21.25parylgr0mit: http://pastebin.ca/52260
20:21.42gr0mitgot it
20:22.47gr0mityou want to remove switchtype - national
20:22.58gr0mitthis only referrs to PRI/ISDN lines
20:22.59FlosoftI can't start asterisk although I did everything Voiceone told me to do
20:23.14Flosoftdebian:/var/www/vhosts/debian.FLOSOFT-NET/httpdocs# /etc/init.d/asterisk start
20:23.14FlosoftAsterisk not yet configured. Edit /etc/default/asterisk first.
20:23.22Flosoftthat is what it says :S
20:23.26parylgr0mit: ok, done, but still the same error
20:23.28ManxPowerFlosoft, That is not an Asterisk message.
20:23.44Flosofthttp://www.voiceone.it/download/
20:23.45Flosoftall done :S
20:24.01gr0mithmph please paste the error you are getting to pastebin, with a bit of context above the error
20:24.09ManxPoweranother one for the ignore lists.
20:25.12parylgr0mit: http://pastebin.ca/52262
20:25.36gr0mitgot it
20:25.54blitzragenrw: I'd like to prevent the phone from performing the redirect (I know -- my reply has several hours latency :))
20:26.24gr0mitwell it looks like it cant load chan_zap
20:27.16parylriiiight?  :)
20:27.34gr0mitparyl, 2 secs....
20:28.40gr0mitpls check that chan_zap.so is actually in /usr/lib/asterisk/modules directory
20:29.03gr0mitif not, then looks like zaptel did not compile and install correctly
20:29.38codebreakeralways when a phone registers can i asterisk let run a script/externalcommand?
20:30.02parylgr0mit: yes, it's in there
20:30.08gr0mitho hum
20:31.41parylit looks to me that it's something to do with either the configuration or the drivers...
20:31.51paryli compiled from the 1.2.5 tarball on asterisk.org
20:32.09gr0mitdrivers are fine. if ztcfg says it has configured the channels then it is telling the truth.
20:32.13paryland compiled libpri also, though i don't know if it's needed with the tdm cards
20:32.26parylso... it must be the config?
20:34.15blitzrage1.2.5?
20:34.19blitzrage1.2.7.1 is out :)
20:36.17parylno, i'm talking about zaptel-1.2.5
20:36.28blitzrageoh I see -- I gotta try and keep up
20:36.38gr0mitwell i am somewhat stumped, paryl!
20:37.24paryli just don't get it
20:38.25gr0mitoh wait!
20:38.35gr0mitdoh!
20:38.41paryl??
20:39.16gr0miti think you should remove all the => from zapata.conf and only use =
20:39.41*** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
20:39.42surfduehello
20:39.54surfduei am installing astrisk it needs to be run as the same user as apache
20:39.59gr0miti am not sure that => is valid in zapata.conf, only extensions.conf
20:40.06surfduebut im on a shared server with cpanel so i cant make apache run as asterisk what do i do?
20:40.12gr0mitplease try this
20:40.58parylgr0mit: no dice :(
20:41.00gr0mitunless anyone else can assure me that => is valid
20:41.03gr0mithmmm
20:41.11gr0mitwell it was worth a try
20:44.08gr0mitok, can you just start with the context signalling and channel lines in zapata.conf ?
20:44.12paryli use => on both of my other servers with no problems
20:44.17hads|homegr0mit: In zapata.conf = is for assigning variables and => is for creating objects, i.e channels.
20:44.19gr0mitmkay
20:44.50gr0mithads|home, yes, you are right
20:45.11gr0mitbut callprogress => no is not correct
20:45.40hads|homeYou are correct, it should only be used for the channel entries.
20:45.43gr0mitthe only place to put a => is the channels line at the bottom
20:45.57gr0mitpls try this, paryl
20:46.05parylgr0mit: i've shortened zapata.conf to:
20:46.06paryl[channels]
20:46.08parylsignalling = fxs_ks
20:46.09parylcontext = incoming
20:46.11parylchannel => 1-4
20:46.31gr0mitand...?
