IRC log for #asterisk on 20120326

00:23.08*** join/#asterisk Carlos_PHX1_ (~Carlos@ip68-2-227-192.ph.ph.cox.net)
00:24.06*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
01:09.40*** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
01:09.40*** mode/#asterisk [+o mjordan] by ChanServ
01:10.00*** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
01:53.05*** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com)
02:12.24*** join/#asterisk tengulre (~tengulre@182.148.109.115)
02:12.57*** join/#asterisk KNERD (~KNERD@adsl-99-65-0-241.dsl.hrlntx.sbcglobal.net)
02:23.09KNERDReading up on dial plans...if I get this correct..... exten => _X.,1,GotoIf($["${EXTEN}" = "6245"]? 6) once completed will continue on to _X,6, once executed?
02:23.24KNERDor should I say IF excuted
02:23.55[TK]D-Fendergive or take that spacer after the ? that you shouldn't have there
02:24.56KNERDwell I think this is from 1.4 *
02:25.15KNERDbut thanks
02:25.54[TK]D-Fendernever put extra spaces where they don't belong
02:26.28KNERDthanks for the advice. was looking at this example for implementing a dial plan
02:28.30p3nguin1.4 didn't use extraneous spaces, either.
02:28.50p3nguinAnd you should consider using proper labels instead of numbers for the target.
02:29.27KNERDThe next line basically goes into a script, then a Hangup. _X,6, is the same script again then another Hangup...should "6245" be able to be dialed at anytime then? Or have to wait until script finishes
02:29.45KNERDp3nguin: what proper labels would you reccomend?
02:31.23p3nguinIf you aren't going to use a proper extension for 6245...
02:31.25p3nguinexten => _X.,1,GotoIf($["${EXTEN}" = "6245"]?any-label-you-like)
02:31.37p3nguinBut extension 6245 would be my preference.
02:32.06p3nguinexten => 6245,1,Stuff()
02:32.43p3nguinI don't see the point of using a "catch all" extension just to later evaluate what extension was called.
02:32.54KNERDWell maybe you should look at the example I am looking at.. http://pastebin.ca/2132139
02:34.23p3nguinI guess it does reduce the number of lines of dial plan.
02:35.05*** join/#asterisk serafie (~erin@75.76.38.159)
02:35.22KNERDthe one I showed you, or the example you gave?
02:37.00p3nguinOther wise you'd end up with this:  http://pastebin.com/jrxU7kc7
02:37.26p3nguinThat's correct usage of extensions, but it adds more lines of dial plan.
02:38.06p3nguinI hadn't given the example at the time you asked about the example I gave.
02:39.03KNERDI would think the example you gave would require one of the 3 being dialed, and if they dialed none?
02:39.25p3nguinI just didn't include that part in my example.
02:39.36p3nguinI was only illustrating the difference.
02:39.42KNERDyes, I see
02:39.51KNERDjusty include another without any extension
02:40.30p3nguinI just found it odd to use a catch-all pattern and then evaluate what was actually called.
02:41.07p3nguinI'm not saying it's wrong or won't work, just saying I found it unusual.
02:41.26KNERDI see
02:41.46p3nguinIt isn't something I would ever do.
02:41.51KNERDwhat about a user entering one of those extensions while the script is executing?
02:42.00KNERDWhy would you not?
02:42.11p3nguinThe extension would run as many times as someone calls it.
02:42.30p3nguinUnless something else prevents it from running, of course.
02:42.32*** join/#asterisk kdmessano (~nonya@unaffiliated/kdmessano)
02:42.59KNERDokay..thanks for that advice
02:43.27KNERDnow I have to figure out how to implement the IVR to tell the person what to dial
02:43.43KNERDbecause the script is already telling them what to dial...ugg
02:43.49KNERDthanks a lot  p3nguin
02:45.22p3nguinOkay, for IVR, I can kind of see that method making a little more sense.
02:45.51p3nguinIVR would ask for input, caller would press some keys.
02:46.05p3nguinThen GotoIf() could go somewhere based on the entry.
02:46.17p3nguinBut I didn't see a Read() so I didn't recognize it as IVR.
02:46.51KNERDbecause it is really not...the script is running it's own IVR
02:47.15p3nguinits
02:47.16KNERDand reccomendation is to add your own similar to what I showed you
02:48.45KNERDi guess what I want to add is an announcement which states "for customer service press xxx" then goes into a queue
02:50.18p3nguinThat's not IVR.
02:50.25p3nguinThat's an auto attendant.
02:50.33p3nguinAnd you should use normal extensions for that.
