IRC log for #asterisk on 20090612

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00:48.12*** join/#asterisk neoTheCat (n=agrossma@68.80.60.63)
00:50.47neoTheCathello.  i am a total newbie here.  i need to create a bulk call generator to do some TDM<>SIP and SIP<>SIP testing.  can i sue asterisx to do that?
00:51.03neoTheCats/asterisx/asterisk/
00:53.41mmlj4there's a stress-test app, i don't know the name of it
00:54.20theharsipp
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00:55.03theharhttp://sipp.sourceforge.net/
00:55.37neoTheCatthehar: thanks.  i am trying to see if i can ssip can send actual voice messages so i can get PESQ/MOS scores
00:57.10theharyou can also generate outbound calls with .call files
01:00.35neoTheCatthehar: thanks.  i am pretty new to voip, so i have a bit of research to do, but at least i know where to start.  thanks
01:02.07theharnp
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01:42.52Aiatekwell see you tomorrow people i need to rest
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02:12.30kb3ienhas there been any progress with the parkinglot to make it multi-tennant friendly; or do all parked calls co-mingle?
02:18.54neoTheCatsince everybody here was very helpful, i have another question. does anybody know any software to generate bulk TDM calls (using dialogic borads and asterisk)?
02:21.45eppigygnu dialer?
02:22.45neoTheCati am forbidden from going onto gnu dialers website... :(
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02:27.08jeff"gnud" sounds too much like "nude" to some censorware somewhere? ;)
02:30.07bkw_what is gnud?
02:30.19bkw_I bet you pronounce it like nude
02:30.37bkw_neoTheCat: you're making the rounds
02:31.57neoTheCatbkw_: what do you mean?
02:32.18bkw_neoTheCat: you were just in #freeswitch asking the exact same question :P
02:32.21neoTheCatbkw_: yeah, i am on freeswitch also.
02:32.23neoTheCat:)
02:33.21neoTheCatnot the same exact question.  i replaced "asterisk" with "freeswitch"....
02:38.55bkw_let me translate "how can I annoy the shit out of A LOT of people as fast as possible"
02:39.06bkw_:P
02:42.05*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
02:55.04ectospasmhmmm, I need to get DTMF CID working in Saudi Arabia, but I can't find the way they do DTMF documented anywhere.
02:55.30ectospasmAny ideas on how I should go about searching for it?  Searching ITU hasn't given me any hints.
02:55.51ectospasms/DTMF d/CID d/
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03:02.44bkw_https://issues.asterisk.org/view.php?id=7787
03:03.34bkw_2006-08-23 03:28
03:03.36bkw_thats rather old
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03:11.57ectospasmI was looking at https://issues.asterisk.org/view.php?id=9096
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03:35.31Hydranthey all... I've installed asterisknow, and I'm not sure how to setup my polycom phones
03:35.48Hydrantwith trixbox the phones would boot off the server, but I'm not sure how it works with this
03:39.28jplankis dahdi's system.conf the same as zaptel.conf?
03:41.27ectospasmjplank: yes, for the most part
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03:41.58ectospasmwhat's new in system.conf is that you select software echo cancelers
03:42.03jplanklike could I just do mv /etc/zaptel.conf /etc/dadhi/system.conf ?
03:42.07ectospasmjplank: yes
03:42.10jplankthanks
03:42.13*** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58)
03:42.58ectospasmnote that if you're using a software echocan you MUST set it in system.conf, jplank
03:43.06jplankits hardware
03:43.14jplankthats actually why I'm moving to dahdi
03:43.32jplankhaving issues with hardware echo can that digium told me is fixed in dahdi 2.2.0
03:43.42ectospasmOK, then you don't specify that other than telling Asterisk that there's echocancellation on the channel(s)
03:43.57ectospasmyeah, try 2.2.0-rc5+2.2.0-rc3
03:44.05jplankis the software echocan any good in dahdi?
03:44.13jplank(compiling that right now)
03:44.16ectospasmit's OK, but the hardware is better.
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03:48.10jplankgrrrrr
03:50.05jplankthis can't be good
03:50.15jplankwctdm24xxp:  FATAL: Error inserting wctdm24xxp (/lib/modules/2.6.18-53.1.4.el5/extra/zaptel/wctdm24xxp/wctdm24xxp.ko): Unknown symbol in module, or unknown parameter (see dmesg)
03:50.34astruttRUN!
03:50.47jplank?
03:51.34*** join/#asterisk mike-ekim (n=digiport@204.13.2.30)
03:52.05mike-ekimHi, I am having errors using dahdi - wct4xxp driver, it says [29326.413515] wct4xxp: Unknown parameter `opermode' in DMESG.
03:52.37mike-ekimThe other drivers seem to load fine: wct4xxp: error   wcfxo: done   wctdm: done   wcb4xxp: done   wctc4xxp: done
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03:53.04ectospasmmike-ekim: opermode is an invalid parameter for wct4xxp
03:53.13ectospasmdon't set it in /etc/modprobe.d/dahdi
03:53.16mike-ekimoh, where is it setting that?
03:53.22mike-ekimoh
03:53.45mike-ekimeverythings commented in /etc/modprobe.d/dahdi
03:54.00ectospasmyou may be setting it somewhere else
03:54.13mike-ekimah!
03:54.15mike-ekimmodprobe.conf
03:54.23ectospasmOK
03:54.25mike-ekimCurrently: options wct4xxp opermode=USA vpmnlptype=4 vpmnlpthresh=24 vpmnlpmaxsupp=24
03:54.38ectospasmopermode is invalid for wct4xxp
03:54.40mike-ekimdo I remove everything or just opermode
03:55.01ectospasmthe vpm* stuff you shouldn't need
03:55.37mike-ekimgot it
03:55.38mike-ekimthanks!
03:56.01ectospasmjust FYI, opermode=USA is invalid by itself.
03:56.16ectospasmIf wct4xxp had the parameter opermode, it'd need to be FCC, not USA
03:57.13*** part/#asterisk lmsteffan (n=laurent@reef.ac-noumea.nc)
03:58.58Qwellectospasm: besides fcc, do you know off the top of your head what is valid?
03:59.11*** join/#asterisk sergey (n=sergey@sergey-home.iks.ru)
04:00.25ectospasmI know most countries are of the form COUNTRY, like opermode=INDIA, opermode=SAUDIARABIA, etc.
04:00.48Qwellhmm, k
04:01.03ectospasmQwell: it's on the Digium Wiki under "DriverOptions"
04:01.05Qwellthere's probably no reason for USA, US, and UNITEDSTATES to not be valid
04:01.15ectospasmprobably not
04:01.16QwellI mean, besides that they were just never added
04:01.23ectospasmAnd I may be wrong that they're invalid
04:01.49jplankI just compiled dahdi, and I can see it removed zaptel
04:02.02jplankyet when I try and start dahdi I get wctdm24xxp:  FATAL: Error inserting wctdm24xxp (/lib/modules/2.6.18-53.1.4.el5/extra/zaptel/wctdm24xxp/wctdm24xxp.ko): Unknown symbol in module, or unknown parameter (see dmesg)
04:02.26Qwelljplank: depmod
04:03.19jplank-a?
04:03.24Qwellsure
04:03.51jplankstill not loading the module
04:04.02Qwelldid you install dahdi-linux also?
04:04.11jplankthats what I installed first
04:04.23jplankuseing 2.2.0-rc5
04:04.35jplankspecifically dahdi-linux-complete-2.2.0-rc5+2.2.0-rc3
04:04.49ectospasmyou should be able to remove /lib/modules/$(uname -r)/extra/zaptel
04:05.28jplankI should just nuke that directory?
04:05.37ectospasmI would
04:05.59ectospasmyou can always rebuild your zaptel stuff if you need to.  You still have the Zaptel source lying around, don't you?
04:06.06jplanksure
04:06.14jplanki do
04:06.22jplankremoved
04:06.29jplankshould I recompile dahdi or just restart it?
04:06.40ectospasmshould be able to just restart it
04:06.46QwellYou should upgrade your distro BTW...
04:06.51jplankI can't
04:06.57Qwell2.6.18-53 is like CentOS 5.1
04:07.01jplankyea
04:07.11jplankanything new then that causes a kernel panic
04:07.11QwellThere are many severe security vulns.
04:07.12ectospasm...maybe you can get away with "make install" in the dahdi-linux source tree
04:07.31jplank(fyi the module is still failing)
04:07.40Qwelljplank: Have you tried 5.3?  I assume you have an r8189 NIC?
04:07.49ectospasmfailing with what error message?
04:07.57jplankI haven't tried 5.3 yet
04:08.03jplankwctdm24xxp:  FATAL: Could not open '/lib/modules/2.6.18-53.1.4.el5/extra/zaptel/wctdm24xxp/wctdm24xxp.ko': No such file or directory
04:08.10Qwelldepmod -a now :D
04:08.11jplankrealtek check
04:08.16Qwellyeah, 5.3 fixes that bug
04:08.28jplankwoo didn't fail yet
04:08.46jplankI'm actually moving to a different type of board
04:08.58jplankjust testing it now, without realtek nic
04:08.59jplankwooooooo
04:09.01jplankloaded
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04:09.13jplankstill getting one error though
04:09.14jplankopen error on /dev/dahdi/25: No such device or address
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04:10.22jplankwooo compiling asterisk-latest now!
04:10.40jplankshould I do asterisk addons before or after asterisk?
04:10.45Qwellafter
04:11.42ectospasmjplank: the no such device error means you've probably got an extra channel listed in system.conf
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04:15.24jplanklike we said before I could just do a mv /etc/zaptel.conf /etc/dahdi/system.conf to fix that right?
04:17.02jplankerrr guess not
04:17.15jplanknow I get Running dahdi_cfg:  DAHDI_SPANCONFIG failed on span 1: Invalid argument (22)
04:18.50jplankis there something different with a TE122 config with dahdi?
04:20.01jplankwhats the dahdi equivalent of genzaptelconf? gendadhi_conf or something like that?
04:20.56jplankdahdi_genconf, got it
04:21.46jplankthats weird, it switched the card's spans when I ran dahdi_genconf
04:23.42jplankis there any reason I couldn'
04:23.51jplankcouldn't switch them back?
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04:37.42jplankanyone know what /dev/dahdi/25: Inappropriate ioctl for device means?
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05:54.56Hydrantanyone have experience with polycom phones ?
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06:05.08carrarI'm you're huckleberry
06:06.56HydrantI think I've found my problem
06:07.20Hydrantthe digitmap... when I dial with the phone off the hook it screws up, otherwise it's fine
06:07.28carrarwait, people can resovle there own issues here?
06:07.31Hydranttried to write my own digitmap in sip.cfg
06:07.33Hydrantsometimes
06:13.04carrarimpossibleMatchHandling!!
06:13.20carrarread the admin guide
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06:16.05jplankwooohooo upgrade complete, dahdi installed, half duplex audio on fxo ports gone
06:16.16jplanksleepy time
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07:43.14PeacefulHelp!  We rebooted our asterisk server (1.4.22) tonight, and "AgentLogin" disappeared from the applications list -- when it's called from the dialplan it gives an error to that effect, and "show applications" doesn't list it.  What could have happened???
07:44.08PeacefulI mean, besides the disk going bad, but that doesn't seem to be the case.  fsck passed, smartmontools passed the self check, and I re-"make installed" asterisk from the same source directory as before...
07:45.44Peacefulfears that everyone is asleep...like I ought to be.
07:50.54Peacefulk, well, guess I'll have to go in and physically bang on the chassis or something.  I may be back.
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08:29.47lpccoderHello, I need some help
08:30.01lpccoderIs impossible to send or receive faxes over GSM?
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08:57.28*** join/#asterisk Jarak01 (n=no-body0@e179192042.adsl.alicedsl.de)
08:57.34Jarak01hi community
08:57.41Jarak01good morning;)
08:58.22Jarak01iam trying to get my asterisk server work on suse11.1, but it doesnt.
08:59.13Jarak01becoz when iam going to install libpri i get this error message :
08:59.14Jarak01linux-ir68:~ # cd /usr/src/libpri-1.4.10
08:59.15Jarak01linux-ir68:/usr/src/libpri-1.4.10 # make
08:59.15Jarak01gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD -MT copy_string.o -MF .copy_string.o.d -MP -c -o copy_string.o copy_string.c
08:59.15Jarak01gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD -MT pri.o -MF .pri.o.d -MP -c -o pri.o pri.c
08:59.16Jarak01gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD -MT q921.o -MF .q921.o.d -MP -c -o q921.o q921.c
08:59.19Jarak01gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD -MT prisched.o -MF .prisched.o.d -MP -c -o prisched.o prisched.c
08:59.22Jarak01gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD -MT q931.o -MF .q931.o.d -MP -c -o q931.o q931.c
08:59.25Jarak01gcc -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -g -fPIC   -MD -MT pri_facility.o -MF .pri_facility.o.d -MP -c -o pri_facility.o pri_facility.c
08:59.28Jarak01make: *** [version.c] Error 126
08:59.41ectospasmplease don't flood
08:59.45ectospasm!pastebin
08:59.50Jarak01iam trying to fix this problem for about 2 days
08:59.54ectospasm~pastebin
08:59.54infobot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
08:59.58Jarak01sorry doenstn want to get unmannered
09:00.31ectospasmyou didn't post the error
09:00.52Jarak01http://pastebin.com/m2a212f3
09:00.59Jarak01there is the error
09:01.25Jarak01its allways says: **make_version: permission denied
09:01.40ectospasmls -al build_tools/make_version
09:01.47ectospasmwhat does that output say?
09:02.22Jarak01wich output?
09:02.26Jarak01which*
09:02.30ectospasmof ls ^^^
09:03.03Jarak01isnt this the output?^^
09:03.29ectospasmtype this:  ls -al build_tools/make_version
09:03.40ectospasmwhat does the output of that command say?
09:03.46ectospasmit should be one line
09:04.16Jarak01http://pastebin.com/m7ffe257c
09:04.24ectospasmno need for pastebin
09:04.28Jarak01http://pastebin.com/m7ffe257c
09:04.30Jarak01ahh ke
09:04.37Jarak01linux-ir68:/usr/src/libpri-1.4.10 # ls -al build_tools/make_version
09:04.37Jarak01-rw-r--r-- 1 root root 1818 Mar 13 02:05 build_tools/make_version
09:04.48ectospasmmake_version is not executable.
09:05.11Jarak01so this means i need a executable version right?
09:05.21ectospasmno, you need to make it executable
09:05.31ectospasmhow familiar are you with Linux/UNIX?
09:05.34Jarak01ah okay isee
09:05.58Jarak01some kind of newbie
09:06.06Jarak011week experience
09:06.09ectospasmit's OK to be a nub
09:06.14ectospasmBut not OK to stay one (-;
09:06.21Jarak01word
09:06.32ectospasmchmod +x build_tools/make_version*
09:06.35Jarak01iam really expressend by the way
09:06.44ectospasmexpressend?
09:06.51Jarak01impressed* gah
09:07.09Jarak01chmod +x build_tools/make_version* -> done
09:07.15ectospasmnow, make
09:07.22Jarak01before make clear?
09:07.26Jarak01or direct make`
09:07.27Jarak01`
09:07.44ectospasmif you want to rebuild everything you've already done, do a make clean, but it really is unnecessary at this point.
09:08.11Jarak01ke i see. and it works
09:08.12Jarak01nice
09:08.26Jarak01so what does this command chmod +x build_tools/make_version* do?
09:08.41Jarak01it makes it executable?
09:09.55ectospasmmakes all the make_version scripts (make_version, make_version_c) executable.
09:10.00ectospasmLinux/UNIX 101
09:10.14Jarak01ah okay. ectospasm i thank you
09:10.21Jarak01how long youre into linux?
09:10.31ectospasm'bout 12.5yrs
09:10.34Jarak01woow!