20:47.26*** join/#asterisk mog_home (n=achika54@68.62.237.103)
20:47.31parylsame error
20:47.40hads|homelooks fine. For a single fxo port right?
20:47.50paryli also tried channel => 1
20:48.13parylhads: 4 ports
20:49.08hads|homeAh, yes sorry, Just woke up :)
20:49.24parylthese will plug into phone lines from the phone company... that does mean i need to use fxs, correct?
20:49.38gr0mityes, paryl
20:49.57gr0mitthe boards are FXO but you need fxs signalling
20:50.11parylright... just making sure :)
20:50.33gr0mitcan you do a cat /proc/zaptel/* and paste for me
20:52.06parylhttp://pastebin.ca/52269
20:53.37gr0mitok thanks
20:53.52*** join/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
20:53.56gr0mitlooks ok to me
20:55.03*** part/#asterisk Laureano (n=mdelia@host172046.metrored.net.ar)
20:55.21parylok, this is just crazy
20:55.29parylhow could it not work?
20:55.33*** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
20:55.36gr0mitwell i hate to say this paryl but without sshing into your box to look, I really am rather puzzled
20:55.57gr0mitand as it is bedtime here i am not about to offer to do that!
20:56.09parylhaha, well, i do appreciate your trying
20:56.44gr0miti managed to get ghost99 to the point where he was able to make a call yesterday, so i was happy
20:57.07gr0mitand so was he, despite ManxPower's unhelpful comments
20:58.16gr0mithope you find someone in your timezone to get you running, paryl.
20:58.37parylalright man, thanks again
20:58.37gr0mitAlso, don't forget that Digium will get you to the point that the cards run
20:58.50parylbut they aren't open on the weekends are they?
20:58.52gr0mitthis is part of what you pay them for the cards
20:59.05gr0mitno, they are not open at the weekend!
20:59.18parylstupid non-workaholics ;)
20:59.56gr0mittraditional British habit, that
21:00.52*** part/#asterisk gr0mit (n=guest@extrt.txrx.org.uk)
21:05.00tparcinadeam, if digium was from dalmatia, they wouldn't work even during weekdays :)
21:05.49*** join/#asterisk Assid (n=assid@203.115.64.12)
21:07.04parylhaha
21:07.34Assidi cant seem to access the call log list
21:07.42Assidon polycom 301
21:07.54Assidnvm
21:07.58Assidgotta do it through menu
21:09.12*** join/#asterisk websae (n=icechat5@h69-129-251-26.69-129.unk.tds.net)
21:09.32*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
21:09.49*** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk)
21:10.00websaesure is quiet out there again :)
21:10.40robin_szprobably because people are busy
21:10.55tekatiShhhh...
21:11.03robin_szbusy cutting up their GXP200s, digging deep, lime filled pits, and burying them
21:11.25websaebusy hehehe
21:11.26websaei hear ya on that one
21:11.38websaeyou don't like your Grandstream?
21:13.04timscottI don't mind my GXP2000.
21:13.11websaeI like mine :)
21:13.15websaeit works well
21:13.25websaeexcept the speakerphone, but it has other good qualities :)
21:13.25tekatiI don't have one.  But am looking for a good IP phone is that a recommended phone?
21:13.35tekatiAh I need a good speakerphone.
21:13.39tekatiWho has a good one?
21:13.41websaeand once you put some clay in the headset so it hangsup everytime, it works well
21:13.47websaePolycome  501
21:13.52websae$200ish
21:13.53BadPacketpolycom
21:13.56BadPacket(obviously)
21:14.01websaeor cisco :)
21:14.17BadPacketyeah, the cisco 79xx have good speakerphones too
21:14.33websaevery good speakerphones
21:14.38websaeeverything cisco is good :)
21:14.43websaequality is there
21:15.16timscott$$$++
21:15.18timscottfor cisco
21:15.24websaeyeah!