02:51.44p3nguinFor customer service, press 1 now.  exten => 1,1,Goto(customer-service-queue,s,1)
02:54.46KNERDohhh
03:45.41*** join/#asterisk KNERD (~KNERD@adsl-99-65-0-241.dsl.hrlntx.sbcglobal.net)
04:06.20*** join/#asterisk KNERD (~KNERD@adsl-99-65-0-241.dsl.hrlntx.sbcglobal.net)
04:13.55*** join/#asterisk kessius (~cassio@201.21.173.58)
04:44.52*** join/#asterisk nix8n82-phone (~AndChat@65.161.180.230)
04:44.54*** join/#asterisk Sorcier_FXK (~nsystem@unaffiliated/sorcierfxk)
04:49.15*** join/#asterisk gajini (~root@61.12.17.171)
05:09.07*** join/#asterisk Sorcier_FXK (~nsystem@unaffiliated/sorcierfxk)
05:12.11*** join/#asterisk mintos (mvaliyav@nat/redhat/x-hzklgtzjcphtvvpa)
05:20.12*** join/#asterisk stasdizzi (~stas@198-153-133-95.pool.ukrtel.net)
05:20.32*** join/#asterisk ayrjola (~ayrjola@89.18.236.11)
05:24.21*** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net)
05:30.39dijibthis place is quiet
05:33.20*** join/#asterisk Sorcier_FXK (~nsystem@unaffiliated/sorcierfxk)
05:37.42*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)
05:50.22*** join/#asterisk KNERD (~KNERD@adsl-99-65-0-241.dsl.hrlntx.sbcglobal.net)
05:56.04*** join/#asterisk lanning (~Lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
05:56.40*** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com)
05:59.57*** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4)
06:00.12*** join/#asterisk Tim_Toady (~fuzzy@130.43.51.201.dsl.dyn.forthnet.gr)
06:08.01*** join/#asterisk beginer (~beginer@unaffiliated/beginer)
06:08.06*** join/#asterisk lanning (~Lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
06:11.00*** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105)
06:34.11*** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at)
06:34.15schmidtsgood morning
06:44.57*** join/#asterisk jsjc (~Adium@199.Red-79-150-67.dynamicIP.rima-tde.net)
06:47.07*** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no)
06:49.14*** join/#asterisk phix (~threat@123-243-44-131.static.tpgi.com.au)
06:52.59*** join/#asterisk BarthezZ (~bart@2001:41d0:2:9d0c::2)
07:07.23KNERDgodd night
07:12.04*** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net)
07:14.30*** join/#asterisk topriddy (~Seamfix@41.155.98.230)
07:15.56*** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net)
07:22.47*** join/#asterisk mpoole (~mpoole@minotaur.apache.org)
07:23.55*** join/#asterisk freeman_u (~freeman@gw.quart-soft.net)
07:25.05*** join/#asterisk Nasga (~Nasga@82.113.117.78.rev.sfr.net)
07:26.26*** join/#asterisk d00gster (~dt@77.31.13.27)
07:30.17*** join/#asterisk chasing`Sol (~cS@197.132.146.165)
07:33.41*** join/#asterisk Faustov (user@gentoo/user/faustov)
07:36.09*** join/#asterisk qakhan (~qakhan@203.130.22.202)
07:36.34qakhandoes anyone know how to install UniMRCP dependencies?
07:42.05*** join/#asterisk d00gster (~dt@77.31.13.27)
07:52.35jercosqakhan: yes.
07:52.51qakhanplease help me how?
07:56.31*** join/#asterisk frawd (~francois@221.red-80-28-139.adsl.static.ccgg.telefonica.net)
08:01.25*** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
08:37.51*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
08:44.04*** join/#asterisk TriJetScud (~TriJetScu@testarossa.smb.curriegrad2004.ca)
08:51.43qakhanjercos u there?
08:55.56*** join/#asterisk bulkorok (~bulkorok@85.183.36.36)
08:56.43jercosqakhan: think about your question, then about what we all know about you.
08:56.56*** join/#asterisk disposable (disposable@shell.websupport.sk)
08:57.22bulkorokis there a good dialer for asterisk at the moment?!
09:00.25*** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr)
09:03.41qakhanjercos i didnt get you
09:11.55*** join/#asterisk lanning_ (~Lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
09:16.15KNERDnothing?
09:16.50KNERDbulkorok: yeah..* has a great one...it's called DIAL
09:19.52*** join/#asterisk lanning_ (~Lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
09:38.45*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
09:40.59*** join/#asterisk ruben23 (~RLACUMBA@121.97.111.142)
09:41.07ruben23hi guys
09:44.23ruben23guys any help my phone extensions cannot register on my asterisk even the credentials are correct
09:44.29*** join/#asterisk sekil (~sekil@78.24.104.73)
09:44.38ruben23NOTICE[2401]: channel.c:3271 __ast_request_and_dial: Unable to request channel SIP/cc121
09:44.43*** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md)
09:46.11*** join/#asterisk Sconk (~krh@2a01:7e8:a0:1:c8fe:ff:fe00:babe)
09:46.16schmidtsruben23 you mixed two things, what you have shown us is when you try to dial a peer, but not what you get when a peer tries to register
09:46.49*** part/#asterisk topriddy (~Seamfix@41.155.98.230)
09:48.35Sconkis there a way to see members of my sales que ?
09:49.19Sconkqueue show
09:49.20Sconk:)
09:50.54KNERDfreepbx
09:52.00ruben23schmidts: sorry let me cehck the lgos again
09:52.04ruben23log*
09:53.01ruben23i see this on the log ---> [Mar 26 11:52:07]     -- Got SIP response 405 "Method Not Allowed" back from 81.133.43.31 ---> [Mar 26 11:52:07]     -- Got SIP response 405 "Method Not Allowed" back from 81.133.43.31
09:54.19KNERDsounnds like a codec issue?
09:56.32KNERDhttp://en.wikipedia.org/wiki/List_of_SIP_response_codes#4xx.E2.80.94Client_Failure_Responses
09:59.02ruben23KNERD: this regards with codec..? ------------>405 Method Not Allowed The method specified in the Request-Line is understood, but not allowed for the address identified by the Request-URI.
09:59.04KNERDruben23: oh....type=peer or type=friend? try swapping
10:00.11KNERDof course would help ifwe saw more
10:00.15kaldemarruben23: look at SIP debug. you'll see what the 405 is an answer to. however, you phones not being able to register is a separate issue.
10:04.46*** join/#asterisk mpe (~mpe@office.ipvision.dk)
10:05.15*** join/#asterisk mpe (~mpe@office.ipvision.dk)
10:05.30ruben23guys would this help----------------------------------> http://pastebin.com/vjN72gxe
10:07.35*** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com)
10:08.19ruben23its strange coz when i used zoiper it work with other it work with 3cxphone it does not work at all but for teh record teh 3CX phones work already for 2 weeks then suddenly today all not working
10:08.31*** join/#asterisk timahvo1 (~rogue@41.81.172.193)
10:09.12davlefouhi,
10:09.34KNERDruben23: actually it would help if we saw the client and seever settings for this
10:11.30kaldemarruben23: your asterisk is sending voicemail notificacations to 81.133.43.31, which it probably should not. don't define a mailbox for it in sip.conf if it is a provider.
10:12.21KNERDhasvoicemail=no ?
10:14.01davlefouis it possible to use same numbre for fax and phone with asterisk? I think with sub number?
10:14.57KNERDof course it is. you just tell it to listen for fax
10:29.09*** join/#asterisk wonderworld (~ww@dsdf-4db53af5.pool.mediaWays.net)
10:31.35qakhananyone using uniMRCP?