09:10.36Jarak01amazing
09:10.55Jarak01so i can call you a guru:P
09:11.02ectospasmI wouldn't go that far
09:11.15Jarak01^^
09:11.26ectospasmlet's just say I'm much more comfortable with a small OS from Finland than one from Redmond, WA, USA
09:11.40Jarak01:D
09:11.57ectospasmThat being said, I bow before some of the developers in here
09:12.49Jarak01well well
09:12.50ectospasmQwell, russellb, file, Corydon76-dig (forgive me if I'm leaving anyone out!) all surpass me in breadth and depth of Asterisk knowledge.
09:13.21Jarak01y but maybe you got some points
09:13.33ectospasmpsst:  I work for Digium.
09:13.33Jarak01with more experience
09:13.52Jarak01who is digium?
09:14.03ectospasmyou really ARE an Asterisk nub!
09:14.04*** join/#asterisk aurax (n=aurax@bzq-84-108-166-31.cablep.bezeqint.net)
09:14.31ectospasmhttp://digium.com
09:14.39ectospasmTHE Asterisk folks.
09:14.50auraxhello, i'm suddently geting SIP Status: 407 Proxy Authentication Required, but nothing was changed in my configuration
09:14.53auraxany idea?
09:14.59Jarak01y sorry i add this site to my favorites
09:14.59ectospasmaurax: that's a normal message
09:15.07auraxi'm not able to receive calls because of it
09:15.42ectospasmtypically with the first INVITE, the system will respond with 407 Proxy Authentication Required, then the side sending the INVITE sends another INVITE with the authentication credentials.
09:16.04ectospasmif you've got the authentication credentials wrong, you've got a problem.
09:16.09Jarak01excuse me. I have to correct myself i ofc know digium!
09:16.21auraxbut i'm able to perform calls.. so the credentials are right
09:16.34auraxit's a net2phone trunk... and outgoing work's just fine...
09:17.06auraxi'm also getting: 407 Unauthorized    (1 bindings)
09:17.08ectospasmoutgoing and incoming are separate.  Usually you have a user and a peer, and I can never remember which one sends calls or receives them
09:17.22auraxi have only outgoing settings and registration string.
09:17.29ectospasmone way of getting around this is by declaring a "friend" for the trunk in sip.conf
09:17.31auraxthey never said anything about incomign settings :)
09:17.44ectospasmA friend is both user and peer in one stanza
09:17.56auraxshould i type=friend instead of type=peer in outgoing ?
09:18.19ectospasmin the SIP peer/user context in sip.conf, try type=friend, yes
09:18.40auraxshort question, type=friend will identify them by IP right?
09:18.42ectospasmyou may also need a register line, but the ITSP should be able to tell you you need that
09:18.47ectospasmno
09:18.51ectospasmhost=IP will
09:18.59auraxhmm
09:19.04auraxtesting ..
09:19.14ectospasmsip reload, cross your fingers and hold tight!
09:19.27Jarak01mr.guru i have one more question for you now iam trying to install zaptel
09:19.30auraxi see it... there's host=ippbx.net2phone.com but there's no reverse dns
09:19.35*** join/#asterisk GameGamer43 (n=GameGame@cpe-67-247-172-185.rochester.res.rr.com)
09:19.35auraxfor that hostname.
09:19.42ectospasmJarak01: why Zaptel?
09:19.49ectospasmZaptel is obsolete
09:20.03ectospasmwhat version of Asterisk do you have installed Jarak01?
09:20.16Jarak01i havent installed asterisk yet
09:20.25Jarak01first i have to install libpri
09:20.29Jarak01zaptel
09:20.31Jarak01and then asterisk
09:20.36Jarak01but i will install version
09:20.39ectospasmit goes:  zaptel/dahdi, libpri, asterisk, addons, etc.
09:20.42Jarak01asterisk-1.4.25.1
09:20.49ectospasmGo with DAHDI
09:20.51ectospasmNot Zaptel
09:20.59ectospasmZaptel is no longer being developed.
09:21.07ectospasm...due to trademark issues.
09:21.10Jarak01okay but tutorial told me to install libpri first
09:21.14Jarak01ah ke
09:21.19ectospasmwhat tutorial are you reading?
09:21.36Jarak01http://pbx-manager.de/installation-suse-10.2-asterisk.php
09:21.45ectospasmActually, I think it's really immaterial which one you do first, but I've always done Zaptel/DAHDI, then libpri, then Asterisk
09:21.58ectospasmyak
09:22.01ectospasmDon't use SuSE
09:22.10auraxectospasm, tried to change type=peer to type=friend and changed the hostname from ippbx.net2phone.com to IP addr, also in the registration string, still same problem
09:22.11ectospasmthey're definitely not LSB
09:22.16Jarak01lawl
09:22.27ectospasmaurax: you did remember to do a sip reload?
09:22.31auraxyep
09:22.49ectospasmcan you pastebin the sip debug?
09:22.57ectospasmof the failed incoming call?
09:23.01ectospasmDoes anything show?
09:23.13auraxyou want me to turn debug on ?
09:23.18auraxand post pastebin ?
09:23.21ectospasmyes
09:23.27auraxsure, i'll just pm you the information
09:23.33auraxthx
09:23.37ectospasmOK
09:24.02ectospasmNo need for pastebin (you can blank out the passwords and usernames in pastebin) <- aurax
09:24.16ectospasmer
09:24.18ectospasmNo need for PM
09:24.26ectospasmbah, it's late
09:24.30ectospasmhard days night
09:24.32ectospasmand all that
09:24.42Jarak01whats the time?
09:24.59Jarak01i just woke up
09:25.01ectospasm0424
09:25.03Jarak01wow
09:25.16Jarak01from usa?
09:25.19ectospasmyup
09:25.25ectospasmHuntsville, AL
09:25.46Jarak01huntsville, heard about it
09:25.57ectospasmRocket capital of the world
09:26.05ectospasmNASA and a lot of military rockets are designed here
09:26.08Jarak01yea
09:26.12Jarak01ofc
09:26.14Jarak01i remember
09:26.33Jarak01which cinema-movie your waiting for?
09:26.35Jarak01(:
09:26.52ectospasmactually I'm hoping to see Star Trek this weekend
09:27.02ectospasm...haven't seen it yet
09:27.07Jarak01i watched allready
09:27.08Jarak012 times
09:27.15Jarak01great movie
09:27.16ectospasmthat bad huh?
09:27.26Jarak01^^ naah
09:27.42ectospasmj/k, I've been told it's good.
09:28.00auraxoh sorry ectospasm, missed your msg.
09:28.06Jarak01it is ofc but nevermind why you yaked at my suse
09:30.50ectospasmthey're definitely not LSB
09:30.53Jarak01anyway i uplloaded dahdi now i have to resume with ./configure  and then make linux26 make install?
09:31.06ectospasmmake linux26?  Are you crazy?
09:31.18ectospasmthat hasn't been required in a LONG time
09:31.28Jarak01lol xD
09:31.54ectospasmit should just be "make && make install" in the linux+complete source directory
09:32.23Jarak01linux is so damn difficult
09:32.27Jarak01but i will learn this!
09:33.04auraxJarak01, why are you compiling frmo source?
09:33.13auraxnever heard of precompiled binaries?
09:33.32auraxmakes your life easier :)
09:33.42Jarak01hmm
09:33.49Jarak01i dont know aurax
09:33.57Jarak01i guess becoz tutorial told me so
09:34.10ectospasmmost precompiled binaries are woefully out of date
09:34.18Jarak01okay
09:34.21auraxnah, not always
09:34.36Jarak01y iguess thats why i said most:P
09:34.55Jarak01so what other ways do i have
09:35.03auraxappliance?
09:35.09Jarak01means?
09:35.17Jarak01.rpm files?
09:35.28auraxno
09:35.37auraxtrixbox, or all of those AIO's
09:35.50ectospasmaurax: point me to a precompiled binary that's 100% up to date, and I'll be an impressed man
09:36.12Jarak01^^
09:36.19Jarak01there you go aurax:P
09:36.21auraxsure, but you will find those packages in test repository
09:36.34auraxof beta releases.
09:36.49ectospasmthat shouldn't be considered for a production system
09:36.49auraxnot in 'stable' repository.
09:36.56Jarak01so is it impossible to work with this "compiling from source" ?
09:37.00ectospasmJarak01: nope
09:37.03auraxsure it is
09:37.05Jarak01or where do i find precompiled binaries
09:37.09auraxyou just need to know how to compile things :)
09:37.25ectospasmit's just more difficult than your average Linux nub can handle (-;
09:37.39Jarak01okay
09:37.50ectospasmAnyway, I'm off to bed.
09:37.58Jarak01i actually i did like this: i downloaded the tar.gz files
09:38.04Jarak01extract em and uploaded em
09:38.30ectospasmupload?
09:38.43Jarak01y iam connected via remote
09:38.49Jarak01samba
09:38.51Jarak01and putty
09:39.04Jarak01to the linux server
09:39.26Jarak01aurax?
09:39.31auraxye?
09:39.39auraxnight ectospasm thx for the help
09:39.41ectospasmwhy not wget and be done with it, extract them on the linux machine?  Your permissions will probably be all wrong otherwise (as you've already seen)
09:39.46Jarak01do you have any uptodate tutorial
09:39.48ectospasmOK, that's my queue
09:39.52ectospasm~thebook
09:39.52infobotextra, extra, read all about it, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org
09:40.01auraxehe
09:40.06auraxJarak, get elastix :)
09:40.11Jarak01wget
09:40.11ectospasmno don't
09:40.14auraxhehehe
09:40.15ectospasmDo yourself a favor
09:40.16auraxwhy ectospasm
09:40.23auraxthe damn thign is working
09:40.25ectospasmnon-genuine Asterisk distro
09:40.27auraxfor noobs
09:40.32ectospasmtry AsteriskNOW
09:40.45ectospasmAll the goodnes of FreePBX without the ickyness of Trixbox
09:40.56Jarak01asterisknow http?
09:40.58Jarak01:P
09:40.58ectospasms/nes/ness
09:41.08ectospasmhttp://asterisknow.org
09:41.26auraxit's a digium :)
09:41.44ectospasmwell, yeah :-D
09:41.46auraxi remember trying to install it once, and it requset for a userpassword for each trunk, one of my providers doesnt use password
09:41.51auraxscrewed!
09:41.58Jarak01asterisk now is for windows?
09:41.59ectospasmwhat version did you use?
09:41.59*** join/#asterisk GameGamer43 (n=GameGame@cpe-67-247-172-185.rochester.res.rr.com)
09:42.02ectospasmJarak01: no
09:42.06auraxah it was 6 month ago...
09:42.08auraxi guess, old..
09:42.16ectospasmNo official version of Asterisk runs on Windows
09:42.34ectospasmAsteriskNOW is a complete Asterisk Linux distribution
09:42.38ectospasm...based on CentOS
09:42.44ectospasmAnyway, I'm going to bed now
09:42.46Jarak01i dont want mine on windows either
09:42.51Jarak01okay extospasm
09:43.00Jarak01thanks for your help
09:43.08Jarak01have a good night
09:43.14auraxnight ectospasm
09:43.24Jarak01aurax my friend;)
09:43.27*** join/#asterisk SkywaIker (n=pirch@58.147.17.166)
09:43.34auraxyes?
09:43.38Jarak01i need a tutorial or something like this
09:43.48Jarak01can i find one on asterisknow?
09:43.58GameGamer43its not hard
09:43.59auraxJarak01, download it, install it, get the pdf
09:44.01auraxit's very easy
09:44.13auraxJarak01, your server is local or remote?
09:44.18auraxyou have physical access to it?
09:44.21Jarak01iam a linux noob
09:44.24Jarak01y
09:44.24auraxwe know.
09:44.27Jarak01:D:D
09:44.27GameGamer43as aurax said
09:44.49Jarak01i can get physical access to it
09:45.09Jarak01and i can install it via opensuse?
09:45.11auraxyour arch is x86 or 64?
09:45.31Jarak01x86
09:45.36auraxhttp://www.digium.com/elqNow/elqRedir.htm?ref=http://www.asterisknow.org/files/downloads/load_balance.php?q=AsteriskNOW-1.5.0-i386-1of1.iso
09:45.38auraxget the iso
09:45.38Jarak01i guess
09:46.02Jarak01wich command let me show version infos?
09:46.03auraxhttp://www.asterisknow.com/support <- go there
09:46.03auraxand READ!
09:46.12auraxof what asterisk?
09:46.32auraxasterisk -V
09:46.38Jarak01naah of opensuse
09:46.43auraxcat /etc/issue
09:47.40Jarak01the iso is linux and server in 1?
09:47.54Jarak01linux and asterisk server all in one?
09:48.07auraxthe iso is "operating system" with asterisk binary packages
09:48.13Jarak01ahh
09:48.15auraxyes, it's all in one
09:48.17Jarak01kay
09:48.21Jarak01thank you aurax
09:48.22auraxyou install and poof you have asterisk with freepbx
09:48.32auraxand modules.. and all that crap
09:48.44auraxyou welcome
09:49.09Jarak01iam trying to install asterisk for about 4 days atleast 3hours a day
09:49.25Jarak01freepbx
09:49.29Jarak01and modules
09:49.32Jarak01like addons
10:01.51*** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr)
10:02.17*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
10:12.12auraxJarak01, let asterisknow do it for you
10:23.22*** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-1ca6700c32b93d96)
10:30.36thomashello
10:30.43thomaswhere i can find devstate() ?
10:33.08thomashttp://svncommunity.digium.com/view/russell/asterisk-1.4/func_devstate-1.4/
10:33.10thomasNot Found
10:34.54*** join/#asterisk hi365 (n=hi365@94.159.176.156)
10:47.45gr0mitanyone know how to push a message onto a polycom 430 during a call?
10:51.28*** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
10:54.41ZhadI take it SendText doesn't work then.
10:54.59gr0mitZhad, let me try!
10:55.35Zhadreally should play a bit more with his 501.
10:59.56*** join/#asterisk frk2 (n=frk2@zivios/member/fkhan)
11:00.59*** join/#asterisk sujith_h (n=kde-deve@59.96.20.186)
11:02.59sujith_hcan anyone help me for this error i get when I run asterisk with command, 'asterisk -U asterisk -C /etc/asterisk/asterisk.conf -d' . http://pastebin.ca/1457639
11:17.51tzafrir_laptopsujith_h, any chance you use the Debian package?
11:19.19*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
11:19.19*** mode/#asterisk [+o leifmadsen] by ChanServ
11:20.37Peacefulwhat would cause my AgentLogin application to disappear after a reboot?
11:21.26sujith_htzafrir_laptop: no I had used ubuntu package
11:21.36tzafrir_laptopthe same
11:21.54tzafrir_laptopsee the sample modules.conf
11:21.56sujith_htzafrir_laptop: Oops. So you mean issue with the ubuntu?
11:21.59tzafrir_laptopspecifically:
11:23.06*** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
11:24.14tzafrir_laptopspecifically:    noload => app_directory_odbc.so
11:24.25tzafrir_laptopnoload => app_voicemail_odbc.so
11:24.30tzafrir_laptopnoload => app_voicemail_imap.so
11:24.39tzafrir_laptop(unless you actually need them)
11:34.16sujith_htzafrir_laptop: ok. But my error seems like its having issues with connecting to mysql db itself. http://pastebin.ca/1457659
11:42.41*** join/#asterisk SAT1 (i=d945ad33@gateway/web/ajax/mibbit.com/x-f0254d90b82b8dff)
11:43.20SAT1Hi, is there any tutorial available to install asterisk with E1 card with SS7 support?
11:45.40*** join/#asterisk Heretic (n=BuRn@ZA1-securenode.echelon.co.za)
11:48.23Peacefuldahdi 2.1.0.4 won't build under gcc 4.3.2 on my new Gentoo box.  :-(  Gets this error:
11:48.34leifmadsengooooooooooooooooooooooooooooooooooood morning vietnam!
11:48.44leifmadsens/vietnam/#asterisk/
11:49.08SAT1vietnam?