21:15.39websaebut if it's quality you need.........it's quality they have :)---people pay for quality
21:15.41Assidcisco costs wayyy too much
21:15.46Assidpolycoms got the best value for money
21:15.47websaejust like in termination and origination services
21:16.03BadPacketwebsae: exactly
21:16.20ManxPowerPolycoms can be gotten amazingly cheap and you don't have to by the power supply and you don't have to buy the firmware.
21:16.22websaeBadPacket: i sense you have experience that feeling
21:16.30BadPacketdefinitely
21:16.36BadPacketI've tried grandstream and nufone
21:16.46BadPacketand now I use a Cisco 7960 and voicepulse
21:16.56ManxPowerBadPacket, and you are still in therapy because of the grandstream?
21:16.59websaeNuFone----one word....yuck
21:17.07BadPacketManxPower: hah
21:17.18BadPacketmore because of nufone
21:17.24ManxPowerNufone was good until their recent...issues.
21:17.38tekatiBut is the speakerphone part of the Polycoms good.
21:17.40BadPackets/issues/MAJOR issues/
21:17.41Assidcurrently im using poly301.. with sipdiscount for personal use.. and another line configured to my box's extension
21:17.42websaeI use to always have issues with them
21:17.45websae*ALWAYS
21:18.13BadPacketthe polycom speakerphone is even better than the 7960, but the polycom phone is such a pain in the ass to configure
21:18.22Assidnot really
21:18.30Assidits one of the fastest ive configured
21:18.31ManxPowerHuh?  Basic config can be done via the web interface.
21:18.37Assidweb interface is easy
21:18.46Assidonly pain in polycom is.. the wait time for reboots
21:18.52ManxPoweradvanced config can be....challanging.
21:19.00BadPacketthe built-in web server is buggy and slow... maybe I need newer firmware if you guys say it's good
21:19.11ManxPowerAssid, Yeah, and in production enviroments you dno't normally reboot the phone
21:19.15Assidyeah
21:19.23ManxPowerBadPacket, the built in web server IS buggy and slow.
21:19.25Assidnormally.. you provision the phone via ftp/tftp
21:19.41BadPacketthat's what I meant by it being a pain in the ass
21:19.50Assidi still wanna get this ringtone into my phone dammit..  just cant get it
21:20.00BadPacketAssid: yes, but I was testing stuff, so I was changing a bunch of stuff via the web interface
21:20.12Assidweb interface frankly sucks
21:20.28robin_szwebsae, sorry was AFK
21:20.28Assidyouu change one thing.. and if you wanna access some other menu.. you gotta save it
21:20.31Assidthats more time gone
21:20.36websaeohh okay
21:20.39robin_szwebsae, I dont know if I liek mine or not ...
21:20.46websaedid you just get it?
21:20.49robin_szits hard to tell if its any good
21:20.56robin_szno had it 8 months
21:21.05BadPacketAssid: yeah - the sipura/linksys are better for that
21:21.09Assidyep
21:21.20robin_szbut I accidentally "upgraded" to the beta firmware and cant go back
21:21.21Assidthats why i live provisionng
21:21.27Assidyou change everything you want to
21:21.29Assidand then reboot
21:21.36robin_szdisplay has been fscked ever since
21:22.18*** join/#asterisk TripleFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
21:22.23TripleFFFFFFFFFhi all.
21:22.33timscotthello.
21:22.36TripleFFFFFFFFFim wondering why latest svn doent compile with mysql on freebsd
21:22.41TripleFFFFFFFFFanyone have a hint ?
21:23.01tekatiSo between the Polycom 301 and the 501 is the 501 the way to go?
21:23.13tekatiAnd if so what are the main reasons?
21:23.19Assidokay someone try and get this wav file running for me.. i just cant get it
21:23.24Assidnnot on the 301 atleast
21:23.48Assidtekati: 3 line phone .. bigger screen.. full duplex speaker
21:24.18robin_szTripleFFFFFFFFF, did you compile it?