10:48.20*** join/#asterisk RiceCracker (~RiceCrack@59.152.236.158)
10:48.31KNERDqakhan: .000000000000000000001 seconds via Bing http://www.voip-info.org/wiki/view/Asterisk+cmd+MRCPSpeech
11:04.06*** join/#asterisk freeman_u (~freeman@gw.quart-soft.net)
11:07.17qakhanKNERD anything else except this link
11:08.05KNERDqakhan: yeah. www.bing.com
11:08.29*** join/#asterisk lanning (~Lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
11:09.40*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)
11:10.39*** join/#asterisk lanning_ (~Lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
11:35.08*** join/#asterisk fred9999 (~chatzilla@AMarseille-151-1-67-149.w83-205.abo.wanadoo.fr)
11:35.26fred9999good morning everyone
11:38.35fred9999I am mixing up extension and context in conf file. I would like to have an extension that kicks the user directly to a meetme room with fixed number. I have 2 sterisk linked by a sip trunk. I can call remote extension but when I try to define an extension that would kick into a meetme room. I have a Call from '' to extension '8000' rejected because extension not found.
11:38.46fred9999Here are the conf file
11:40.23fred9999on the target * (number 2)  extension.conf
11:40.41fred9999[internal]
11:40.43fred9999exten=>9600,1,Dial(SIP/9600,,r)
11:40.44fred9999exten=>9601,1,Dial(SIP/9601,,r)
11:40.46fred9999;exten=>8000,1,Dial(SIP/9600,,r)
11:40.46kaldemar~pb
11:40.46infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
11:40.47fred9999exten=>8000,n,MeetMe(100,1dqF)
11:41.22fred9999I'll use the pastebin for the config file
11:42.04kaldemaryou need to paste sip.conf aswell and a CLI output of a call with verbosity and sip debug for someone to be able to specifically tell you what's wrong.
11:53.48*** join/#asterisk Bullmoose (~Bullmoose@71-33-30-40.bois.qwest.net)
11:55.18fred9999seems ok now. I have defined an extension 8000 in the sip.conf file. I guess when calling SIP/XXXX, the XXXX has to be defined in the sip.conf file, right ?
11:59.58*** join/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497)
12:00.07*** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162)
12:00.14kaldemaryes. but what you define in sip.conf is not an extension but a device. an extension is something that begins with "exten =>" in extensions.conf.
12:09.40*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
12:12.36*** join/#asterisk slidesinger-lt (~jtatum@173-161-172-126-Philadelphia.hfc.comcastbusiness.net)
12:25.10*** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
12:26.33*** join/#asterisk fisted_ (~fisted@unaffiliated/fisted)
12:26.53*** join/#asterisk akrohn (~akrohn@38.101.60.42)
12:28.15*** join/#asterisk domi (domi@80.86.84.141)
12:31.32*** join/#asterisk gonewage (~gonewage@72.2.130.205)
12:32.27*** join/#asterisk mjordan (~mjordan@nat/digium/x-kvgmzrsthzwsaduc)
12:32.27*** mode/#asterisk [+o mjordan] by ChanServ
12:33.59*** join/#asterisk adeel (~adeel@72.53.78.136)
12:51.53*** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co)
13:00.46*** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek)
13:05.37fred9999thanks, these are the things I am mixing up.
13:06.46*** join/#asterisk mr_pete (5ce892a4@gateway/web/freenode/ip.92.232.146.164)
13:06.58mr_peteAfternoon all :)
13:08.24mr_peteCan anyone help me with keeping asterisk in channel to keep features working?  Moved from 1.4 to 1.6 and for some reason (with dtmfmode set to rfc2833 on all extensions & trunks) and canreinvite=no - features WERE working outbound on 1.4
13:08.27mr_peteand they aren't no 1.6
13:09.17*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
13:11.37*** join/#asterisk anonymouz666 (~anonymouz@189.25.74.253)
13:13.45*** part/#asterisk gonewage (~gonewage@72.2.130.205)
13:17.22*** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl)
13:17.24jacc0hi all
13:17.26jacc0:)
13:18.02*** join/#asterisk Delido192123 (91fdec13@gateway/web/freenode/ip.145.253.236.19)
13:19.52Delido192123Hello i need Help with Asterisk´s CallCompletionRequest, CCBS is not working at all, but ccnr. i think this is happend because i dont use the DIAL Function on Busy state. Can someone help me to correct this?
13:24.21Delido192123i cant set call-limit to 1 because noone can do callforwarding. asterisk ignore the busy-level and the dial-command are executet. I have Polycom IP320 SIP PHones at all. The User can choise within our intranet the callwaiting state (on / off)
13:27.17fred9999thank you kaldemar for your help. have a nice day !
13:28.23*** join/#asterisk tamiel (~tamiel@85-171-171-179.rev.numericable.fr)
13:28.34kaldemarfred9999: thanks, you too.
13:28.57[TK]D-Fendermr_pete, canreinvite became "directmedia".  and dmdmode is unchanged
13:29.57mr_pete[TK]D-Fender: incoming calls (any trunk) features can be activated fine - and local ext2ext calls work, but outgoing, no dice :\
13:31.06[TK]D-Fenderwhat "features"?  Your terminology is very vague
13:31.20[TK]D-Fenderand you aren't showing us the actual failure for us to advise you on.
13:31.29[TK]D-FenderPASTEBIN it all up so we can see what's happening.
13:31.30[TK]D-Fender~pb
13:31.30infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
13:31.31[TK]D-Fender^^
13:37.04mr_peteanything active in features.conf
13:37.39mr_petelike blind transfer.  It's why I was referring to canreinvite as I suspect Asterisk is going out of the loop and thus not "listening" anymore
13:40.48mr_petetweaking debug...