11:49.15SAT1vietnam = asterisk?
11:49.29leifmadsenheh... no, I was thining about a movie
11:49.38SAT1:)
11:49.43SAT1Good Morning
11:49.58leifmadsenalthough I'm sure some comparisons with the vietnam war could be made with #asterisk
11:50.14SAT1haha yeah
11:51.25SAT1leifmadsen do you know some tutorial for newbies to install asterisk + E1 card with SS7 support?
11:51.57leifmadsenI do not.
11:52.10leifmadsenI'd be surprised if there were...
11:52.20leifmadsenthat is kind of an edge case in the asterisk world (at this point)
11:52.49SAT1hmm can you give some hints what would i need to get it working?
11:52.59SAT1like asterisk + zaptel + libss7? thats all?
11:53.02SAT1or something else as well?
11:53.26SAT1so i can start looking everything individually
11:54.40leifmadsenI don't do hardware, and I don't do SS7... so, no :)
11:54.50SAT1ok no problem
11:55.02leifmadsenI'm a dialplan and SIP guy
11:55.08SAT1anyone else pleaseee?
11:55.43SAT1me too, SS7 is greek to me, but its part of one project and i've to do it...
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11:58.40*** join/#asterisk eykosioux (n=eykosiou@87.194.124.8)
11:58.57eykosiouxhi
11:59.05*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
12:01.15*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
12:02.55ectospasmSAT1: libss7 is available for Asterisk 1.6.2, see russelb's blog post about it (I can't seem to find it at the moment)
12:03.37SAT1ectospasm many thanks i'll have a look
12:03.59*** join/#asterisk mgroman (n=miles@unaffiliated/mgroman)
12:04.07ectospasmSAT1: note that libss7 will only work with DAHDI
12:04.39ectospasmIs it pronounced "doddy" or "daddy?"  I always say "daddy".
12:05.01ectospasmalthough most people aren't comfortable saying it that way.
12:05.10ectospasm...so they say "doddy"
12:05.15SAT1hahaaa
12:05.21auraxectospasm, go to sleep dude
12:05.23aurax:P
12:05.26SAT1in german we pronunce it differently
12:05.26ectospasmcan't sleep
12:05.29auraxrofl
12:05.31SAT1daah diii
12:05.31ectospasmMeds haven't kicked in yet
12:05.38auraxlol why meds?
12:05.41*** join/#asterisk tamiel (n=tamiel@213.30.183.226)
12:05.56ectospasmbipolar I disorder with psychotic features.  And proud of it!
12:05.57mgroman!oh my | ectospasm
12:07.02auraxah manic depressive
12:07.06ectospasmyep
12:07.36*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
12:07.38ectospasmI was late taking my meds last night 'cuz a friend got into town, and I almost forgot to take my evening dose
12:07.41leifmadsenectospasm: doddy
12:07.58SAT1ectospasm the first challenge is to find russelb blog, google isn't helping much :(
12:07.59leifmadsenmost because I think 'daddy' is the dumbest sounding acronym in the world
12:08.00ectospasmleifmadsen: I'm still gonna say,"daddy," despite what everyone else says.
12:08.22ectospasm"who's your DAHDI!"
12:08.40ectospasmSAT1: lemme see if I can drum up something
12:08.54SAT1ectospasm thanks
12:11.46ectospasmhttp://www.voip-info.org/wiki/index.php?page=Asterisk+libss7
12:11.56SAT1http://www.russellbryant.net/blog/
12:12.13SAT1ectospasm let me chk
12:12.15ectospasmJust replace the version with 1.6, DAHDI 2.2.0-rc5+2.2.0-rc3, and libpri 1.4.10 with libss7
12:12.35mgromanah damn, i was gonna say that joke but i wasnt paying attention
12:12.58ectospasmthe joke is the whole reason why I say it like that!
12:13.01leifmadsen"Who's your DAHDI, and what does he do?"
12:13.11mgromanhaha
12:13.18SAT1ectospasm ok thanks...starting...
12:13.23auraxrewr
12:13.24ectospasmI was thinking more of a Donald Duck sketch
12:13.35*** join/#asterisk esaym (n=user@cpe-24-174-186-34.satx.res.rr.com)
12:13.49ectospasm...well, an imitation of Donald Duck, not a Disney-sanctioned sketch.
12:15.32*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
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12:20.42ectospasmhears crickets
12:20.49auraxectospasm feels like having a look at my pastebin again ? :
12:20.51aurax:)*
12:22.16ectospasmaurax: lemme look
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12:23.45*** mode/#asterisk [+o jtodd] by ChanServ
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12:38.12*** join/#asterisk jozza (n=chatzill@89-212-75-91.static.t-2.net)
12:39.53jozzahey all, i have aproblem with asterisk sip. I have to asterisk server with same extension numbers on both server. When i try to call from server 1 from extension 100 to another server to extension 100, i always get a 403 forbidden response. How do i fix that?
12:40.51Fernandojdkjozza: how are your extensions.conf from that extension?
12:41.49jozzafrom debug view it never seems to come to extension.conf definitions, it simply rejects the call. What do you mean how they are?
12:41.51leifmadsenjozza: fromuser probably
12:42.17leifmadsenjozza: relevant parts of dialplan and sip.conf will be necessary (from both servers) along with the sip debug
12:42.26jozzaok
12:42.35leifmadsen~pastebin
12:42.35infobot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
12:42.37*** join/#asterisk Aiatek (n=Asterisk@75.112.88.200.m.sta.codetel.net.do)
12:46.06*** join/#asterisk [jmc] (n=[jmc]@93.45.201.28)
12:46.57[TK]D-Fenderjozza: because the INVITE "from" on system "A"'s phone matches a local user on "B" it tries to auth against that user but using the inter-server details which don't match
12:46.58*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
12:47.12jozzai know
12:47.17jozzabut how to fix that
12:47.26[TK]D-Fenderjozza: You need to use "fromuser=" and "sendrpid" on the sender side, and "trustrpid" on the receiver
12:47.31jozzai have to have the freedom to usea any extension  want
12:47.44jozzaaha
12:48.24leifmadsenneeds to document this in chapter 4, if he hasn't already
12:48.25jozzawhats fromuser
12:48.31leifmadsenjozza: the user to authenticate as
12:49.06jozzai dont get it
12:49.41[TK]D-Fenderjozza: separation between callerid-type info (normally used as auth" vs "auth-user" style
12:49.42leifmadsenAst_A --> calls --> Ast_B
12:49.43jayteechapter 4? 3rd edition?
12:49.49leifmadsenAst_A fromuser=Ast_A
12:49.51jozzawhat do i do with pastebin when i-m done writing_
12:50.03leifmadsenjozza: submit it, then post the link here
12:50.07leifmadsenjaytee: ya...
12:50.23leifmadsenI gotta get the outline done on the 3rd edition before I can really start updating it though
12:50.23jayteelookin forward to it
12:50.30leifmadsenya... don't hold your breath :)
12:50.48eykosiouxdoes anyone use Asterisk in combination with Telephone (osx softphone)
12:50.55jozzahttp://pastebin.com/d3c9afe82
12:51.02eykosiouxi'm not getting any sound... :-S
12:51.44jayteeeykosioux, osx client and * on same net?
12:51.58leifmadseneykosioux: same system?
12:52.08leifmadseneykosioux: or different systems? same network? different network?
12:52.37leifmadsenjozza: well you provided part of what I asked for anyways....
12:53.05jozzawhat elzse do zou need_
12:53.06leifmadsenjozza: anyways, you add 'fromuser=server1' in the [server2] peer on server 1, and vice-versa
12:53.51jozzaok, i-ll trz
12:53.52leifmadsenjozza: otherwise, don't use type=friend -- use type=peer to match on the IP address and not the username in your [incomingserverX] definitions
12:54.01leifmadsenheads off to do some real work
12:54.02jozzaok
12:54.05[TK]D-Fender[08:47]<[TK]D-Fender>jozza: You need to use "fromuser=" and "sendrpid" on the sender side, and "trustrpid" on the receiver
12:54.23jozzai'll try
12:56.19jozzasendrpid is what?
12:57.06jozzaaha its yes or no
12:57.18leifmadsen;trustrpid = no                 ; If Remote-Party-ID should be trusted
12:57.18leifmadsen;sendrpid = yes                 ; If Remote-Party-ID should be sent
12:57.24leifmadsen(from the sip.conf.sample file)
12:57.56leifmadsenalthough I've never seen RPID settings cause issues with authentication
12:58.11*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
12:58.11*** mode/#asterisk [+o putnopvut] by ChanServ
12:58.14leifmadsenputnopvut: !
12:58.27putnopvutAITCH AYE!
12:58.39leifmadsenlol
12:59.04PeacefulWhy the heck would the "AgentLogin" application be missing???
12:59.12Peacefulon 1.4.22
12:59.45leifmadsenPeaceful: the module didn't load on startup?
12:59.48PeacefulI've been up all night trying to figure out where the heck it went.  Our asterisk server rebooted and after the reboot, it no longer had AgentLogin()
12:59.49[TK]D-FenderPeaceful: Shouldn't
13:00.06Peacefulso what can I try?
13:00.09leifmadsenPeaceful: it didn't get compiled when you upgraded? it tried to load the module from a different version of asterisk and failed?
13:00.22leifmadsenPeaceful: you can start asterisk in the foreground and look at the errors or warnings on startup
13:00.26leifmadsenPeaceful: asterisk -cvvv
13:00.44putnopvutActually I'd leave the v's off since you really only care about warnings and errors.
13:00.53putnopvutThe v's will just give you extra crap to sort through.
13:01.11leifmadsenputnopvut: oh right :)
13:01.18leifmadsenI just habit with 3 v's
13:01.34leifmadsenputnopvut:  which module has AgentLogin() again?
13:01.41putnopvutleifmadsen: chan_agent.so
13:01.45leifmadsenthat's what I thought
13:01.52Peacefulbingo
13:01.53leifmadsenPeaceful: ls /var/lib/asterisk/modules/chan_agent.so
13:01.54Peaceful.....[Jun 12 07:00:56] WARNING[6277]: config.c:772 process_text_line: parse error: No category context for line 1 of /etc/asterisk/agents.conf
13:01.55Peaceful[Jun 12 07:00:56] NOTICE[6277]: chan_agent.c:1141 read_agent_config: No agent configuration found -- agent support disabled
13:02.08leifmadsenPeaceful: whenever you have that kind of error, that is always the way to debug it
13:02.11*** join/#asterisk Greek-Boy (n=greek@41.222.89.77)
13:02.26putnopvutLet me see where that warning is generated.
13:02.44leifmadsenit's amazing how much information you can get by starting your asterisk in the foreground
13:02.54leifmadsenmissing a [general] section?
13:03.31putnopvutleifmadsen: it looks like it's something like that.
13:03.42leifmadseneither [general] or [agents] probably
13:03.45putnopvutA setting that is not inside a category like [general] or [agents]
13:03.50leifmadsenheh :)
13:04.14putnopvutPeaceful: Yeah, I'd check your agents.conf file and be sure all settings are inside a specific category.
13:04.23leifmadsenI think I will start on my hot-desking article this morning
13:04.34putnopvutAnd I think I will go get a bagel
13:04.40leifmadsenputnopvut: BAGEL FRIDAY!
13:04.44Aiatekhi, a friend recently sent me an agi script to read and say social security numbers
13:04.49Aiatekhow i can use it
13:04.52Aiateklook
13:04.55Aiateks,n,AGI(dice-numeros.agi,3,${CED})
13:04.55leifmadsenputnopvut: too bad bagels have so much sugars and such in them....
13:05.16Aiateki put it ./var/lib/asterisk/agi-bin
13:05.26leifmadsenAiatek: we don't know the script -- but it looks like you need to set the CED channel var and pass it to the script
13:05.29*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:05.46*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
13:05.47Aiatekhow could be the sentences
13:05.50leifmadsenZeeek: oh snap
13:05.54Zeeekdeath sentence
13:05.55leifmadsenAiatek: ?
13:06.20Zeeekleifmadsen: ?
13:06.24ZeeekAiatek: ?
13:06.25Aiateky mean
13:06.32*** join/#asterisk phr3ak (n=nnnnphr3@gnet.hu)
13:06.33ZeeekZeeek ?
13:06.40Aiateki need to capture first
13:06.55Aiateklater the scritp will tell me the numbers right?
13:07.22leifmadsenAiatek: we have no idea what the script does
13:07.29leifmadsenZeeek: !!
13:07.34Zeeekwhat does any script do?
13:07.43Zeeekit gives lines to read!
13:07.57leifmadsenZeeek: and stuff
13:08.44Zeeekfor ($ever)  do  SomeThing }
13:09.55leifmadsenyou're missing an opening bracket
13:10.09ZeeekI know
13:10.09*** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-e81ea816775fe094)
13:10.18ZeeekI'm mùissing a lot more than that
13:10.42leifmadsenof all the things I've lost, I miss my mind the most
13:10.44[TK]D-Fenderstill can't find Zeeek's Clue
13:10.46Zeeekhow is the Asterisk wold today? Ready for the Skype for Asterisk test at 12 Noon EDT?
13:11.15ZeeekI thought Glue Con had to do with sniffing ports... get it?
13:11.39Zeeek...  ....    ..   _
13:11.56mmlj4.... . ....
13:12.08Zeeek:)
13:12.17[TK]D-Fender____^____^^^^^^_____________________________________________________
13:12.32Zeeekwhite guys have no sense of rhythm
13:12.38leifmadsenplays drums :)
13:12.55putnopvutis white and has no rhythm
13:13.00putnopvutproves Zeeek correct
13:13.01Zeeekwhat do you call the guy who loads the truck for the band?
13:13.02putnopvutQED
13:13.07putnopvutZeeek: the drummer
13:13.12Zeeekzackly
13:13.22leifmadsendrummers get all the chicks
13:13.29ZeeekPeaceful was looking for Asterisky info
13:13.33Zeeekno way
13:13.35leifmadsenit is SO friday
13:13.41Zeeekgotta be a front man to do that
13:13.55*** join/#asterisk bgmarete (n=marebri_@196.201.208.129)
13:14.15putnopvutZeeek: btw, in case you don't recognize me in my IRC glory, I am "the other Mark" that you met in Rostock.
13:14.33ZeeekMark what is your last name, peterson?
13:14.36Zeeekor close?
13:14.41putnopvutNope, Michelson
13:14.54Zeeekdamn. Well I am very old as you know
13:14.56putnopvutMark Peterson works for a Danish company...can't remember what now.
13:15.03ZeeekI knew ther was a 'son' at the end :)
13:15.07putnopvut:)
13:15.31ZeeekSo wake up Steve Sokol, we're beating on Skype for Asterisk at 12 Noon EDT today
13:16.28Zeeekthere is a Skype link to the ZipDX bridge that will be up around then
13:16.51ZeeekI believe there are more Matts than Marks in Huntsville
13:17.00jozzaok i got it working now, thanks guy for your experties
13:17.40putnopvutZeeek: probably
13:18.05BlargMaN00When I use a call file to dial a number, I answer my handset, and then it doesn't start ringing in the handset, but the # I dialed starts ringing...  is it supposed to work like that??
13:18.20Zeeekputnopvut:  so you saw the "Rawhide" video?
13:18.31putnopvutZeeek: The one on your blog? I did.
13:18.50Zeeeknice little ad for Astricon. Or not...
13:20.11leifmadsenZeeek: steve isn't online right now -- I'd email him
13:22.15Zeeekyeah wake him up, it's his baby
13:22.30Zeeekhe doesn't have to talk, but he might be interested in observing
13:22.33Zeeekor not...