21:24.30TripleFFFFFFFFFwell im in ports and making as we speak
21:24.39TripleFFFFFFFFFas the /usr/src/asterisk-addons is not working
21:24.48tekatiFull duplex speaker works for me.  Enough right there.  I have more conference calls then I care to talk about.  Speaker phone on my Panasonic MultiTalk8 over a PAP2 just does not do to well all the time.
21:25.09TripleFFFFFFFFFvoice# gmake
21:25.09TripleFFFFFFFFF./mkdep -fPIC -I../asterisk -D_GNU_SOURCE   -I/usr/local/include/mysql  `ls *.c`
21:25.09TripleFFFFFFFFFapp_addon_sql_mysql.c:25:27: asterisk/file.h: No such file or directory
21:25.09TripleFFFFFFFFFapp_addon_sql_mysql.c:26:29: asterisk/logger.h: No such file or directory
21:25.14TripleFFFFFFFFFlots of that
21:25.23websaeyou sure have quite a few "F"s in your nick there...
21:25.33NuggetHe's whiter than white!
21:25.39TripleFFFFFFFFFyeah
21:25.48TripleFFFFFFFFFcan never remember my pass
21:25.51TripleFFFFFFFFFso i add a F
21:26.03TripleFFFFFFFFFso im doing the ports one
21:26.12TripleFFFFFFFFFjust hope it wont pbreak my asterisk
21:26.22Assidokay so anyone wanna volunteer in seeing why this wav file just refuses to load up on poly301
21:27.20TripleFFFFFFFFFdo i need a flag for mysql on ports for freebsd ?
21:27.20X-RobTripleFFFFFFFFF, you need to do a 'make install' in asterisk before you install addons
21:27.22paryloh my word... i just figured out the solution to my problem
21:27.23TripleFFFFFFFFFoh well
21:27.24TripleFFFFFFFFFlol
21:27.37parylis anyone around that was helping me before?
21:28.18*** part/#asterisk tiCo89 (i=mario@debian.uid0.ch)
21:28.46parylit seems the tdm2400p comes default with the moules installed backwards... so my one fxs module was in port 6, which meant i had to reference channels 21-24, rather than 1-4... sigh
21:29.40TripleFFFFFFFFFbtw even when i got latest diium source i cant compile now
21:29.40X-RobTripleFFFFFFFFF, you need to do a 'make install' in asterisk before you install addons
21:29.42Assidwww.pienotech.com/ctu.wav .. i am trying to get this to work on polycom 301
21:30.05TripleFFFFFFFFFyeah
21:30.07mutilatoryea
21:30.09TripleFFFFFFFFFasterisk is already running
21:30.12TripleFFFFFFFFFfyi
21:30.30tekatiWho has the best deal on the Polycom 501s?
21:30.34nrwctu.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 44100 Hz
21:30.35tzangervoipsupply likely
21:30.43nrwit has to be a 8khz 8 bit file
21:30.48nrwthats why its not working
21:31.03Assidnrw: i tried .. sox ctu.wav -c1 -r8000  ......
21:31.07Assiddidnt work..
21:31.14Assidthen i tried with -r16000 didnt work
21:31.25TripleFFFFFFFFFlatest svn
21:31.26TripleFFFFFFFFFgmake[1]: Entering directory `/usr/src/asterisk/apps'
21:31.27TripleFFFFFFFFFMakefile:16: *** missing separator.  Stop.
21:31.27TripleFFFFFFFFFgmake[1]: Leaving directory `/usr/src/asterisk/apps'
21:31.31TripleFFFFFFFFFlol
21:32.15TripleFFFFFFFFF<<<<<<< .mine
21:32.17TripleFFFFFFFFFon this line
21:32.20TripleFFFFFFFFFin apps/Makefile
21:32.47Assidnrw?
21:33.28*** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no)
21:33.51TripleFFFFFFFFF<<<<<<< .mine
21:33.56TripleFFFFFFFFFwhat are these in the makefiles ?
21:34.05russellbTripleFFFFFFFFF: svn revert -R .
21:34.12TripleFFFFFFFFF??