13:43.04mr_peteI'm seeing DTMF locally (<< [ TYPE: DTMF End (1) SUBCLASS: * (42) ] [SIP/6000-00000050] )
13:43.12mr_petebut not when calls go via trunk
13:46.17*** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan)
13:48.10*** join/#asterisk serafie (~erin@nat/digium/x-smzmnnysisexpcov)
13:52.37[TK]D-Fendermr_pete, if you set the dial options to allow for them in the first place no reinvite is permitted regardless
13:52.48*** join/#asterisk digiv (~textual@as1.si.umich.edu)
13:53.17*** join/#asterisk polysics (~polysics@host7-14-static.4-79-b.business.telecomitalia.it)
13:53.27polysicshello, Asterisk people!
13:53.38mr_peteI've got "DIALOPTIONS = KkXxtT" set in extensions.conf, and the features are enabled in features.conf
13:53.59polysicsi know about attended transfer using either Dial() or features.conf
13:54.10polysicsbut how would I pilot that using AMI?
13:54.12mr_peteand as far as I can tell macro-trunkdial-failover-0.3 should call thos in?
13:54.32polysicsgoal is to integrate with a small web-based panel that shows people's satatus
13:54.54polysicsthere is no single app that does that, right?
13:57.42mr_petehmm, think I found the culprit...
13:59.06qakhandoes anyone know how to install sofia-sip?
14:00.03mr_petefixed it
14:00.18mr_peteit's a GUI issue, relying on macro-trunkdial
14:00.31mr_petethe macro doesn't call in "exten = s,n,Set(__DYNAMIC_FEATURES=${FEATURES}) "
14:00.45mr_peteand doesn't append DIAL_OPTIONS
14:01.52[TK]D-Fender<mr_pete> I've got "DIALOPTIONS = KkXxtT" set in extensions.conf, and the features are enabled in features.conf <- this is not a magical global variable to my awareness... I'd like to see the actual CALL.
14:02.10[TK]D-Fenderqakhan, You don't.  Asterisk does not use that stack
14:02.38mr_pete[TK]D-Fender: I'm using asterisk-GUI - it's put DIALOPTIONS in the conf file and updates it
14:03.05mr_peteand it's calling them via it's built macros (ala "exten = s,n,Dial(SIP/6000&SIP/6001,20,${DIALOPTIONS}i)"
14:03.32mr_petebut it doesn't append that to the trunk-dial macro which all outgoing calls go to, so they don't get activated
14:03.37polysicsi was thinking of manually parking a call, dialing out, then bridging the second party to the first
14:03.43polysicsthere has to be an easier way :-)
14:03.58[TK]D-Fendermr_pete, FYI that GUI is practically unmaintained and has extremely minimal support at best.  Consider this if you are planning on using it for anything other than some personal laerning experience
14:04.29[TK]D-Fender<polysics> i know about attended transfer using either Dial() or features.conf <- same thing
14:04.49mr_pete[TK]D-Fender: am gathering that :)  It's for a home system. I was on 1.4 CLI before
14:04.55[TK]D-Fenderpolysics,  but how would I pilot that using AMI? <- no such thing, only a hard dialplan forward, or a "Bridge"
14:05.06mr_petejust migrating to a Debian-Squeeze install on a Proliant microserver which has 1.6
14:05.17[TK]D-Fendermr_pete, Still generally advise against, so just keep this in mind
14:05.59polysics[TK]D-Fender: so i would send the caller to another extension, dial out to  the potential receiver, and eventually connect them through a series of bridges and redirects?
14:06.07*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
14:06.19polysicsit's either all in features.conf or totally manual
14:06.35[TK]D-Fendermr_pete, do yourself a favour and join #asterisk-gui ... and help double their population ;)
14:06.49[TK]D-Fenderpolysics, No, those were 2 separate solutions
14:06.50mr_petewell for the most part it's worked :P
14:07.12mr_peteonly problems I've had are becuase I'm using an SPA3102 for POTS interface (and digest auth had to be set manually via sip.conf)
14:07.19mr_peteand now this :)
14:07.29polysicswhat is a "hard dialplan forward", please?
14:07.47[TK]D-Fenderpolysics, And the bridge way you'd Originate a call to a receiver, set a channel variable to the channel to hijack, and on accept, call Bridge() from the dialplan
14:08.23polysics[TK]D-Fender: in that scenario, where is the caller waiting in the meantime?
14:08.32*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
14:09.03[TK]D-Fenderpolysics, Wherever they are.... I suppose you could couple that action with forwarding them somewhere else like a loop of MusicOnHold or something like that...
14:09.33mr_peteright, now all features.conf features and my applications are now in channel. hurrah :)
14:09.41polysicsi know this is probably easy, but what would the best way of providing an "endless" waiting "place" for a caller?
14:11.01[TK]D-Fenderpolysics, hard redirect to an exten dumping them in a loop of MusicOnHold
14:11.15*** join/#asterisk nickfennell (~nick@unaffiliated/nickfennell)
14:12.05[TK]D-Fenderpolysics, I might also swap that for having a channel variable set for ID purposes, and then use Park.  The var might help ID it for tracking it for pickup via ParkedCall
14:12.08*** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com)
14:13.11*** join/#asterisk Azrael808 (~peter@212.161.9.162)
14:13.12fred9999good evening,
14:13.13polysicswell, it is actually parked in a sense, i think
14:15.07fred9999I am trying to pass a meetme parameter from one asterisk calling another asterisk on a device 8000 that kicks the call directly to the room number. How do I do that. The dialplan is calling the dial function.
14:16.18[TK]D-Fenderfred9999, what is "device 8000"?  what are you trying to pass?
14:17.28Delido192123http://pastebin.com/hqpk1CQ3  Callcompletion dont work (ccbs) please help
14:17.50fred9999it is a fake SIP device.  I am trying to pass the room number the call should be redirected to on the second *
14:18.08[TK]D-FenderHow is it "fake"... when it leads to ANOTHER Asterisk?
14:18.25mr_peteOk, next silly question :)
14:18.27fred9999there is no physical phone attached
14:18.32mr_petewhat did "show globals" migrate to under 1.6?
14:18.37nickfennellonly theoretical ones
14:18.41[TK]D-FenderWho said SIP had anything to do with a phone?