13:22.49*** join/#asterisk ickmund (n=ickmund@91.126.225.2)
13:23.55*** join/#asterisk coppice (n=chatzill@204.196.17.210.dyn.pacific.net.hk)
13:24.11*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
13:24.34*** join/#asterisk tobias (n=tobias@cpe-071-070-219-040.nc.res.rr.com)
13:24.45ickmundHow can I monitor incoming calls over isdn (debugging purposes)? First go with isdn, always used ngrep for ethernet
13:28.41Zeeekas soon as a real asterisk issue comes up, radio silence is in effect
13:29.05auraxanyone ever encoutered 407 proxy auth failed error?
13:29.17auraxonly on incoming... outgoing works fien
13:29.19auraxfine*
13:30.34[TK]D-Fenderaurax: pastebin.....
13:30.43leifmadsenZeeek: I meant YOU should email him :)
13:30.47leifmadsenI'm busy working on other things right now
13:31.19leifmadsenaurax: that's normal... that's how authentication happens
13:31.21Zeeekhahaha oh
13:31.46Zeeeknaw, I'm not paid for that. I'm not even in the beta!
13:31.51leifmadsenRecv INVITE, Send 407 Proxy Auth, Recv 200 OK, Recv INVITE w/ authentication
13:31.56ZeeekACk
13:32.30auraxi would like to pastebin debug info. but there's too much priv information
13:32.40[TK]D-Fenderleifmadsen: Or like his case probably requires : INVITE, 200 (insecure=port,invite) <- fix your peer
13:32.40auraxmind if i PM?
13:32.41leifmadsenaurax: so take out the priv info
13:32.47[TK]D-Fenderaurax: See above
13:32.52leifmadsenregardless -- configuration issue
13:33.40*** join/#asterisk Polysics (n=Luca@host95-70-dynamic.16-79-r.retail.telecomitalia.it)
13:33.47Polysicshi to all
13:34.06Polysicsdoes anyone know about a good browser-based SIP client, please?
13:34.11auraxhttp://pastebin.com/d53aeb7e3
13:34.36auraxthat's a sip debug output, the problematic trunk host is ippbx.net2phone.com
13:34.52[TK]D-Fenderaurax: [NP-Vancouver} insecure=port,invite
13:34.58Polysicsi'm trying to make my way around a project but i fear i do not understand the Asterisk/Openfire/SIP stack at all :-)
13:35.02auraxi can esteblish calls outside but when i call in it fails, other trunks that does not require registration will work fine
13:35.12auraxok trying
13:36.39Polysicsbasically, i'm trying to figure out if i actually need asterisk to enable browser-based voice communication, with call redirection on phisycal phones if needed
13:37.46*** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net)
13:37.49ickmundIs there a way to capture packets on a isdn-card much like ngrep does on ethernet?
13:37.57Polysicslooks like the weak link is the clients
13:38.15*** join/#asterisk elguero (n=elguero@ns1.nashuacs.com)
13:38.44*** join/#asterisk NoxIn- (n=noxin@zorlit.org)
13:38.47aurax[TK]D-Fender, same...
13:38.56[TK]D-FenderPolysics: What does * have to do with browser based communications?  thats a BROWSER APP.  * is a PBX & telephony toolkit that might serve some sort of back-end role.  What do you ned to actually do?
13:39.08[TK]D-Fenderaurax: PB your config and new failed call.
13:39.20auraxyes sir
13:39.58*** join/#asterisk aksyn (n=aksyn@94-193-98-124.zone7.bethere.co.uk)
13:39.59eppigyHELLO
13:40.18Polysics[TK]D-Fender, what i need to do is: some people are professional translators for a social service for foreigners. they usually sit at their desks, and each of them can need the help of one of the others, with a foreign person sitting in front of them speaking a language they do not know
13:40.51Polysicsin normal conditions, they can call each other using a browser-based app, after seeing whoever is online
13:41.10Polysicsbut they might be outside the office and the call will then need to be routed to a cellphone
13:41.46Polysicsbased on a table which says who of the, say, arab-speaking persons is "on call" that day or night
13:41.49auraxhttp://pastebin.com/d7ba143ae
13:42.17*** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) [NETSPLIT VICTIM]
13:42.33Polysicsi'd say * is the correct basis for the app, seeing as voice is way more important than IM/presence, which might not even get implemented if too difficult
13:43.17aurax[TK]D-Fender >http://pastebin.com/d7ba143ae
13:43.30Polysicsanother functionality that is needed is: someone call a #, an IVM tells them to push the number of the language they need, then the call gets routed
13:43.39auraxthere is no incoming context settigns with net2phone, (as they explained) only registration string and outgoing context
13:44.06[TK]D-Fenderaurax: You've chopped off bits....
13:44.34Polysicsi suppose that a web-based softphone with VPN support basically solves the above
13:44.43*** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844)
13:45.03Polysicsbut since i couldn't find one, i stumbled upon Openfire/Sparkweb and was wondering if it made sense to look at those for a solution
13:45.41Polysicsam I totally off track? :-)
13:45.42[TK]D-FenderPolysics: Yes, * can act as a PBX core for this.
13:45.51auraxwhat do you mean ?
13:45.54*** join/#asterisk juanIMP (n=Juancho@200.71.41.254)
13:46.02[TK]D-Fenderaurax: I don't see a sip.conf section name, etc
13:46.19*** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net)
13:46.56[TK]D-Fenderaurax: You also shouldn't need "port=" for that.
13:47.25*** join/#asterisk sikanrong (n=sikanron@211.105.219.87.dynamic.jazztel.es)
13:47.27Polysics[TK]D-Fender, what I am trying to understand is if the above is possible without having to code a client myself or that level of things - i could, but i do not have the time
13:47.47[TK]D-FenderPolysics: No, there are java SIP apps, etc you can deplo for this
13:47.55[TK]D-FenderPolysics: deploy*
13:48.12Polysicsso Openfire/Spark are not in the picture?
13:48.26Polysicsi do not need a HOW, i'd be happy with an IF :-)
13:48.32Zeeekmust run
13:48.37sikanronganybody have any experience with voovox? I have a new personal # from them and I'm trying to get asterisk to handle the incoming calls but it's not, keeps going to voicemail - I have the register line set right in sip.conf and outgoing calls work fine
13:48.40Zeeeksay hi th Katty
13:49.14Zeeekcome and visit us in #voip-users-conference any old time or see http://vuc.me for today's party info
13:49.17*** part/#asterisk gego (n=rick@b238085.customer.hansenet.de)
13:49.18*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
13:49.36[TK]D-FenderPolysics: You just said "IVR" + web based soft-phone.  So far * can do all that
13:50.16Kattywhat
13:50.30KattyWHO"S MAKING ME BEEP
13:51.16[TK]D-Fenderkatty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY katty KATTY
13:51.19[TK]D-Fender;)
13:51.27[TK]D-FenderKatty: .... mew
13:51.41Kattytwitch
13:51.54Kattygives [TK]D-Fender a cookie.
13:52.16Polysicsone last question, any names for the above Java web-based softphones?
13:52.48[TK]D-FenderPolysics: Go searh the WIKI as they have a list of them
13:52.50[TK]D-Fender~wikis
13:52.50infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
13:53.08Kattysip-communicator comes to mind.
13:54.10Kattynot sure on its embedability.
13:54.12Kattyis that a word?
13:55.35Polysicsprobably not but i thank you anyway :-)
13:55.52Kattyit sounds kinky to me.
13:55.59Kattywhat's your embedability rating?!
13:56.00Kattydundundun
13:56.38Polysicsit actually does exist, if 58300 results on Google mean that a word is existing :-)
14:02.23*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
14:02.56*** join/#asterisk jtodd (i=u2wktkq9@ns.fox-den.com)
14:02.56*** mode/#asterisk [+o jtodd] by ChanServ
14:05.19leifmadsenDarthPointer: ping
14:07.41*** join/#asterisk lpccoder (n=chatzill@81.203.255.90.dyn.user.ono.com)
14:07.51lpccoderhello, i need help about faxes
14:07.57*** join/#asterisk [jmc] (n=[jmc]@93-45-201-28.ip104.fastwebnet.it)
14:08.06lpccoderis any way to send and receive faxes over GSM?
14:08.31*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
14:08.31*** mode/#asterisk [+o Deeewayne] by ChanServ
14:09.54lpccoderhelloooooo
14:09.58lpccoderanybody here?
14:10.50*** join/#asterisk Maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com)
14:11.19*** join/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net)
14:11.24ariel_lpccoder: yes people are here and over GSM not that I know of, Faxing only really works using ulaw sometimes and or via T38.
14:12.20lpccoderulaw? T38?
14:12.30jaytee~book
14:12.30infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
14:12.39jaytee~google
14:12.39infobothmm... google is http://lmgtfy.com/?q=google
14:12.41kaldemargsm codec lacks the bandwidth
14:13.10lpccoderok
14:13.28lpccodercan take a look to this device? http://www.junghanns.net/en/quad_gsm_pci_produkt.html
14:13.38jayteeand T38 should not be confused with the USAF jet trainer of the same name that is also used by NASA
14:14.01lpccoderjeje
14:14.18lpccoderis a good option to create a PBX using asterisk?
14:14.24*** join/#asterisk eykosioux (n=eykosiou@87-194-124-8.bethere.co.uk)
14:14.40lpccodercan send or receive faxes this device?
14:14.48coppiceI feel so good when my T.38 pages start to rise in the Google rankings against aircraft fan pages :-\
14:15.09kaldemari've tried quadgsm briefly. seemed to work nicely.
14:15.26NMR_1122hi, Is there a log that shows step-by-step what asterisk did during a given call? Like what the console would show if you were connected to it?
14:16.01kaldemarlpccoder: forget fax over gsm
14:16.02ariel_Well I like the T38's and also the F5's.  There great jets to fly in...
14:16.12[TK]D-Fenderlpccoder: NO
14:16.20lpccoderok, thansk
14:16.22lpccodera lot of thanks
14:16.23jayteeariel_, an F4 is good too
14:16.24lpccoderjejeeejej
14:16.35lpccodersorry for my newbie questions...
14:16.37ariel_F4 was way too big and noisy.
14:16.39[TK]D-Fenderjaytee: Alt-F4 is much better ;)
14:17.02coppicean F4 is a whisper jet compared to a Viggen
14:17.16ariel_when I was young I was in the USAF......Flying in nice aircrafts....
14:17.16eykosiouxhi, i'm having problems calling from my pc sip client through our asterisk server, I can call, but no sound inbound nor outbound,
14:17.32*** join/#asterisk seanmh (n=johndoe@198.59.129.24)
14:17.33eykosiouxcould anyone give me clues as to what could be happening?
14:17.36jaytee[TK]D-Fender, you weren't in here for the whole day on Wednesday. We need to see a note from either your mother or your doctor for the unexcused absence.
14:17.47*** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:4021:e3e2:fa0f:498)
14:17.49ariel_eykosioux: nat issues.
14:18.07[TK]D-Fenderjaytee: BS, I was in from around 5:30 onwards :)
14:18.19gr0mitanyone got polycom push messaging to work?
14:18.23jayteeyeah, but not during the normal dayshift
14:18.25eykosiouxariel_ i tried enabling nat=yes but no luck, is that what you're referring to?
14:18.32gr0mitwant to push queue stats to an IP430
14:18.33[TK]D-Fenderjaytee: But all morning I was in my company's old manufacturing plant... taking them down forever.
14:18.47jayteeeykosioux, we asked earlier if you were on the same NETWORK etc but YOU NEVER ANSWERED
14:18.48ariel_eykosioux: that is a start but just a start
14:19.11eykosiouxariel_: okay i guess i'll look up more info on nat :-)
14:19.16ariel_~nat
14:19.16infobothmm... nat is Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
14:19.35[TK]D-Fendergr0mit: I do PULL on mine with the MB.  Normally agents on the phone don't need to know how much work is waiting for them :)
14:19.47[TK]D-Fender~sipnat
14:19.47infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
14:19.49[TK]D-Fender^^^^^^
14:19.53jayteeeykosioux, also make sure your firewall on the client has the ports open for RTP
14:20.22eykosiouxthanks, for the info :-) (i didn't even check the firewall ports, stupid me)
14:21.01gr0miti have the idle screen displaying ifo
14:21.36gr0mitbut the only way I can get it to display the information in a call is pressing the menu, choosing applications
14:21.43gr0mitwhich is a bit of a Long Way round
14:22.01gr0mitI just want the same idle screen displayed during a call as well
14:22.13gr0mithas been tearing his hair out
14:23.00*** join/#asterisk VoipForces (n=kvirc@office.prival.ca)
14:23.27VoipForcesAnyone has experience with PRI failover devices like this one: http://www.failsafevoip.com/index.php?main_page=product_info&products_id=1%22
14:23.41gr0mit[TK]D-Fender, do you have any solutions to displaying this during calls?
14:25.05[TK]D-Fendergr0mit: gr0mit Nope, haven't touched the active app stuff yet
14:25.33gr0mitI have to believe it is possible
14:25.58gr0mitbut the documentation doesn't seem to have enough complete examples for numpties  like me
14:26.37gr0mitI have read and re- read the documentation,
14:26.38VoipForcesgr0mit: Missed a part, what are you trying to do?
14:26.51*** join/#asterisk virtualme123 (n=chatzill@fentech.gotadsl.co.uk)
14:27.05gr0mitokay, in the idle screen I am able to display two lines of text on my Polycom. Relating to the status of a queue.
14:27.12gr0mitIt is perfect during idle.
14:27.27gr0mitAll I want to be able to do is display the same two lines of information whilst the agents are actually on the call!
14:27.40VoipForcesgr0mit: Ok, sorry there can not help you on Polycom, I mainly work with aastra :-)
14:28.00gr0mitWould this be possible using Astra phones?
14:28.26VoipForcesDuring a call... don't think so. Then again I never tried.
14:28.44gr0mitWhy is it so difficult to display something during a call?!
14:29.05VoipForcesMaybe you could push data to the screen while in a call. I know I can toggle the phone leds during a call, never tried the screen.
14:29.07gr0mitSnom's don't seem to do it, Polycom's don't seem to do it
14:29.24gr0mitthis is what I wanted to do, but I can't get my head around the manual
14:29.34gr0mitthe phone apparently accepts push messages
14:29.42gr0mitbut maybe there is a concept I am missing.
14:29.56VoipForcesgr0mit: I can give you an exemple that toggles an Aastra led.
14:30.16gr0mit<PROTECTED>
14:30.21VoipForcesgr0mit: On the Aastra you need to allow reception of xml push fro perticular IP.
14:30.27gr0mitThat might be very helpful actually
14:30.41gr0mitwhat format does the push message take?
14:30.53VoipForcesgr0mit: Yes an xml puch that plays with the phone LEDS, I also have code that actually makes the phone dials by itself
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14:32.00VoipForcesgr0mit: http://pastebin.com/m6587c216
14:32.16VoipForcesgr0mit: This is a php script that will toggle an aastra phone LED
14:32.46VoipForcesgr0mit: script.php phone_ip prgkey_number state
14:33.02gr0mitcool, thank you very much
14:33.09VoipForcesgr0mit: Exemple: script.php 192.168.2.212 7 on
14:33.14gr0mitI will see if this works for a Polycom
14:33.23gr0mitwith a certain amount of tweaking, obviously
14:33.24VoipForcesgr0mit: will turn on the prgkey 7
14:33.45VoipForcesgr0mit: I don't think the Polycom push is remotly similar to the aastra.
14:34.21VoipForcesAnyone about the T1 failover switch ?
14:35.20gr0mitno, but I think the basic concepts might help me understand
14:35.46ariel_VoipForces: Failover how?  Right now I am working on a similar setup testing a Red-Fone fonebidge and a unit from Xorcom
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14:36.05gr0mitall I'm looking for is an example from Polycom about a "hello world" method of flashing something on a display during a call
14:37.58gr0mitall I want to do is push a message to the phone and says i"display the idle screen URL"
14:38.05ariel_gr0mit: have you looked into how hint works.  It sends messages to the button as to if on a call or not.
14:38.36gr0mitthis uses zip notify does it not?