21:34.34TripleFFFFFFFFFwow
21:34.36TripleFFFFFFFFFrusse
21:34.38*** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no)
21:34.39robin_szugh ... over a month since the last (broken) gxp 2000 release, and still no sign of a fix .. im beginning to wonder if they are using the infinite-number-of-minkeys method of development
21:34.43TripleFFFFFFFFFyou da man.. so what did that do ?
21:34.47russellbTripleFFFFFFFFF: :D
21:34.48robin_szseldom have I seen such slow progress
21:35.00X-Robrobin_sz, sure you have.
21:35.06russellbTripleFFFFFFFFF: you had a conflict when you updated for some reason, so that command reverted local changes.
21:35.06TripleFFFFFFFFFit just reverted my svn to old version ?
21:35.07X-Robthink of, uh..
21:35.08X-Robum
21:35.13TripleFFFFFFFFFoh
21:35.14X-RobOK, no, you win.
21:35.18X-RobIt's crap.
21:35.19russellbTripleFFFFFFFFF: earlier something was removed and then added back, so that probably caused it
21:36.02TripleFFFFFFFFFso from svn i say nothing more to make resmysql ?
21:36.13X-RobTripleFFFFFFFFF, you need to do a 'make install' in asterisk before you install addons
21:36.16TripleFFFFFFFFFjust install base.. then go addons and make mysql then remake base ?
21:36.38nrwtriplef: I am curious why you dont just install the asterisk and asterisk-addons ports.
21:36.44TripleFFFFFFFFFoh
21:36.47nrwinstead of trying to compile the latest svn
21:36.49TripleFFFFFFFFFtheres asterisk-addons port lol
21:36.53*** join/#asterisk Abydos313 (i=abydos31@ppp-71-133-210-73.dsl.irvnca.pacbell.net)
21:37.00TripleFFFFFFFFFok doing
21:37.22TripleFFFFFFFFF./usr/ports/net/asterisk-addons
21:37.23TripleFFFFFFFFFtrue
21:37.25TripleFFFFFFFFFok
21:37.28TripleFFFFFFFFFthat good then
21:37.28robin_szX-Rob, where?
21:37.31TripleFFFFFFFFFlol
21:37.36robin_szok I win >
21:37.38robin_sz:)
21:37.58TripleFFFFFFFFFwow with h323 also .. nice.. now i can code my video phone
21:38.09TripleFFFFFFFFFanyone got xten for pocket pc to work ?
21:38.16Assidnrw: any luck?
21:39.03nrwassid: im not trying. I just wanted to make sure you changed its format. I'm at home i couldnt tell you if it worked even if i did do it
21:39.06X-RobIF any of the *bsd'ers would like to do a freepbx install documentation, I'd be most appreciative. It's very much centos based at the moment
21:39.22X-Robbut there's no reason why it wouldn't work on any of the bsds
21:39.49robin_szI am willing to wrote up the debian install documentation if you want
21:39.59fourcheezecan the MYSQL command be used with a persistent connection somehow
21:40.01robin_szok ... here goes ...
21:40.11tekatiAnyone have a polycom 501 or cisco 79XX phone they want to trade for a TDM40B?
21:40.15robin_sz"apt-get install asterisk"
21:40.22robin_szthere, how did I do?
21:40.35X-Robrobin_sz, well that's a start.
21:40.39fourcheezerobin_sz: you missed out "apt-get update"
21:40.40Assidnrw: how did you get the info?
21:40.48nrwassid: the file command
21:41.01robin_szdang, me and my partial documentation
21:41.10X-Robhttp://aussievoip.com.au/wiki/freePBX-Debian
21:41.30Assidctu2.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 16000 Hz -- that doesntwork
21:41.47russellbfourcheeze: yeah, use source and checkinstall
21:41.57robin_szalways remember to apt-get remove before soing a make install ....
21:42.04robin_szor Bad Things happen
21:42.17fourcheezerussellb: what's checkinstall ?