14:18.50fred9999my mistake
14:18.53nickfennellI've just used SIP to get a cuppa made
14:19.05[TK]D-Fenderhi-5's nickfennell
14:19.10nickfennelllol
14:19.18[TK]D-Fendernickfennell, I use * for coffee ... and as a jukebox....
14:19.48nickfennellJukebox is good although I find myself dialling in now to listen to streams rather than just using a native client...
14:19.55[TK]D-Fenderfred9999, have you considered making the romm # part of the extension you dial on the other server?
14:20.33[TK]D-Fendernickfennell, I just dial into my player script, pick my collection, and run it on speakerphone :)
14:21.02nickfennellSo you're hosting media files on the * itself?
14:21.05fred9999that could do. How would I define all those devices in the sip.conf ?
14:21.23[TK]D-Fenderfred9999, You don't.. I said EXTENSION.  that = extensions.conf
14:21.56nickfennellI need to get SMS Call back working
14:22.00mr_petewhat did "show globals" migrate to under 1.6?
14:22.09[TK]D-Fendernickfennell, If by "hosting" you mean on the same box as my server sitting in my livingroom... then yes :)
14:22.16nickfennelllol
14:22.20nickfennellYou have * at home too ?
14:22.22[TK]D-Fendermr_pete, probably "core ..."
14:22.24nickfennellhahaha.
14:22.31[TK]D-Fendernickfennell, Have since 2004.
14:22.34mr_petenot that I can see :P
14:22.42nickfennellI'm the same. I must be the only person I know to have hold music and IVR at a house
14:22.55mr_petenope, I had hold music until yesterday :)
14:23.06nickfennell"Thanks for calling. Press one for the kitchen, Two for the Garden, Three for the Shed"
14:23.06*** join/#asterisk StaRetji (~LittleAll@178.79.11.103)
14:23.09[TK]D-Fendermr_pete, You should be reading the "ugrade" docs, etc in your tarball.  They detail all of these sort of changes.
14:23.15mr_peteI had IVR for anon callers
14:23.31StaRetjifolks, does asterisk default install translate between gsm and ulaw?
14:23.37nickfennellI had it so *genuine*callers would be able to redirect to my mobile if needed
14:23.47nickfennellCold callers didn't seem to listen to the options
14:23.50[TK]D-FenderStaRetji, * translates whatever it can translate whenever it has to
14:24.03[TK]D-FenderStaRetji, which by stock distribution does include those 2 codecs
14:24.05StaRetjiI have strange situation, calls made to outside numbers (landline, mobile)
14:24.15StaRetjiare in ulaw
14:24.18nickfennellAt one point I had it running a dictation service so I could dial in and set memo's for myself
14:24.22StaRetjibut if I make callback
14:24.25nickfennellcan't remember how I wired that up though
14:24.29StaRetjiboth legs are in gsm
14:24.43StaRetjiand no sound is heared by both parties
14:24.55StaRetjishow channels says gsm
14:25.02nickfennellstix, inbound calls are GSM?
14:25.04[TK]D-FenderStaRetji, If you had a codec issue the call would drop like a rock instantly
14:25.05StaRetjiwhile I don't gsm set in sip.conf
14:25.10[TK]D-FenderStaRetji, You have a networking issue
14:25.37StaRetji[TK]D-Fender: thx for answerz
14:25.58StaRetjiI have to understand why are callback calls made in gsm
14:26.00stixnickfennell: if you say so??
14:26.10StaRetjiulaw alaw, sound is there
14:26.17StaRetjigsm, no sound
14:26.21nickfennellstix, I must have done ?
14:26.33nickfennellstix, I mean StaRetji
14:26.37nickfennellApologies.
14:26.39StaRetjioh
14:26.41StaRetjisorry mate
14:26.50StaRetjiI was confused, looks like it is for me
14:26.55StaRetjibut different name
14:27.07[TK]D-Fender<StaRetji> I have to understand why are callback calls made in gsm <- yes... you should be actually looking at that call right now.....
14:27.10nickfennellIgnore me. I'm silently failing.
14:27.23*** join/#asterisk GameGamer43 (u5533@gateway/web/irccloud.com/x-uluuzlgyatpdmgio)
14:27.30StaRetjiI mean, I call my self with same trunk
14:27.34mr_petezapateller worked nicely on most auto-diallers for me
14:27.37StaRetjiand call is ulaw
14:27.40StaRetjiif I make callback
14:27.46StaRetjito my phone and another phone
14:27.53StaRetjicallback is gsm
14:28.16nickfennellcodec preferred on handset?
14:28.20StaRetjiso, even though in sip.conf is never put allow=gsm
14:28.33StaRetjibut same handset ulaw
14:28.40StaRetjiif I call not with my sip account
14:28.46StaRetjicall now*
14:28.59StaRetjibut if I initate callback, it is gsm
14:29.16StaRetjiI bang my poor stupid head with this lol
14:29.27[TK]D-FenderAnd I'm no sensing that you've actually started really looking at that call....
14:29.30StaRetjiI mean, no gsm codec set, anywhere
14:29.43mr_petemy answer = dialplan show globals
14:29.50StaRetjilooking, like keeping my eyes wide open? :D
14:29.50nickfennellI don't follow the 'initiate callback'
14:30.00*** join/#asterisk Joel_Oliveira (~chatzilla@estrela-adm.nortenet.pt)
14:30.09nickfennellIs that calling from landline/mobile to the Asterisk?
14:30.20StaRetjiput number for leg1, put number for leg2, press callback
14:30.31StaRetjino, it is asterisk, calling 2 phones
14:30.37nickfennelloh right
14:30.37StaRetjiconnects them together
14:30.39nickfennellGot it
14:30.42nickfennellNot seen that
14:30.48nickfennelllives in a dark hole
14:31.28*** join/#asterisk bmg505 (~leon@196-209-152-75.dynamic.isadsl.co.za)
14:31.30StaRetjilet me test some more...