14:38.41gr0mitI mean sip
14:40.23VoipForcesariel_: Physical PRI. Have 2 asterisk boxes that are exact copies of each other. If the primary fails for some reason, the PRI needs to be automatically redirected to the active standby box.
14:41.46ariel_ok that is what I am working on as well.  And difference between the redfone and xorcom astribank is one is via TDMoE and the latter over USB
14:42.43VoipForcesariel_: This is already a production setup using Sangoma cards so I can not swith to an astribank or other eternal PRI gateway
14:43.52VoipForcesariel_: For some reason pumping calls via a USB post does not sound like a good idea.
14:44.09ariel_VoipForces: OK but I have not seen what you posted before or in production.  It's just a layer 1 device which I figure is a passthrough device.
14:44.32VoipForcesariel_: Here is the link: http://www.failsafevoip.com/index.php?main_page=product_info&products_id=1%22
14:44.38VoipForcesariel_: Seems nice
14:45.39VoipForceshas just recorded his temporary greeting for next week vacation :-)
14:47.05ariel_VoipForces: saw it when you posted it before.  It seems a pass through hardware. Which I feel it's just adding another hardware to the mix.  But I also have not tried something like that.
14:48.20ariel_I would rather have the actual device that has the PRI terminated too be able to switch between end points
14:48.20[TK]D-FenderRedFone... GAH
14:48.27[TK]D-FenderVoipForces: http://rhinoequipment.com/1portfail.html
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14:49.58VoipForces[TK]D-Fender: Interesting. Ever tried it?
14:50.45[TK]D-FenderVoipForces: Nope, but I would never touch Xorcom or redFone personally... I'd sooner trust a little A/B switch type device
14:51.04eppigyDONDE ESTA
14:51.16[TK]D-FenderTRABAJO
14:52.30Aiatek<[TK]D-Fender> hablas espanol?
14:52.40eppigySI
14:52.50eppigyNO NECESITO TRABAJO
14:53.00eppigyQUE PASA ME AMIGO
14:53.09Aiatekhahahahahahahha
14:57.00ariel_wow, don't need work.... hum and why the yelling?
15:01.33aurax[TK]D-Fender still around ?
15:02.48[TK]D-Fenderaurax: Yes
15:03.58auraxpb my sip config
15:04.00*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
15:04.00auraxsec
15:04.56gr0mithttp://wendolene.txrx.org.uk/secretstuff/IMG_4165.jpg  <-- i just want to display these 2 idle screen lines during all calls
15:05.23gr0mitrefreshed every 10 seconds
15:06.30*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
15:06.33gr0mit<clarkson> How hard can it be?</clarkson>
15:06.43auraxhttp://pastebin.com/m48ae228b
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15:10.55Kattyrandomly.
15:11.00Kattywhy would i keep dreaming that i'm smoking?
15:12.01ariel_because your hot?
15:12.25aurax[TK]D-Fender?
15:12.54Kattyariel_: i am not hot.
15:12.54eppigyhi Katty :>
15:13.00eppigyKatty is hot
15:13.01Kattyohaider
15:13.03eppigyand a liar
15:13.12Kattyeppigy: why do i keep dreaming that i'm smoking?
15:13.32lftsyHello all, I have a "Installed a Wildcard TC: Wildcard TC400P+TC400M" with "(G.729a / G.723.1) Transcoder", I have installed dahdi 2.2.0-rc5 and Asterisk 1.4.25 but I have some clock snc I think. Have you got any idea how to troubleshot this?
15:13.47ariel_has not really seen or met Katty but feels that in a dream your always hot....no matter what.
15:14.06auraxthinks ariel_ needs to leave his computer once in a while
15:14.17aurax:)
15:14.49ariel_aurax: what are you trying to do, seems like a normal setup on your pastebin
15:15.09ariel_aurax: I do leave my pc allot....
15:16.19ariel_has one at work, on his cell phone (HTC) and his BB, oh and on his home tv there is a pc that runs it, and my house runs off my asterisk pbx for all it's lites etc, and also I have cams all over the house. But I am not a geek...
15:17.40Kattyyou say geek like it's a bad thing.
15:17.52*** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115)
15:18.00[TK]D-Fenderaurax: make a separate user entry for net2phone
15:18.00ariel_no it's not
15:18.09shyam_k[OT] does anyone have ebook of telecommunication networks?
15:18.15Katty[TK]D-Fender: why do i keep dreaming that i'm smoking?
15:18.24shyam_kby schwartz
15:18.32[TK]D-FenderKatty: Because you want to feel "cool"?
15:18.34*** join/#asterisk ramindia (n=Administ@202.63.96.10)
15:18.42Katty[TK]D-Fender: i am cool.
15:18.50[TK]D-FenderKatty: /badass/sophisticated
15:18.57Katty[TK]D-Fender: hmm.
15:19.11Kattyi guess it's possibly, but i already feel totally awesome in my own little way.
15:19.30Kattys/possibly/possible.
15:20.18Kattysmoking is not cool, badass, or sophisticated tho:<
15:20.38Kattyit's smelly.
15:20.47Kattynot to mention very bad for your health.
15:20.54*** join/#asterisk icyValk77 (n=icyValk7@host217-43-31-121.range217-43.btcentralplus.com)
15:21.02Kattyeppigy: mexican for lunch?
15:21.09*** join/#asterisk mltlnx (n=mltlnx@asams.mserve.com)
15:21.09Kattyeppigy: chimichanga is sounding goooood.
15:21.47ariel_talk about not being good for your health, Mex food????
15:21.49mltlnxHello, Is anyone using multiparking in 1.6?
15:22.17eppigyi like to say chimichangas
15:22.30eppigyKatty: I am having strange dreams myself
15:22.32eppigylately
15:22.55Katty:<
15:23.12Kattyariel_: mexican food does not leave tar all over ones lungs.
15:23.39Kattyeppigy: DUDE
15:23.40*** join/#asterisk agx (n=Antonio@host63-216-static.34-88-b.business.telecomitalia.it)
15:23.44Kattyeppigy: can you say margarita
15:23.46*** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net)
15:23.49Kattyeppigy: for LUNCH
15:24.04Kattyeppigy: strawberry, of course.
15:24.14eppigy:D
15:24.16eppigyI am down
15:24.22Kattyeppigy: might have to substitute vodka or rum for the tequila.
15:24.43eppigyI like good tequila
15:24.45ariel_Katty: I agree I don't smoke either and don't intend on starting any time soon....
15:24.45Kattyeppigy: we can still call it a margarita tho.
15:24.56agxHi all drunked people around :) is there a way to signal Busy on an incoming FXO call? i just used: exten=>s,1,Busy  and exten=>s,2,Hangup but the telco keep ringing me
15:25.09Kattywho said anything about being drunk?
15:25.44Kattyagx: you could set the status.
15:25.56Kattyagx: but, why would you want to do that?
15:26.26agxKatty: i want this guy calling me to think i'm busy
15:26.38Kattyagx: i would use the blacklist feature to do that.
15:26.56Kattyagx: it checks the number against the blacklist database, and then sends it to a context of your choosing, to do whatever.
15:27.08agxKatty: indeed i want it "Busy()"
15:27.23agxit works with SIP and PRI channels but not with the FXO (analog line)
15:27.25*** join/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej)
15:27.32Kattyagx: it's common for the the audio recording to say please check the number and try again blahblahblah
15:27.42Kattyagx: like it doesn't exist.
15:27.54*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
15:27.59*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
15:28.14agxKatty: i want Busy() working not a message
15:28.30Kattyagx: you can do whatever you want in the context.
15:28.52agxok lets change the question
15:29.05agxdoes everybody knows why Busy() does not works with FXO protocol? :)
15:29.34Aiatekuse congestion
15:30.04agxAiatek: trying....
15:30.28[TK]D-Fenderagx: HUH?
15:30.29*** join/#asterisk marv[work] (n=timr@router.asteriasgi.com)
15:30.29*** join/#asterisk jicksta (n=jicksta@67.164.0.78)
15:30.47[TK]D-Fenderagx: the only way for the telco to think you're busy is if you're busy
15:31.07[TK]D-Fenderagx: You can fake the AUDIO of it, but it will technically be an ANSWERED call
15:31.54Aiatekhttp://www.pastebin.ca/1457864
15:32.00Aiatek<agx>
15:32.14Aiatekbut its like <[TK]D-Fender> told you
15:32.28Aiatektechnically be an ANSWERED call
15:32.39Kattyanswered, played an audio file to, and hung up on
15:32.47Kattyi prefer the please check the number and try again route.
15:32.53*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
15:33.07Aiatekyeah its better
15:33.36Kattya busy audio file would be more believable.
15:33.41agxAiatek, [TK]D-Fender, ah ok... so i've to ANSWER + Congestion(7) + Hangup .... ok this simulate it; there is one more problem... when i Hangup the Telco keep the other channel open
15:33.47Kattyespecially if said person called from another callerid.
15:33.57[TK]D-Fenderagx: What other channel?
15:34.01Kattyit wouldn't be obviously. they'd asume all the lines were busy.
15:34.27agx[TK]D-Fender: another asterisk dialing out using a PRI mISDN.  PBX1 --> mISDN --> TELCO --> FXO --> PBX2
15:34.48mltlnxMultiparking anyone? Is it in trunk?
15:34.57[TK]D-FenderAiatek: And that lcode sample is nifty but shouldn't fool anyone.
15:34.58agxwell i use Congestion(7) perhaps have to use a larger Timeout ?
15:35.16aurax[TK]D-Fender http://pastebin.com/d4ac36037
15:35.24Aiatekwell disconect the lines
15:35.41[TK]D-Fenderaurax: You have no type,host, etc in there... seriously...
15:36.34auraxshould i type=friend there?
15:36.49agxaurax: you missing the server IP or hostname, the username and the password
15:36.51[TK]D-Fenderaurax: Told you already... USER
15:37.04*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
15:37.40ZeeekJust in case you haven't heard: http://vuc.me or go to #voip-users-conference for the Friday call in 10 minutes
15:38.36Kattyhi zeeek.
15:39.16Zeeek{{{{{{{{{{{{{{{{{{{{{{{Katty}}}}}}}}}}}}}}}}}}}}}
15:39.21KattyZeeek: <3
15:39.30Zeeekdid I hurt you?
15:39.36Kattybaroo?
15:39.46Zeeektoo many left braces?
15:40.11Kattyi uhh?
15:40.32Kattybaroo?
15:40.54Kattyinfobot: baroo?
15:41.07Kattyinfobot: baroo is http://www.youtube.com/watch?v=brJvJ2MtQIk
15:41.07infobotokay, Katty
15:41.11Zeeek~seen Zeeek
15:41.14infobotzeeek is currently on #asterisk (4m 10s). Has said a total of 5 messages. Is idling for 3s, last said: '~seen Zeeek'.
15:41.29ZeeekI've been idling all day so the bot is crap
15:42.18[TK]D-Fender[11:37]-->|Zeeek (n=Zeeek@pdpc/supporter/active/zeeek) has joined #asterisk
15:42.24[TK]D-FenderZeeek: Seems rather precise to me
15:42.35ZeeekI'm a jochk strap
15:43.11*** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com)
15:43.26Kattyyou're a mock strap.
15:43.26Zeeekyes the money I give to Freenode keeps you guys afloat :)
15:43.26agx[TK]D-Fender: thank's anyway i'm fine with: Answer + Busy 15 + Hangup ... its not me i'm paying for the call :-P
15:43.40*** part/#asterisk agx (n=Antonio@host63-216-static.34-88-b.business.telecomitalia.it)
15:43.46Kattyinfobot: baroo?
15:43.46infobotbaroo is probably http://www.youtube.com/watch?v=brJvJ2MtQIk
15:44.02Kattyinfobot: good bot :>
15:44.02infobotaw, gee, Katty
15:44.21Zeeekok folks duty calls: #voip-users-conference to hekle me
15:44.25*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
15:45.25angryuserCan someone tell me why sangomas driver require path to asteris sources during install ?
15:45.53MikeJto zaptel maybe
15:45.58MikeJit patches zaptel
15:46.01eppigySTARVING
15:46.02angryuserIt is supposed to patch zaptel only
15:46.13*** join/#asterisk CunningPike (n=CunningP@204.239.10.119)
15:46.25MikeJit shouldn't need the path to asterisk
15:46.44Kattygives eppigy a cookie.
15:47.00angryuserMikeJ, yes but this line during install :Looking for Asteris/CallWeaver directory in /usr/src ... and asking me "m : Enter Asterisk/CallWeaver path manually"
15:47.40MikeJhmmm
15:47.41MikeJweird
15:48.07angryuseri am trying to install lates beta, with stable its the same
15:48.24*** join/#asterisk SlipperyChicken (n=andrew@LONDON14-1168107385.sdsl.bell.ca)
15:48.31MikeJ3.5.2?
15:48.40angryuserMikeJ, yes
15:48.58aurax[TK]D-Fender i've added user context but it's the same, still cannot receive calls.
15:49.02angryuserAny way to omit this step ?
15:49.02MikeJthere are some tdmapi issues in 3.5.2 that should be fixed in the next couple of days, but that shouldn't effect you w/ asterisk
15:49.12[TK]D-Fenderaurax: And you're not showing me anything
15:49.31MikeJangryuser: there is a way to build with just the makefile w/o ./Setup, but i have never done it
15:49.33auraxyou want a pastebin of the user field?
15:49.44*** join/#asterisk |Cybex| (n=John@80.100.126.176)
15:49.48MikeJI have only done tdmapi builds that way.
15:49.56MikeJbut call their support, they should be able to help
15:50.00auraxhttp://pastebin.com/d1585a72e
15:50.29angryuserMikeJ, i am building Bri + zap, and i cant provide sources for a reason for this particular install
15:51.55[TK]D-Fenderaurax: Once again only little bits and pieces and no call debug.
15:52.23MikeJangryuser: when you try to build bri its looking for the ast src dir so it can build chan_woomera.so
15:53.04MikeJso says sangoma guys
15:53.21MikeJif you build not for bri it won't require
15:53.31MikeJbut then you will need to build chan_woomera by hand
15:53.32auraxokay [TK]D-Fender i have no clue how to debug it, you want a sip show debug peer on that specific peer?
15:53.56[TK]D-Fenderaurax: Where's the broken call?  You've shown me SIP DEBUG before.
15:53.58angryuserMikeJ, hm ok, it is looking for asterisk.h, ok thank you for info i will try Bri only
15:54.01MikeJangryuser: http://wiki.sangoma.com/sangoma-wanpipe-smg-asterisk-bri-installation#sangoma_smgbri_install
15:54.13auraxwhat do you mean ?
15:54.22Kattysteals eppigy for chimichanga run
15:54.36*** join/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net)
15:54.46auraxhmm
15:54.49eppigy:D
15:54.53*** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net)
15:54.55eppigyLETS GO
15:55.12angryuserMikeJ, Bri only without tdmvoice is still asking for asterisk.h :(
15:55.33MikeJyes.. bri will do it.. to build chan_woomera
15:56.30MikeJyou don 't have the ability to build a module for that version of ast?
15:56.40*** join/#asterisk lucasb (n=bussey@office.telifon.com)
15:56.52auraxgot confused
15:57.12angryuserMikeJ, well the script searches for asterisk.h, and well will not continue without it in
15:57.26MikeJyes.. it needs it
16:00.35angryuserMikeJ, anyway thank you for help !
16:00.40aurax[TK]D-Fender ?
16:05.04angryuserMikeJ, do you have any info on how can i build chan_woomera.so by hand ?
16:05.52[TK]D-Fenderaurax: Where is my pastebin with the failed call?  Where is your COMPLETE sip entry for it?
16:05.55MikeJlike any other out of tree module
16:06.02MikeJits just the one c file
16:06.09MikeJdoes astxs still work?
16:06.24MikeJit will also need access to those header files of course
16:07.09aurax[TK]D-Fender i send you both...