21:42.22*** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com)
21:42.31nrwassid: and there you have your answer to why its not working
21:42.32russellbit runs make install but keeps track of what gets installed
21:42.42russellbso you can later remove it, just like you had installed it with apt
21:43.02russellbit's pretty nifty
21:43.36Assidnrw: doc says L16/160008 (16-bit, 16 kHz sampling rate, mono) is okay
21:43.42fourcheezerussellb: sounds cool
21:43.47fourcheezeI keep meaning to work out "stow"
21:43.52Assidwhen i tried 8khz .. it didnt work either
21:53.41TripleFFFFFFFFFwow
21:53.43TripleFFFFFFFFFhttp://brands.xten.net/x-lite/download/X-Lite_CE_Install.exe
21:53.45TripleFFFFFFFFF;)
21:53.54TripleFFFFFFFFFfreebie
21:54.01TripleFFFFFFFFFpocket pc
21:54.09TripleFFFFFFFFFsip .. ill try with bluettoth to my asterisk and see
21:57.28Assidaargh.. work you crazy pos
21:58.09codebreakerdo i need zaptel complete or only the zaptel patch if i want to run asterisk without any cards?
22:06.56*** join/#asterisk A-Tuin (n=a-tuin@steves.ip.v4.me.uk)
22:07.06*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
22:10.38*** join/#asterisk RoyK (n=roy@ti211210a080-0268.bb.online.no)
22:12.53Dr-Linuxhi gays
22:12.55Dr-Linuxguys
22:15.33Kattyherro.
22:24.08tainted-Dr-Linux freudian slip?
22:25.22Dr-Linuxtainted-: what's freudian?
22:25.41timscottdoing something accidental on purpose.
22:26.07Dr-Linux:S
22:26.09Dr-Linuxand slip?
22:26.19timscottyou slipped up, make a mistake
22:26.29timscotta freudian slip is a mistake that you meant to make
22:26.33asterboyyour subconcious thoughts are your true intentions
22:26.35Dr-Linuxtimscott: you mean i said "hi gays" ?
22:26.39timscottaye
22:26.44asterboybrokeback asterisk
22:27.14Dr-Linuxi c
22:27.29Dr-Linuxif someone took my mistake on his heart, i'm sorry for him
22:29.23*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
22:38.11*** part/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex)
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22:53.10*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
23:00.39blitzragelol
23:00.58websae[A]blitzrage: how arie ayou doing? how's that support ticket going?
23:02.32blitzragewebsae[A]: is it still not resolved? not sure - unfortunately its out of my hands
23:03.04blitzrageoh I just tried it -- works for me
23:10.10*** join/#asterisk b4ka (i=WinNT@200-127-198-118.cab.prima.net.ar)
23:14.41*** join/#asterisk OloBola (n=not@netblock-68-183-67-158.dslextreme.com)
23:17.17*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net)
23:20.56*** join/#asterisk Mnabil (n=Mnabil@196.205.192.21)
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23:40.18timscott:)
23:43.49*** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net)
23:45.07*** join/#asterisk TripleFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
23:45.37TripleFFFFFFFFF[func_logic.so] => (Logical dialplan functions)
23:45.37TripleFFFFFFFFFApr 29 19:45:17 ERROR[5497]: pbx.c:1333 ast_custom_function_register: Function ISNULL already registered.
23:45.37TripleFFFFFFFFFApr 29 19:45:17 ERROR[5497]: pbx.c:1333 ast_custom_function_register: Function SET already registered.
23:45.37TripleFFFFFFFFFApr 29 19:45:17 ERROR[5497]: pbx.c:1333 ast_custom_function_register: Function EXISTS already registered.
23:45.41TripleFFFFFFFFFOuch ... error while writing audio data: : Broken pipe
23:45.43TripleFFFFFFFFFok
23:45.46TripleFFFFFFFFFthat latest port
23:45.49TripleFFFFFFFFFanyidea ?
23:46.24TripleFFFFFFFFFneed reboot
23:46.24TripleFFFFFFFFFbrb
23:46.27*** part/#asterisk TripleFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
23:47.19Ariel_it's so slow here tonight.....

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