14:32.27StaRetjiyep, called via sip account to phone number1, ulaw
14:32.36StaRetjicalled via sip accont phone number2, ulaw
14:32.52StaRetjinone of the sip setting has gsm in it
14:33.05fred9999Let me explain the way  I do, which might not be the good way. *2 is calling SIP/9600@38-peer on the sip trunk between the 2 *. on the 2nd * I have a SIP/8000 device which does that exten=>8000,1,MeetMe(100,1dqF).The room number is 100. I would like * to pass this info to *2  in some parameter. How do I do ?
14:33.17StaRetjibut if call leg1.number1, leg2.number2 via callback, they are connected in gsm
14:33.19StaRetjiwtf
14:34.29[TK]D-Fenderfred9999,  SIP/9600@38-pee <------ make the room part of the number you dial
14:37.30*** join/#asterisk serafie (~erin@nat/digium/x-shymnlbyjssrntrx)
14:38.00*** join/#asterisk bn-7bc (bjarne@macbook-pro.lan-sx.noare-1.holmedal.net)
14:38.54fred9999something like this SIP/9600/8001@38-peer ?
14:41.40qakhani am getting this message while i am installing unimrcp
14:42.38qakhanthe message is /usr/bin/ld: cannot find -lexpat
14:44.47*** part/#asterisk vipkilla (~chatzilla@unaffiliated/t-dot-zilla/x-2830497)
14:48.12*** join/#asterisk [sr] (~kvirc@pal-213-228-181-48.netvisao.pt)
14:48.40[TK]D-Fenderfred9999, what is the number you were dialing in there in the first place?
14:51.00*** join/#asterisk chasing`Sol (~cS@197.134.183.118)
14:51.57*** join/#asterisk OrNix (~OrNix@l49-246-139.cn.ru)
14:53.46*** join/#asterisk ddickenson (~ddickenso@67-198-0-5.static.grandenetworks.net)
14:54.21ddickensonAnyone else had trouble with asterisk 10.x not starting as a service like the old ones on Centos?
14:57.45fred9999The first number I put is wrong. I am dialing Dial(SIP/8000@38-peer)
14:58.04fred9999I am on 1.6
14:58.31[TK]D-Fenderfred9999, So you are dialing 8000.  make the room PART of what you are dialing.
14:59.25*** join/#asterisk pdtpatr1ck (~pdtpatric@ip184-179-74-42.oc.oc.cox.net)
14:59.42*** join/#asterisk stix (~stix@193.89.191.209)
15:01.26*** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith)
15:07.23leifmadsenQwell: ping
15:10.57*** join/#asterisk jrad (~jrad@173-11-144-149-houston.txt.hfc.comcastbusiness.net)
15:13.10jradAsterisk v10.2.1 - Trying to locate a memory leak that brings down the service once a day during a production work day. Are there any built in commands or build options I can use effectivly?
15:15.58leifmadsenyou could enable some debug options from menuselect like DONT_OPTIMIZE and DEBUG_THREADS and then create a backtrace from the running process and 'core show locks' to gather some information
15:16.34leifmadsenjrad: https://wiki.asterisk.org/wiki/display/AST/Debugging
15:16.49jradThank you for your time. I will review.
15:17.00*** join/#asterisk serafie (~erin@nat/digium/x-koyxqozawmwpsprd)
15:23.09mr_peteo_O
15:23.30mr_petemy POTS provider has just suffered a regional outage....(whilst I'm testing PSTN failover)
15:23.38mr_peteI did not at all think it was anything to do with me.... :\
15:27.51mpoolehey guys, I'm having a random problem on asterisk 1.8. When I dial into a sip trunk 1/10 times the call is answered but silent. The only thing in the log is: chan_sip.c: Unsupported SDP media type in offer: video 0 RTP/AVP 31 34 98 99
15:27.55mpooleany ideas?
15:28.24*** join/#asterisk gruvfunk (18a121cf@gateway/web/freenode/ip.24.161.33.207)
15:28.30gruvfunkgreets all
15:29.20gruvfunkanyone know how I can pull the 10 digit caller ID from SIP headers using Python?
15:30.00gruvfunkcurrently have something but it will pull a Caller ID Name sometimes depending on where the call is coming from (internal PBX calls works, external calls from PSTN do not).
15:46.46gruvfunkawake and alive?
15:47.09bmoraca_workit's too bad the G200 and the G100 aren't a little bit cheaper...if they were, they'd be a good alternative to the Adtran TA900s.  TA900s are cheaper, though.
15:49.39*** join/#asterisk pdtpatr1ck (~pdtpatric@12.249.4.226)
15:50.02*** join/#asterisk Azrael808 (~peter@212.161.9.162)
16:07.28*** join/#asterisk adeel (~adeel@72.53.78.136)
16:07.32jayteeanyone here using the Buddy Watch feature on Polycom phones with Asterisk?
16:08.09bmoraca_workit works OK as a standard BLF against hints in Asterisk
16:08.41jayteebmoraca_work, does it use up a line key exclusively?
16:09.00bmoraca_workit also functions as a speed dial for that user/featurecode
16:09.15bmoraca_workbut, yes, it does dedicate the key
16:09.16*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
16:09.26jayteebmoraca_work, thanks
16:09.55bmoraca_workhas anyone here connected an Avaya IP Office to Asterisk before?  is it possible without accepting anonymous calls?
16:10.22*** part/#asterisk mikemol (~shortcirc@inara.firefly.michael.mol.name)
16:16.45*** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net)
16:19.19[TK]D-Fenderjaytee, more specifically your user Directory will spill onto unused line-keys in "speed-dial index" order which is the best way to see them.  You can also view the status via Buddies on the phone itself
16:21.00[TK]D-Fenderbmoraca_work, What hasn't worked in terms of setting up a peer like normal?
16:21.59bmoraca_work[TK]D-Fender: i don't have one at my disposal, but I've been doing research for a company who maintains a customers' IP Office (who I put in a new Asterisk system for one of their divisions) and they want to do extension-to-extension dialing.