16:07.38[TK]D-Fenderaurax: http://pastebin.com/d1585a72e <-- not complete, and no failed call in there
16:08.22angryuserMikeJ, hm, will try
16:08.37MikeJangryuser: do you not have a src dir?
16:08.41*** part/#asterisk lasko|away (n=lasko@70.102.15.210)
16:09.12angryuserMikeJ, no i dont, its a trixbox pro, so i ahve the zaptel sources no the asterisk ones
16:09.26MikeJfunky..
16:09.34MikeJthey are under gpl..
16:09.42MikeJso you can request the src
16:09.57MikeJI thought they posted all their patches
16:11.32*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
16:11.32angryuserMikeJ, maybe they gave it a different name, i cant find it at their repos
16:12.00MikeJyour going to be sol if you can't get headers to build against
16:12.05*** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ceab63f9a.cpe.net.cable.rogers.com)
16:12.10MikeJunless they ship w/ chan_woomera, which i doubt
16:12.40virtualme123I've been looking at the Asterisk Manager Events, and noticed that there is a hangup cause that can sometimes be thrown of 0, anyone got any ideas as to what type of hangup this could be?
16:12.45MikeJthey do use all sangoma.. they should be able to help you with this, no?
16:13.06angryuserMikeJ, they ship a woomera driver but it is 3.3.6
16:13.25MikeJhmm
16:13.40angryuserand it is way outdated, many problems
16:14.06MikeJthat doesn't have the bri stuff for sure
16:14.30virtualme123In the asterisk file causes.h the various states are mentioned for a hangup but when it comes to hangup cause of 0 it says NOTDEFINED
16:14.35angryuserMikeJ, it has the bri stuff, but the changes since were Very important
16:14.40virtualme123Seems a bit vague ...
16:15.47angryuserMikeJ, need to go, thank yoi for help, i will request source from fonality
16:16.04MikeJvirtualme123: its not vague .. its not defined :D
16:16.19MikeJangryuser: actually.. do you have a chan_woomera.so already?
16:16.27angryuserMikeJ, yes
16:16.36MikeJoh.. well .. then this doesn't matter
16:17.03virtualme123... yep but how might one define it?
16:17.08MikeJjust point it at a regular ast src
16:17.17angryuserMikeJ, oh ok
16:17.25MikeJbackup the chan_woomera.so you have now ... not sure it might overwrite it
16:17.34MikeJif so.. replace back the one you ahve now
16:17.41MikeJchan_woomera should not have changed
16:17.48*** join/#asterisk jlnt (n=lank1@70.255.193.190)
16:17.58jlntanyone around?
16:18.05MikeJjlnt: nop
16:18.06MikeJe
16:18.15*** part/#asterisk peterosd1 (n=Peter@c83-255-69-68.bredband.comhem.se)
16:18.16angryuserMikeJ, but any compatibility problems may occur ? its an old one from 3.3.6
16:18.59angryuserMikeJ, ok many thanks
16:19.14jlntOuch ... error while writing audio data: : Broken pip
16:19.24MikeJangryuser: that mod shouldn't have changed
16:19.46angryuserMikeJ, i am off , thanks's
16:19.55MikeJnp
16:20.10jlntMikeJ: did you see the Ouch ... error while writing audio data: : Broken pip
16:20.26jlntwhat's the best way to fix this, it shows up when I run asterisk -gcvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
16:21.11*** join/#asterisk Tulga (n=chatzill@203.91.114.3)
16:21.15*** join/#asterisk Fernandojdk (n=kvirc@189.24.63.81)
16:21.36MikeJif it hurts......
16:21.42jlntlol
16:22.36TulgaHi guys. I'm trying to build audio book server. can I use asterisk for it? customer call number and listen audio book titles. then select one title and listen.
16:22.39jlntI reseated the Digium card
16:23.06MikeJits with a digium card?  call their support and get help?
16:23.08jlntTulga: that is possible with asterisk, you would just have to setup submenus
16:23.40Tulgajlnt: thanks, I'm planning to buy http://www.synway.net/ cards. is it work with asterisk?
16:23.59Tulgaor asterisk only work with Digium cards?
16:24.00jlntI didn't know if it's the card or whether it's an asterisk config problem with zaptel.conf
16:24.30jlntTulga: I don't know on the card specs, I am sure asterisk supports many different cards
16:24.34jlntyou just have to configure it
16:24.47Tulgaok
16:25.12TulgaI'm very new. so my good start point is asterisknow?
16:25.15leifmadsennever heard of those cards though
16:25.32Tulgaleifmadsen: never heard http://www.synway.net/?
16:25.41leifmadsenTulga: right -- never heard of them
16:25.59leifmadsenif you can't get support / installation help from the manufacturer, you're going to be very much out of luck
16:26.10leifmadsen(i.e. you won't get help from this room)
16:26.22Tulgaok. digium support is best?
16:26.28leifmadsen~bestquestions
16:26.34leifmadsen~thebestquestions
16:26.34infoboti heard thebestquestions is Whenever you ask a "what is the best..." or "who is the best..." type questions, you're asking for trouble, and possibly may be called a troll. These types of questions do not have answers. Your best bet is to rephrase the question as, "What kind of experience do people have with..." or "Who has experience with...".
16:26.57leifmadsenTulga: "best" is always relative -- however, if you want to support the project, I'd suggest a Digium card (plus I've had good luck with them when I've used them)
16:27.15Tulgaok :)
16:27.18leifmadsenTulga: additionally, you get phone support on getting the card setup if necessary
16:28.01TulgaI'm talking mobile operator companies. they suggest me E1 connection. is it work with asterisk?
16:28.04MikeJTulga: leifmadsen is the best
16:28.13leifmadsenlol
16:28.19leifmadsenTulga: yes, E1 and T1
16:28.24TulgaMikeJ: thanks. he sounds like best
16:28.26MikeJI have his number if you need to call him directly :D
16:28.41Tulgaif you give me number, you are best too
16:28.42Tulga:P
16:29.07Tulga1-800-SANTACLAUS ?
16:29.26leifmadsen1-800-ABCDEFG
16:29.33Tulgahehe
16:30.22MikeJoh weird.. I thought I had your number.. its not in this phone anymore :(
16:30.39Tulgaleifmadsen: I imagine operator server -> E1 -> Media converter to IP -> SWITCH -> media converter to E1 -> my card. right?
16:31.17MikeJTulga: you have a strange imagination
16:31.24leifmadsenyes...
16:31.26Tulgahaha
16:31.26MikeJit seems to have a lot of extra parts in it
16:31.34leifmadsenI don't quite get why you would go E1 -> IP -> E1
16:31.46leifmadsenwhy not just plug the E1 directly to the box?
16:31.53leifmadsenthat's kinda the purpose of the card....
16:31.56MikeJleifmadsen: are you still living near me these days?
16:31.58Tulgamy server is far
16:32.01[TK]D-FenderTulga: E! smartJack -> directly into Zaptel/DAHDI compatible E1 card in your server
16:32.03Tulgais it not problem?
16:32.06leifmadsenMikeJ: I'm in Toronto
16:32.21leifmadsenyou tell me :)
16:32.24MikeJI am planning a trip soon.. we should get drinks
16:32.39leifmadsenMikeJ: I like drinks!
16:32.44MikeJindeed
16:32.56TulgaI want drink. but I'm very far
16:32.57Tulga:D
16:35.21ariel_Tulga: I think we might need more info on what your doing. But why do you need 2 e1 cards ?
16:36.31*** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net)
16:36.41Tulgaariel_: 2 E1 cards? my operator company giving to me E1 connection. I have E1 card
16:37.55TulgaI don't understand E1 connection. how to be E1 cable to my office?
16:38.14ariel_Tulga: Tulgaleifmadsen: I imagine operator server -> E1 -> Media converter to IP -> SWITCH -> media converter to E1 -> my card. right?
16:38.29ariel_at your office you have an asterisk system?
16:38.34*** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
16:38.34Tulgayes
16:38.52ariel_ok at the operator location there only giving you an e1?
16:39.00Tulgayes
16:39.01[TK]D-FenderTulga: No need for an E1 card in your server if you're putting a media converter at the source.  * will take that IP connection in DIRECTLY
16:39.29ariel_Tulga: what [TK]D-Fender said
16:39.35Tulgayes
16:39.39Tulgait is possible
16:39.43Tulgathanks!
16:40.11TulgaE1 card expensive than IP card?
16:40.22[TK]D-FenderTulga: IP card?  HUH?
16:40.34ariel_I think he is confusing things
16:40.42[TK]D-FenderTulga: this is a stupid network connection.  5$ <-
16:41.00Tulgaare you talking network card?
16:41.05[TK]D-FenderTulga: Internet Protocol.
16:41.09[TK]D-FenderTulga: Do the math
16:41.15VaGoNeTaShi buddys
16:41.23VaGoNeTaSi gut a little issue
16:41.25Tulgaasterisk can work without any card?
16:41.26VaGoNeTaSi got*
16:41.29VaGoNeTaShttp://pastebin.ca/1457943
16:41.36VaGoNeTaSi can get my calls registered on the CDR
16:41.40*** join/#asterisk jnfuller (n=jnfuller@99.199.170.110)
16:42.04VaGoNeTaSis weird, mysql is running
16:42.22VaGoNeTaSmy configs on the cdr_odbc.conf , res_odbc.conf , odbc.ini , and odbcinst.ini, are ok
16:42.40jnfullerHi there... Is there a setting in any of the conf's that affects the display of x-asterisk headers in dialogues or is that something that would have to be hacked in the code to disable in 1.6.x?
16:42.41ariel_Tulga: yes depending on what your doing with it.
16:43.01VaGoNeTaSi created the correct database, and table
16:43.03Tulga[TK]D-Fender: ariel_: you suggest me install asterisk on simple PC (without any card). then connect my server to operator company by IP connection
16:43.16Tulgaright?
16:43.58[TK]D-FenderTulga: You said location A will have an EI>IP converter.  All you need on your Server is an IP connection to it
16:44.31[TK]D-Fenderjnfuller: Disable display of headers? why?
16:44.36VaGoNeTaS:s
16:44.39TulgaI'm trying to create audio book system. user call to my number (7777), then listen audio books title. select title, listen audio book
16:44.52jnfullerwell, rebranding the useragent for one
16:45.13jnfullerX-Asterisk is hardcoded in chan_sip.c
16:45.32jnfullerso short of changing the code, it's better just to turn them off
16:45.51[TK]D-Fenderjnfuller: Changing code is the only way to do taht
16:46.56*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
16:47.48jnfullerThat was what I was thinking as well. I was hoping for a way to just "turn off" those headers but the logic doesn't look like that's even possible.
16:51.19Tulga[TK]D-Fender: can I record customer voice when he call me by asterisk and IP connection?
16:53.37[TK]D-FenderTulga: Go learn how * works.  You can process calls however you feel like with the tools the dialplan gives you
16:54.29Tulgacan I test it with asterisknow?
16:55.13[TK]D-FenderTulga: Please understand that your control when installing a GUI is greatly limited by the ideas it lets you implement
16:55.27[TK]D-FenderTulga: We recommend you actually lear * yourself bafore going that route
16:56.15Tulgaok. tomorrow I'll try it. thank you very much
16:56.47jnfullerthere is always: http://lmgtfy.com/?q=asterisk+record+incoming+calls
17:01.27plundraHmm, I did a blind transfer on a hardphone, when will it give up calling the "target"?
17:02.21[TK]D-Fenderplundra: It doesn't
17:02.40[jmc]hmm
17:02.47[TK]D-Fenderplundra: Blind transfer = GTFO and its up to the dialplan it calls to do "whatever"
17:03.01[jmc]can I connect through SIP to * with a softphone running on the same machine?
17:03.21[jmc](...the same as * I mean)
17:03.29jnfullerthx for the info [TK]D-F re: X-Asterisk, looks like I'll be needing to do an internal fork to do what I want
17:03.35plundra[TK]D-Fender: Ah, yes, of course! :-D It uses my extension for dialing out.
17:04.08plundraThanks
17:04.19jnfullerbye all
17:05.15[jmc]or do I need to connect from a different machine?
17:07.13*** join/#asterisk juancferrer (n=chatzill@hydro-hawk-dsl.netdoor.com)
17:07.31[jmc]:(
17:08.44*** join/#asterisk aaronorosen (n=arosen@frog22.cs.clemson.edu)
17:10.23juancferrerIn a dialplan I call a fastagi script that sets some variables x1=foo, x2=bar, x3=baz...  I want to access them by doing something similar to (${x}${NUM})... is there a way I could do something like that?
17:12.28*** join/#asterisk Iamnacho (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
17:14.24*** join/#asterisk green-monkey (n=ericshel@70.102.50.18)
17:15.10juancferrerI've been using startpy for  fastagi, and in my script I access a remote DB and get some values and read them out with Festival.  I've realized that its a bad thing to use Festival in the fastagi script because it takes way too long (http://www.amoocon.de/talks/28).  Now I'm just trying to set a bunch of variables with the values needed, and in the dialplan I just have a while loop that...
17:15.11juancferrer...calls festival a bunch of times.
17:17.08juancferrersorry, starpy...http://www.vrplumber.com/programming/starpy/
17:17.19*** join/#asterisk jong2 (n=chatzill@65.100.10.89)
17:18.14*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
17:20.20*** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net)
17:21.08jong2yo anyone bored?
17:22.16Qwelljong2: you're better off asking a real question
17:22.23jong2hehe
17:22.45jong2we just put out beta software  rc.
17:22.57jong2needs some feedboack from people.
17:23.10jong2i mean feedback.
17:23.34jong2can i put out blatant shameless advertisement here?
17:23.36*** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-149.nys.biz.rr.com)
17:24.33Qwelljong2: you'll be booted if it's inappropriate for this channel.  up to you
17:24.38jong2http://www.safisystems.com/index.cfm/2009/6/11/SafiWorkshop-and-SafiServer-Release-Candidates-now-available-for-download
17:24.45[jmc]guys
17:24.48[jmc]if I use
17:24.55jong2it is visual call flow designer for asterisk.
17:24.57[jmc]"console dial <MyExtension>"
17:25.12jong2any one bored should be good entertainment.
17:25.19[jmc]I can hear the voice talking smoothly
17:25.24[jmc]through OSS
17:25.37[jmc]if I call my SIP number from a real phone
17:25.42[jmc]the voice is lagged
17:25.56[TK]D-Fender[jmc]: and "real phone" gets to * HOW?
17:26.02[jmc]is it just because of a network lag?
17:26.07*** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net)
17:26.18[TK]D-Fender[jmc]: And where is your "SIP phone"?  What model?  What codecs?
17:26.19[jmc][TK]D-Fender: I have a phone number assigned to the SIP account
17:26.45[jmc]I call the number, the SIP provider calls my SIP number
17:27.05[jmc]my * is registered to the SIP provider and answers the call as if it was a SIP softphone
17:27.22[jmc]I don't know if that was clear enough
17:28.36[jmc]I just can't understand why that lag
17:28.58[jmc]if it depended on me only, I wouldn't even care
17:29.12[jmc]but I need to give a "demonstration" through that number
17:29.36[jmc]a chunky voice is not exactly what I had in mind :D
17:29.51[jmc]can someone point me to something useful?
17:31.32[jmc]another thing that confuses me
17:31.39[jmc]is that before I tried Asterisk
17:31.53[jmc]I used QuteCom (WengoPhone) to register to the same SIP account
17:32.15[jmc]I could hear myself speaking without lags
17:32.38juancferrerI've had something similar....calling the asterisk box locally through the lan, everything is fine, but through a pstn it lags sometimes.  I think it was a nat issue for me.
17:32.39[jmc]Asterisk does register correctly to the SIP account
17:32.40*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
17:32.43[jmc]but it lags
17:36.06[TK]D-Fender[jmc]: What provider, what codecs, same for your phone, what is it, Where is it, describe the networking.
17:36.15[TK]D-Fender[jmc]: You offered NO details.
17:36.28[jmc][TK]D-Fender: pardon me, I'm quite a newbie
17:36.39[jmc]I'll try giving you something more
17:37.01[jmc]my ISP is Fastweb, my SIP provider is Messagenet
17:37.19[TK]D-Fender[jmc]: I told you to give me these specific details.  twice now.  No amount of "newbness" makes "what model of SIP phone and what codec" any less explicit
17:37.27[jmc]I connect from home
17:37.43[jmc]there's no SIP phone
17:37.50[jmc]it is a standard phone
17:37.59[TK]D-Fender[jmc]: Where is * relative to your phone?
17:38.10[jmc]they're both here at home
17:38.42[TK]D-Fender[jmc]: what is this "standard phone", and how is it conencted to *?
17:39.11[jmc]"standard phone" in the way that is not a SIP hardphone
17:39.39[jmc]is a common telephone, the one you usually find in "normal people's" houses
17:40.08[jmc]my SIP provider gives me a free "geographical number" which I can call from a standard phone
17:40.34[TK]D-Fender[jmc]: How many parties are involved in this call?
17:40.35[jmc]when it receives a call, it is redirected to my SIP account, if a client is registered to answer it
17:40.39*** join/#asterisk jsolis (n=Jimmy@200.121.196.92)
17:40.45[jmc][TK]D-Fender: my phone and *
17:41.00[TK]D-Fender[So really just * to your SIP provider?
17:41.06[jmc]* registers to the SIP provider and answers the call
17:41.15[TK]D-Fender[jmc]: No second phone for 2 actual people talking together?
17:41.22[jmc]well, yes, if there isn't something I don't know about in the middle
17:41.33[jmc]no, I'm trying to build an IVR
17:42.22[TK]D-Fender[jmc]: What are you using as a basis of comparison for lag?  What codec is your call arriving in?  What is your internet connection?
17:42.50tompawhi
17:42.52Kattyhoray!
17:42.57[jmc]hi :)
17:43.02Kattyhas cut & style appt at 2:30!
17:43.06[jmc][TK]D-Fender: hmm that's the bad part I guess
17:43.07tompawwhat is the currently preferred way to set up a chain of sip failover trunks?
17:43.26[jmc][TK]D-Fender: I don't have enough resources for a lag comparison
17:43.43[jmc]but other people calling from "standard phones" have the same lag on my number
17:43.53[TK]D-Fender[jmc]: I mean what are you listening to / recording from to say that there is lag if you're not talking to an actual person?
17:43.59[jmc]I've set up a SIP user and I connect to my * with Twinkle
17:44.09[jmc]and from Twinkle there's no lag
17:44.13[TK]D-Fender[jmc]: Does Twinkle lag?
17:44.35KattyTwinkle Twinkle, little sip phone.
17:44.35[jmc]no, but Twinkle is on my machine
17:44.35[TK]D-Fender[jmc]: Ok, so what codec?
17:44.35[jmc]the same as *
17:44.36[jmc]hmm let me see
17:44.42Kattyhow i wish i could work from my home
17:44.42[jmc]alaw IIRC
17:44.44[TK]D-Fender[jmc]: But your machine is right next to your server effectively, isn't it?
17:44.49Kattyup a above the counter so high.
17:44.54Kattyi need that pie pan to make my pie
17:44.56[jmc][TK]D-Fender: actually, it IS my server :D
17:45.13[TK]D-Fender[jmc]: Ok...
17:45.19[jmc]alaw
17:45.26Kattyyourlaw
17:45.31[TK]D-Fender[jmc]: No reason it should be any different with *...
17:45.36[jmc](in my sip.conf I disallow=all and then allow=alaw)
17:45.39[TK]D-Fender[jmc]: Unless.. are you running a VM on it?
17:45.46[jmc][TK]D-Fender: that's what puzzles me :|
17:45.49[jmc]no, I'm not
17:46.06[jmc]ok
17:46.17[TK]D-Fender[jmc]: So what are you listening to/recording to give you the impression of "lag"?
17:47.01[jmc]I'm listening to the registered voices that come with *
17:47.12[jmc]Playback(demo-congrats), etc...
17:48.40[TK]D-Fender[jmc]: Ok, next : go look in your sounds folder and see if you have ".alaw" versions of those sounds.  You may have a serious trancoding issue.
17:49.18[TK]D-Fender[jmc]: The default option on install only gives you .gsm files.  Compiler optimizations can cause issues with this.
17:49.22[TK]D-Fender~gsmbug
17:49.22infobot[~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read :  http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39
17:49.24[TK]D-Fender^^6
17:49.37[jmc]hmm
17:49.47[TK]D-Fender[jmc]: What distro?
17:49.51[jmc]how can I check which compiler did compile my *?
17:49.53[jmc]Debian
17:50.02[jmc]Debian Squeeze, fwiw
17:50.11[jmc]maybe it's right gcc 4.2
17:50.17[TK]D-Fender[jmc]: Check your kernel clock as well.. IIRC Debian often uses 250 where * needs 1000
17:50.34[jmc]how can I do that?
17:50.37tzafrir_laptopgcc 4.2? more likely 4.3
17:50.41[TK]D-Fender[jmc]: check your sounds.  download .alaw versions.  Test.  if thats OK, then we've pinned it down.
17:51.18tzafrir_laptopon Debian you should just use the system copy of libgsm, which is not buggy like the one included with Asterisk
17:51.48[jmc]hmm, you may be right
17:51.53[jmc]I only have .gsm sounds
17:51.58eppigyI had a calzone
17:52.06[jmc]calzone? good :D
17:52.09eppigyYES
17:52.47mgromanya i just wrote a cool python scrippt w00t ! ya!
17:52.52mgromanhigh fives
17:52.58mgroman* cricket chirps *
17:53.03[jmc]Python is nice ^^
17:53.25juancferrer<3 python
17:54.02juancferrerand twisted is just amazing
17:54.20jsolishey guys what is the version of SIP on asterisk 1.6 and asterisk 1.4 exist any diference?
17:56.06mltlnxcan someone shed some light on parkinglots in features.conf. I am trying to use multiple parking lots
17:59.38Greek-Boywhat would cause segmentation faults on asterisk startups on most modules?
17:59.40*** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net)
18:00.10*** part/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net)
18:02.37*** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net)
18:02.51*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
18:02.55[TK]D-Fenderjsolis: There are all sorts of docs in the tarball.  Go read them
18:05.22[jmc][TK]D-Fender: you were right
18:05.34[jmc]it was a gsm => alaw transcoding issue
18:05.37[jmc]:D
18:05.41[jmc]god, it's working
18:05.44[jmc]I can't believe it
18:05.46[jmc]:')
18:05.51[jmc]sheds a tear
18:06.45VaGoNeTaSthere is an special module to load cdr and mysql?
18:06.48VaGoNeTaS<PROTECTED>
18:07.06VaGoNeTaSi have installed on the server, unixodbc, odbc
18:07.24VaGoNeTaSmysql is working, i've just created the database, the tables, but it doesnt works
18:07.34VaGoNeTaSi cant get my calls registered on the cdr
18:08.34[TK]D-Fender[jmc]: You're welcome.
18:10.45*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
18:15.33*** join/#asterisk felipe_ (n=felipe@my.nada.kth.se)
18:22.40kaldemarVaGoNeTaS: have you defined asterisk2 in /etc/odbc.ini?
18:23.40kaldemarVaGoNeTaS: and the database driver in /etc/odbcinst.ini?
18:25.12VaGoNeTaSyes
18:25.12VaGoNeTaSi did
18:25.22VaGoNeTaSi belive that i need an module or something
18:25.32VaGoNeTaScoz when i try to do "cdr show status"
18:25.52VaGoNeTaSNo such command 'cdr show status' (type 'help cdr show status' for other possible commands)
18:26.08kaldemarprobably cdr_addon_mysql from asterisk-addons
18:26.15VaGoNeTaSyes
18:26.22VaGoNeTaSim trying to install it
18:26.30ariel_*CLI> cdr mysql status
18:26.43VaGoNeTaSXXX 1.  app_addon_sql_mysql
18:26.49VaGoNeTaSDepends on: mysqlclient(E)
18:27.01VaGoNeTaSi have it already installed
18:27.21*** join/#asterisk WHYS (n=drumm@137.28.94.209)
18:28.03*** join/#asterisk lanning (n=lanning@173.8.187.197)
18:28.54*** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no)
18:28.58VaGoNeTaS:s
18:29.03VaGoNeTaSi have asterisk 1.4.25.1
18:29.17VaGoNeTaSand asterisk addons 1.4.8
18:30.13VaGoNeTaSmysql-client-5.0 is already the newest version.
18:30.22VaGoNeTaSmysql-client is already the newest version.
18:30.40VaGoNeTaS:s
18:32.28*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:34.01eppigylibmysqlclient
18:34.08QwellDid you run ./configure after installing the mysql development package?
18:34.11eppigyor mysqlclient-devel
18:34.23ariel_did you install the mysql stuff before you setup the asterisk adons
18:34.40*** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net)
18:34.47VaGoNeTaSjust a second gys
18:34.49VaGoNeTaSguys*
18:34.57jjshoeis there a command which will show me which all apps are loaded currently? chan drivers as well
18:35.30VaGoNeTaSit was libmysqlclient15-dev missing
18:35.58eppigyyes
18:36.07*** join/#asterisk micols (n=mio@rlogin.dk)
18:36.42ariel_jjshoe: like show applications
18:37.15*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
18:37.15*** mode/#asterisk [+o Deeewayne] by ChanServ
18:37.35jjshoeariel_ something like that..
18:38.12jjshoehrm
18:38.37[TK]D-Fenderjjshoe: "core show modules"
18:39.34ariel_did they not change that to modules show
18:39.44eppigyTRABAJO
18:39.47*** join/#asterisk spck (n=spck@unioncab.com)
18:39.54spck~addqueuemember
18:39.55ariel_what about work
18:40.25spckgot a question about addQueueMember, http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember
18:40.47spckon that page it gives this example: AddQueueMember(queuename[,interface[,penalty[,options[,membername[,stateinterface]]]]])): (1.6+)
18:40.56jjshoe"show modules" did it for me
18:41.08spckwhat does it mean by membername and what is it used for?
18:41.41jjshoehrmph. chan_dahdi.so has decided to no longer load
18:41.42jjshoewtf
18:42.02jayteeNO TRABAJO, NO TENGO ENERGIA!
18:42.02VaGoNeTaSis still unable to connecto
18:42.13KavanSno travajo?
18:43.12ariel_more like: no queiro trabajar, pero si nessisito denero tengo que trabajar.
18:43.37jayteesi él utiliza Trixbox entonces él es un hijo de puta
18:43.38jjshoehrm, something is fubar here.
18:43.55[TK]D-Fenderspck: Thats the name that will appear in the queue logs for the agent
18:46.31jjshoewhat would cause
18:46.41jjshoewhat would cause "dahdi show channels" to not exist via the cli?
18:48.57jjshoeconfig issue
18:48.58jjshoefound it
18:53.22bkw_looks around!
18:53.47Qwellbkw_: ?
18:54.05bkw_Qwell: what up man?
18:57.46*** part/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net)
18:58.27mltlnxAny one using multiple parking lots in 1.6.1?
19:01.37*** join/#asterisk sah-work (n=Bawbatos@65-119-47-100.dia.static.qwest.net)
19:01.56sah-workdumb question maybe.
19:02.00sah-worki have a queue_log file
19:02.04sah-workis there something to turn that into a human readable report
19:02.20bkw_perl?
19:02.40sah-workso nothing canned
19:03.18sah-workthe first part 124824224|1244824200 what time format is that?
19:03.51Qwellunix time
19:03.55sah-workcool
19:04.20Qwell...probably
19:04.35Qwelllooks like you missed a digit or something though
19:04.48sah-workyah. just typed not cut & paste
19:06.22sah-workugh. so no canned report tool :( great
19:07.08[TK]D-Fendersah-work: What do you mean "no canned tool"?
19:08.00sah-worki have a queue_log file and i want to turn it into something boss readable
19:08.06[TK]D-Fendersah-work: there are plenty of queue log analysis programs out there
19:08.30[TK]D-Fendersah-work: And its a petty task to decode the unix timestamp anyway
19:09.17sah-workokay, that was what i was asking. any recommendations of a program to analyze this file
19:11.08*** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve)
19:11.23[TK]D-Fendersah-work: Go look on thw WIKI, there are all sorts of links
19:11.27[TK]D-Fender~wikis
19:11.27infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
19:12.02sah-workalready there. i was hoping for a quick use this as i need to get this done fast
19:12.17sah-worki am replacing this system tomorrow with a new install
19:13.05*** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve)
19:14.08mltlnxHas any one at all been able to get multiple parkling lots working? No matter what I do the default parkinglot is chosen
19:17.17batphonei have a dell 1750 that doesnt have a power cord for the my tdm card
19:17.23*** join/#asterisk talirk81 (n=talirk@rrcs-67-78-39-22.sw.biz.rr.com)
19:17.24batphonedoes anyone know if they make a pci power card?
19:17.46batphonei have a firewire card i could run 12v back into the chassis with ...
19:17.53Qwellbatphone: yep, hang on
19:17.54*** join/#asterisk telecos (n=sergio@91.166.219.87.dynamic.jazztel.es)
19:18.11talirk81Can someone look at my output from asterisk @ http://pastebin.ca/1458121  I have  a verbose sending to the log what is it about to send to an EXEC command but no sound is playing
19:18.27Qwellbatphone: http://store.digium.com/products.php?category_id=39
19:19.26Qwellbatphone: alternatively, you can get SATA-style adapters that go to molex
19:19.38Qwell(assuming you have spare SATA-style power cables)
19:20.36talirk81btw i had a similar issue when i tried  SAY ALPHA instead
19:22.09batphoneQwell: thanks man
19:22.31batphoneQwell: the 1750 components get power from backplanes and such
19:22.44batphoneQwell: do you know what the amp draw is on a TDM card?
19:22.49Qwellbatphone: the former would probably be better then
19:23.06Qwellno clue
19:23.12batphonei guess that value depends on the length of the wire
19:23.48batphonewell, for the $110 i spent on that piece of gear i could be half way to a whole new server and use this dual xeon for something else
19:24.16plundraWhy do the one who calls, always get a beeping/tone (whats the term?) after a hangup? The phone always gets that the call has ended (both from a landline and internally from a sip-phone), but always with that beeping/buzzing too.
19:24.52plundraI tried setting an argument to my hangup(), without any change.
19:25.28talirk81plundra, i had that issue and i have to change my rfc type to address it at one point
19:26.32plundratalirk81: Regarding dtmf?(!)
19:27.03talirk81yeah i think i had to set  dtmfmode=rfc2833
19:27.42talirk81i also have rfc2833compensate=yes set btw
19:27.51mltlnxmultiple parking lots?
19:28.48plundrarfc2833 is supposed to be default
19:29.00bkw_mltlnx: http://www.freeswitch.org/asterisk_stuff/app_valetparking_1.2.c
19:30.01mltlnxbkw_: Thanks, but I am try to implement the multiple parking lots from 1.6.1. This is added and somewhat documented in features.conf
19:30.25Kobazanyone have any idea what voip.ms uses as their provider?
19:30.36plundraIt feels like it should close the channel or something, since the phone still is active. Or something :-P Hmm
19:32.35mltlnxbkw_: My irc a broken...Did you send me a message
19:32.46bkw_jes
19:32.57mltlnxwhere it went i dont know?
19:33.02bkw_haha
19:33.05bkw_happens to me all the time
19:33.23*** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be)
19:33.25mltlnxI saw it show up and then..........gone
19:34.10*** join/#asterisk seanmh (n=johndoe@198.59.129.24)
19:36.20WHYSAnyone using ABE on rpath?  The admin throws error right out of the box
19:39.12spckin the addqueuemember function what is an use of the stateinterface parameter?