16:22.12bmoraca_work[TK]D-Fender: every walkthrough I find says that I need to allow anonymous calls through
16:22.17bmoraca_workwhich doesn't seem right
16:22.28bmoraca_workbut i don't know what the actual SIP dialog looks like
16:22.56bmoraca_worki can't imagine that a standard peer with insecure=port,invite wouldn't work...
16:23.04bmoraca_workbut you never know!
16:23.37[TK]D-Fenderbmoraca_work, Don't let your imagination wander... it's too little to be allowed out on its own ;)
16:23.45bmoraca_workpfft
16:24.18bmoraca_worki don't want to take anything for granted because i've dealt with this company before and they're not very bright
16:26.52*** join/#asterisk Defraz (~Defraz@70.36.76.167)
16:29.27*** join/#asterisk vinhdizzo (~vinh@dhcp-v027-166.mobile.uci.edu)
16:29.42*** join/#asterisk engrxyz (~fgdfgf@wempex01.inclarity.co.uk)
16:30.49*** join/#asterisk techwerkz (~Justin@c-76-101-15-40.hsd1.fl.comcast.net)
16:32.36techwerkzI have a list of 309 local prefixes I need to go out a specific dahdi span. Is there a good way to handle local prefixes within Asterisk without having to create the dialplan for each prefix? I would do some pattern matching, but none really are in a series and are pretty random. Calls outside these prefixes need to go out my other span.
16:33.04*** join/#asterisk SupYoshi (SupYoshi@ip51cc8577.speed.planet.nl)
16:33.13SupYoshiHi im having issues with ASterisk installation
16:33.15jradTo whom it concerns. I have started the memory leak debugging stated in the Asterisk wiki. We have to wait for the problem to reoccur, however this is one of the errors that has me concerned. http://pastebin.com/VwpRbsc8
16:33.18SupYoshiI get errors when i try make install
16:33.25SupYoshiThis is my pastebin of .configure
16:33.27SupYoshihttp://pastebin.com/wkkK2xs0
16:35.01jrad@SupYoshi - I am no pro. However, have you tried the command "make distclean" start fresh. Then I would refer you to Wiki of Asterisk.
16:35.41SupYoshii just eh...
16:35.43SupYoshiI followed a guide
16:35.58SupYoshiI installed asterisk before with different guides, im just having this error now ;)
16:39.32*** join/#asterisk NightMonkey (~NightMonk@pdpc/supporter/professional/nightmonkey)
16:40.47SupYoshiMHm
16:49.02SupYoshihttp://pastebin.com/PPQN8ezR
16:49.05SupYoshibetter? -.-
16:49.06SupYoshilol
16:53.06p3nguintechwerkz: No, you have to use dial plan to do it.
16:55.00[TK]D-Fendertechwerkz, you could put them into a file and une func_shell or an AGI to get to them, or put them in a DB somehow (AstDB is fairly easy) and just use a quick lookup.  Not a big deal not to have to make static dialplan for them.
16:55.06p3nguintechwerkz: It wouldn't be that hard to do.  You could have one line of dial plan to "catch" each number called, and then use a macro to reduce duplicate lines of dial plan for every other line that would be identical.
16:56.32*** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
17:05.55SupYoshiCan someone help me?
17:07.06pabelangerSupYoshi: what OS?
17:08.07*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
17:08.26pabelangerleifmadsen: qwell is on vacation I think
17:08.45ChannelZThat's odd, why does it say 'Killed'
17:13.54*** join/#asterisk _Corey_ (~chatzilla@64.121.4.75)
17:17.23*** join/#asterisk Joel_Oliveira (~chatzilla@estrela-adm.nortenet.pt)
17:18.41leifmadsenpabelanger: coolio thanks -- no worries I figured out my issue
17:19.36SupYoshiCentois 5
17:19.38SupYoshiCEntos 5
17:20.33*** join/#asterisk devmikey (~irc@unaffiliated/devmikey)
17:21.28leifmadsenSupYoshi: o.O
17:22.45SupYoshiCEntos5, I have the following error
17:22.50SupYoshiWhen I http://pastebin.com/wkkK2xs0
17:22.51techwerkz[TK]D-Fender and p3nguin thanks for the ideas. I will just use AstDB
17:22.53SupYoshiConfigure it
17:27.13SupYoshihelp?
17:30.14dijibhowdy yall
17:30.23SupYoshihi
17:32.10pabelangerSupYoshi: what is your processor?
17:32.14pabelangercat /proc/cpu
17:32.28pabelangercat /proc/cpuinfo
17:33.24pabelangerSupYoshi: $ make distclean
17:33.26pabelanger./configure
17:33.27pabelangermake
17:33.43dijibhe has ld? just dirty?
17:34.12SupYoshiIntel(R) Xeon(R) CPU           E5620  @ 2.40GHz
17:34.19SupYoshidid that
17:34.23SupYoshii keep doing these things
17:34.25SupYoshifor 3 hours
17:34.28SupYoshiand I keep gettign the same error
17:34.41SupYoshiake: *** [channels] Killed
17:34.46SupYoshimake: *** [channels] Killed
17:34.58pabelangerare you killing it?
17:34.58leifmadsensounds like a problem with menuselect
17:35.00SupYoshiand i did distclean 3 times
17:35.03SupYoshino im not killing it
17:35.07pabelangerthen what is
17:35.08SupYoshiyes it is a problem with menuselect
17:35.12SupYoshibut ive no idea what it is
17:35.13dijibbinutis missing?
17:35.19leifmadsencd menuselect
17:35.19dijibbinutils
17:35.20leifmadsenmake clean
17:35.41leifmadsenhonestly, just delete the entire asterisk source directory and re-extract or checkout from svn clean
17:35.42SupYoshidone that too
17:35.43SupYoshi3 times
17:35.47SupYoshioh i did that
17:35.57leifmadsenyou're doing something wrong then
17:35.59SupYoshieven a Yum install did better
17:36.02leifmadsentar zxvf asterisk-1.8.8.1.tar.gz
17:36.06leifmadsencd 1.8.8.1
17:36.10SupYoshii actually got the yum install working
17:36.13SupYoshiif that matters
17:36.14leifmadsen./configure && make && make install
17:36.17leifmadsenthat's all you should have to do
17:36.18SupYoshibut then i reinstalled
17:36.35leifmadsenwhy are you mixing compiling source and package management?