19:39.22*** join/#asterisk techie (n=techie@adsl-76-214-9-192.dsl.lsan03.sbcglobal.net)
19:39.23*** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net)
19:39.46spcklike if you have two clients (1 soft, 1 hard) on the same interface?
19:40.53spcker rather for the same person?
19:41.45*** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com)
19:42.42*** join/#asterisk andres833 (n=andres83@190.144.75.22)
19:43.52andres833hi all
19:46.59*** join/#asterisk Greek-Boy (n=greek@41.222.89.77)
19:49.20*** join/#asterisk alrs (n=lars@dsl211-216-141.lax1.dsl.speakeasy.net)
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19:54.43[TK]D-Fenderspck: What device to check to see if the final "target" is to be considered busy
19:56.44ariel_WHYS: ABE I think the original login is admin/digium
19:57.56*** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br)
20:03.38andres833i need know what is a form that using the same extension in a 2 direferents phones and that the first phone register after register the second phone the fisrt phone unregistert
20:03.59VaGoNeTaSfuck i cant make this shit working
20:04.19VaGoNeTaSi've installed the cdr_addon_mysql from asterisk-addons
20:04.29VaGoNeTaSbut asterisk doesnt enable the module
20:04.48VaGoNeTaSwhere should i enable the module cdr_addon_mysql.so ?
20:04.55VaGoNeTaSwhat config file
20:05.24andres833VaGoNeTaS, you talk with me
20:05.46*** join/#asterisk propellerhead (n=yogurt2u@host223.190-225-214.telecom.net.ar)
20:05.49*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
20:07.30VaGoNeTaSno , but if u can help me i will appreciate that
20:09.58VaGoNeTaSdatapartner-asterisk*CLI> cdr mysql show
20:09.59juanIMPhave you try "load cdr_addon_mysql.so" VaGoNeTaS , the load in modules.conf load =>  cdr_addon_mysql.so
20:09.59VaGoNeTaSNo such command 'cdr mysql show' (type 'help cdr mysql show' for other possible commands)
20:10.03VaGoNeTaS:(
20:10.16juanIMPthen*
20:10.41Greek-BoyVaGoNeTaS: Which version of asterisk are you running?
20:11.04VaGoNeTaSlast one
20:11.06VaGoNeTaS1.4.25.1
20:11.20VaGoNeTaSdahdi is 2.1.0.4
20:11.30VaGoNeTaSasterisk-addons is 1.4.8
20:11.42VaGoNeTaSi've compiled it with cdr_addon_mysql support
20:13.55juanIMPVaGoNeTaS: load in modules.conf " load =>  cdr_addon_mysql.so" and CLI>load cdr_addon_mysql.so ....... whats the problem?
20:14.22andres833i need know what is a form that using the same extension in a 2 direferents phones and that the first phone register after register the second phone the fisrt phone unregistert
20:14.41VaGoNeTaSit doesnt do anything
20:14.41VaGoNeTaSdatapartner-asterisk*CLI> module load cdr_addon_mysql.so
20:14.42VaGoNeTaSdatapartner-asterisk*CLI>
20:14.50VaGoNeTaSdoesnt load the module properly
20:18.20VaGoNeTaSi dont know what's wrong
20:18.21VaGoNeTaS:s
20:18.59VaGoNeTaS<PROTECTED>
20:19.01VaGoNeTaS<PROTECTED>
20:20.42bkw_well cdr_odbc is not cdr_addon_mysql
20:20.53VaGoNeTaSwhat is it then?
20:21.06bkw_cdr_odbc: Error SQLConnect -1
20:21.09bkw_you have both loaded
20:21.10VaGoNeTaSi've just created the database called cdrdb
20:21.16bkw_but cdr_odbc is loaded and not configured
20:21.21bkw_so don't load cdr_odbc
20:21.23VaGoNeTaSi've created the cdr table
20:21.33VaGoNeTaSbkw_ : it is configured
20:21.42VaGoNeTaSim gonna pb all my configs related to cdr
20:21.54bkw_cdr_odbc is configured?
20:22.00bkw_then why are you talking about cdr_addon_mysql?
20:22.02VaGoNeTaSyep, just gimme 1 min
20:22.20VaGoNeTaScoz somebody here told me that i have to have cdr_addon_mysql in order to make cdr works
20:24.03bkw_cdr_odbc can do it also
20:24.33*** join/#asterisk bullium (n=RAIN\wbr@216.68.250.30)
20:24.34VaGoNeTaSbuddy
20:24.35Aiatekdid you installed mysql-devel?
20:24.35VaGoNeTaShttp://pastebin.ca/1458190
20:24.39VaGoNeTaSyes, i did
20:25.41VaGoNeTaSin the pb is my config files
20:25.47bulliumhow do you do setup a bridged appearance? (phone 1 see's when phone 2 is getting a call and can answer it if need be)
20:25.48VaGoNeTaSthey seems to be ok right?
20:28.30VaGoNeTaS:s
20:30.41*** join/#asterisk jjshoe (n=jjshoe@h69-129-142-83.mdsnwi.tisp.static.tds.net)
20:30.42jjshoeanyone know by chance what needs to be set in a polycom to set the digeset username?
20:31.09*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
20:32.21bkw_jjshoe: what do you mean?
20:33.10[jmc]anyone knows a good Java/JavaFX/Flash webphone?
20:33.15[jmc](SIP)
20:33.45batphoneQwell: i rigged up another ATX power supply to it ;)
20:33.56batphoneQwell: jumped the power-good to ground with two nails and a paper clip
20:34.13batphoneQwell: there will need to be extensive wiki documentation on 'how to troubleshoot the pbx'
20:34.15M1s3ry[jmc], didn't xfighter send you a link yesterday?
20:34.28[jmc]M1s3ry: yes
20:34.30batphone1) ensure paper clip is making contact with both nails
20:34.35jjshoebkw_ there's a config option I'm missing in the polycom config, so it'll send the username in the digest portion
20:34.36[jmc]an ActiveX one though :D
20:34.41M1s3ryah
20:34.42[jmc]it's not what I need
20:34.43batphone2) power on the PBX
20:34.50jjshoebkw_ Authorization: Digest username="", realm="asterisk", nonce="5f16cfa4", uri="sip:10. for ex.
20:34.51batphone3) unload wctdm,zaptel modules in that order
20:34.55batphone...
20:39.11*** join/#asterisk jeffgus (n=jeffgus@green.zimage.com)
20:41.42jjshoebkw_ some config option or another is used to fill out the username portion of the digest, just trying to find out which
20:42.28*** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58)
20:42.30*** join/#asterisk jsolis (n=Jimmy@190.43.77.200)
20:42.47jjshoehrm I wonder if it's retardly the label option
20:43.08jsolishey guys , how can i change the time GMT to GMT - 5 in the Master.csv is possible?
20:43.42*** join/#asterisk Iamnach0 (i=Iamnacho@ip174-70-137-120.ks.ks.cox.net)
20:45.20jsoliscan i set the time GMT-5 in the cdr.conf ?
20:45.42[jmc]guys, for MOH
20:45.48[jmc]can I use any MP3?
20:46.00[jmc]do I just have to copy to the MOH directory and set it to playback?
20:46.05keith4jsolis: I think the idea is to keep times in GMT, internally, and convert to the local timezone when displaying
20:46.13[jmc]or do I have to convert it to a certain rate?
20:46.25keith4[jmc]: http://www.voip-info.org/wiki/view/Asterisk+cmd+MusicOnHold
20:46.38[jmc]thanks ;)
20:46.43keith4check the "See also" section
20:48.37[jmc]just seen that, thanks :)
20:48.40[jmc]great resource
20:49.51jsolisthank you keith ;)
20:57.03*** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net)
20:57.46DefrazHow can I force an extension to unregister.
20:58.43DefrazI had two devices register with the same extension by accident but even though I registered one of them to a differenet extension it still shows it registered to the old extension and calls still go to that device for some reason when there pointed to that extension.
20:58.55bkw_jjshoe: thats easy reg.1.auth.userId=
20:58.56Defrazthought I used to use sip unregister but doesn't seem to be there.
20:59.44jjshoebkw_ hrm, I'll have to look, swear I have that mofo set though.
21:03.31*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
21:04.19*** part/#asterisk MikeJ (n=MikeJ@freeswitch/developer/mikej)
21:05.14jsolishey guys
21:05.25[jmc]hi
21:05.43jsolisif i set with usegmtime=no in cdr.conf
21:05.52jsoliswhat will happend?
21:06.46jsolissorry for english :(
21:08.09[jmc]I don't know
21:08.10[jmc]:d
21:08.10*** join/#asterisk marv[work] (n=timr@router.asteriasgi.com)
21:08.12[jmc]:D
21:08.53jsolis:(
21:09.15jsolisif usegmtime=yes set the time at GMT
21:09.17jsolisbut
21:09.30jsolisusegmtime=no set the time at UNIXTIME???
21:11.19*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
21:13.06*** join/#asterisk jong2 (n=chatzill@65.100.10.89)
21:13.32KavanShow can I "reset" a zap channel?
21:17.33jjshoehang it up?
21:19.54KavanSn/m...used zap restart in console
21:19.59KavanSthat seemed to have worked
21:22.48VaGoNeTaSwhat's wrongggggggggggggg
21:23.04jjshoebkw_ yup, have that.
21:25.24*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
22:18.47BlargMaN00in MeetMe(), what is the difference between option 'A' and option 'a'??
22:19.33Qwell<PROTECTED>
22:19.33Qwell<PROTECTED>
22:19.48BlargMaN00i got that, but what is the difference between them???
22:20.04Qwellone is an admin, one is a marked user?
22:27.40*** join/#asterisk ruben23 (n=RPL@124.107.3.178)
22:28.08ruben23hi nayone have idea on this error:  Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)
22:29.03ruben23i check the directory got this http://pastebin.com/m7c876e4
22:29.11juanIMPtake a look at the messages when asterisk -vvvc
22:32.50ruben23juanIMP: it says its ready
22:32.56ruben23asterisk ready..
22:33.00[jmc]ok
22:33.06[jmc]that's fine then
22:33.12[jmc]if you run asterisk in the background
22:33.23[jmc]you have to "reconnect" to it via that file
22:33.34ruben23juanIMP: whats the purpose of that command and why did that happen.
22:33.59[TK]D-Fenderruben23: It happened because * wasn't running
22:34.23[jmc]guys, new question for you :)
22:34.27ruben23<PROTECTED>
22:34.37ruben23getting the same error
22:34.42ruben23again
22:34.47[jmc]ruben23: if you stop it
22:34.51[TK]D-Fenderruben23: because its STILL not running
22:34.57[jmc]then you can't "reconnect" to nothing
22:34.58[TK]D-Fender(as a daemon)
22:35.06[jmc]exactly
22:35.17[jmc]you need asterisk to be run in the background, as a daemon
22:35.32[jmc]and then you use -r to talk to that one instance running in the background
22:36.29[jmc]ok, now for my question :P
22:36.31ruben23yeah got it now guys thanks
22:36.41[jmc]when I get to the asterisk console
22:36.53[jmc]and I 'core show file formats'
22:37.12[jmc]there is "slin       mp3        mp3"
22:37.27[jmc]so I defined in musiconhold.conf the section
22:37.32[jmc][mymusic]
22:37.35[jmc]type=files
22:37.44[jmc]directory=/path/to/my/music
22:37.59[jmc]when * answers the call it does
22:38.21[jmc]exten => number,1,Answer()
22:38.29[jmc]exten => number,1,SetMusicOnHold(mymusic)
22:38.35[jmc]exten => etcetera
22:38.47[jmc]oh sorry, it was
22:38.50[jmc]exten => number,2,SetMusicOnHold(mymusic)
22:38.51*** join/#asterisk StephenF (n=none@198.144.201.106)
22:38.58[jmc]after setting MOH
22:39.04[jmc]I do a Wait(5)
22:39.08StephenFWhats the best TAPI connector for Asterisk, supporting Windows Vista??
22:39.11*** join/#asterisk denon (i=denon@synapse.subneural.net)
22:39.11*** mode/#asterisk [+o denon] by ChanServ
22:39.23[jmc]the call then waits for those 5 seconds
22:39.28[jmc]but without any music
22:39.35[jmc]is it normal?
22:39.47[jmc]do I have to use a different application?
22:40.17*** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net)
22:41.56StephenFAnybody using a TAPI connector for Asterisk?
22:42.13[jmc]hmm if I use MusicOnHold() instead it "works"
22:42.22[jmc]but only for, like, one second
22:45.04ruben23[jmc]..?
22:45.19[jmc]yes?
22:45.39ruben23how do is set asterisk to run again upon reboot, coz when i reboot again i get the same error
22:46.19[jmc]ruben23: which distro are you running_
22:46.20[jmc]?
22:46.42ruben23centos
22:46.49*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
22:48.42[jmc]ruben23: chkconfig asterisk
22:48.42[jmc]I don't recall the exact syntax
22:48.43[jmc]check the manual
22:56.26*** join/#asterisk ikks (n=ikks@190.144.69.234)
22:57.00[jmc]well that's strange
22:57.28[jmc]when I do a 'console dial <Extension>' it works
22:57.35[jmc]a little choppy but it goes
22:57.52[jmc]when I call through Twinkle I hear a little bit of music
22:57.58[jmc]and then it silences itself
22:59.36[jmc]it does even when I put it on hold and I start it again
23:06.28[jmc]no clue? :(
23:13.06*** join/#asterisk hesco (n=hesco@24.99.160.121)
23:15.48hescohow do I ensure that a zaptel timer is running on my server?
23:16.11hescoI see nothing in ps list about it
23:16.12*** join/#asterisk Greek-B0y (n=greek@63.100.198.2)
23:17.13[jmc]lsmod | grep -i 'zaptel|ztdummy'
23:17.19hescothanks
23:18.47*** join/#asterisk Greek-B0y (n=greek@41.221.43.250)
23:18.57hescook, that gives me nothing
23:19.09hescomodprobe fails to load either of those modules
23:19.25hescoI use an IAX provider to reach the outside world
23:19.39hescodoesn't IAX require a zaptel timer?
23:21.21[jmc]I don't know ^^
23:21.33[jmc]stupid question: are the modules installed?
23:22.16[intra]lanmananother stupid question: does dmesg or syslog not have any errors?
23:22.17hescomake config says: DAHDI has been configured.
23:23.01hesconothing in dmesg
23:23.23jjshoehesco "fails" ?
23:23.59[TK]D-Fenderhesco: IAX Trunk Mode does
23:24.49hesconothing in syslog either
23:25.38hescoI just used an IAX trunk line to reach a friends voicemail on the pstn
23:29.33hescobut when I try to use the MeetMe application, I get in the CLI, the following error: "app_meetme.c:861 build_conf: Unable to open pseudo device"
23:32.16hescomake menuconfig indicates that I configured things enabling app_meetme, its marked by an asterisk
23:32.42hescoI wouldn't think the option would be available without Dahdi in place
23:34.46hescolsmod | grep -i dahdi says that dahdi is loaded, should I also see a dadhi-dummy ???
23:39.44*** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com)
23:39.51[TK]D-Fenderhesco: Might imagine this happening if * doesn't have the RIGHTS to initialize the device
23:42.36hescoif asterisk is started by root, and I've taken no special steps to setuid the server, wouldn't it have root privileges?
23:44.37[TK]D-Fenderhesco: what versions?
23:45.45*** join/#asterisk joako (n=joako@opensuse/member/joak0)
23:47.19hesco1.6.0.3
23:48.11hescodahdi-tools-2.0.0
23:49.24[TK]D-Fenderhesco: stop *.  run "dahdi_cfg -vvvv".  If that works OK, start * and retest
23:56.22*** join/#asterisk [jmc] (n=[jmc]@93-45-201-28.ip104.fastwebnet.it)
23:56.39hescowill do, thanks
23:56.48[jmc]wow
23:57.00[jmc]Twinkle locked up my computer
23:57.01[jmc]lol
23:58.34*** join/#asterisk Aiatek (n=Alfio@190.94.58.219)

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