17:36.38SupYoshimkay i will just reinstall this VPS and start over
17:36.46SupYoshiBecause I did one reinstall already
17:36.50SupYoshi:) Doing Yum
17:36.57SupYoshihowever i messed up on mysql then (bad guide)
17:36.59leifmadsenwell sounds like you've messed something up
17:37.08leifmadsenjust use asteriskdocs.org
17:37.41SupYoshiwhats the easiest OS?
17:37.43SupYoshito install it on?
17:38.03leifmadsendoesn't matter
17:38.05SupYoshiok
17:38.07leifmadseninstructions show ubuntu and centos
17:38.14SupYoshicentos 5 minimal or centos 5 normal better?
17:38.14leifmadsenpick whatever one you prefer
17:38.23leifmadsenfollow the installation instructions
17:38.27leifmadsenit tells you
17:38.42SupYoshik
17:38.42pabelangerSupYoshi: try ./configure CFLAGS=-fPIC"
17:38.43pabelangermake
17:38.49leifmadsenhttp://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-Install.html
17:38.49pabelangererr
17:38.53pabelangerSupYoshi: try ./configure CFLAGS="-fPIC"
17:39.01SupYoshiyea to alte
17:39.02SupYoshilol
17:39.08SupYoshiim pissed off on asterisk today
17:39.09SupYoshilol
17:39.20leifmadsenok
17:39.25SupYoshijust reinstalling the box
17:39.28SupYoshistarting again
17:39.30leifmadsenit's not asterisk that's the problem
17:39.36SupYoshino its me
17:40.06dijibcould someone look at this PB and tell me why with the app_meetme.so module loaded the meetme app in dialplan is failing? http://pastebin.com/8XAKjQdY
17:40.09leifmadsendigiv_away: please don't change names when setting away
17:40.45jradConcerning the possible memory leak. Is there a way to force a core dump instead of waiting for it to run out of all RAM.
17:40.49*** part/#asterisk digiv_away (~textual@as1.si.umich.edu)
17:41.11leifmadsenjrad: you don't need it to coredump to get a backtrace
17:41.16leifmadsenyou can connect to the running process
17:41.44jradThanks.
17:41.51dijibbrb making coffee
17:46.40*** join/#asterisk serafie (~erin@nat/digium/x-gqfuctsplptrwksf)
17:52.15*** join/#asterisk timahvo1 (~rogue@197.178.29.45)
17:52.45gruvfunkany Python gurus in here? I need to strip the phone number from the SIP_FROM header ?
17:53.12gruvfunkcurrent script is pulling the caller's NAME, but we need the NUMBER
17:53.34gruvfunkFrom: "CITY, NY" <sip:13475551212@10.9.9.5>;tag=as5ca29464
17:53.54gruvfunkclid, uri = from_hdr.split(" <sip")
17:54.17gruvfunksplit goes to the left, I think rsplit goes to the right, but I need to stop at  "@"
17:54.21gruvfunkanyone help me?
17:54.49gruvfunktried #python but that was well over my head, amazed really since there are so many people in there who would quickly solve this
17:54.56leifmadsengruvfunk: I'd try #python or something as that is more of a python parsing question than an Asterisk question
17:54.59[TK]D-Fendergruvfunk, is that somehow not in the CALLERID?
17:55.46gruvfunkleifmadsen: yeah I did, concepts over my head and got RTFM
17:55.49gruvfunk[TK]D-Fender: ?
17:56.29[TK]D-Fenderthe FROM is typically parsed out into the callerid values for the channel.  Are they not as they should be>?
17:57.08*** join/#asterisk its_jeremy_ (~omghax@gateway/tor-sasl/itsjeremy/x-75806909)
17:57.30*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
17:57.41gruvfunk[TK]D-Fender: I guess this isn't so much an Asterisk setup question, this is a small local script that launches on calls received into Twinkle voip client - so it takes the sip header coming in
17:58.31[TK]D-Fendergruvfunk, Well if you're transfixed on a pre-made script which already breaks the header up then you're going to have to bite the bullet and learn how to program in python then.
17:59.06gruvfunk:-) right, was hoping somebody in here knows enough to point me in the right direction
17:59.29gruvfunkbut tha'ts what #python is for i guess
17:59.34gruvfunksigh
18:06.00*** join/#asterisk glaz (strke@hiro.glaciuz.com)
18:08.09glazI'm trying to get operator = yes to work in voicemail.conf but seems like asterisk is ignoring the DTMF inputs while playing the unavailable message. I've reloaded dialplan and voicemail. Any suggestion?
18:09.57glazLooks like someone has open a bug request Today, https://issues.asterisk.org/view.php?id=14731
18:11.09glazOops, that's just the current time/date. The ticket was open in 2009
18:13.29*** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk)
18:13.35glazBut it answered my problem anyway, I need the a extension to be created.
18:19.40jradThanks for your inputs. I've created official bugtracker @ https://issues.asterisk.org/jira/browse/ASTERISK-19596
18:19.47techwerkzIf the dial plan contains a Macro() does it continue to execute the next line after Macro(), or does it jump to the Macro and stay there?
18:26.00[TK]D-Fenderglaz, Let us know if there's anything else we don't need to do for you ;)
18:26.15[TK]D-Fendertechwerkz, "core show application MacroReturn"
18:27.40*** join/#asterisk tzafrir (~tzafrir@212.179.75.202)
18:28.26glaz[TK]D-Fender: :p
18:39.53*** join/#asterisk YelBo (~YelBo@81.130.125.130)
18:40.46YelBohey
18:41.25glazho!
18:41.32YelBo:)
18:43.22autofsckkhello, i want to use a voip provider to usewith my asterisk, any recomendations? i just want sip trunking with good price
18:43.34techwerkz[TK]D-Fender: Doesn't find that, in all reality I don't want it to return
18:46.59ryduhautofsckk: i use voip.ms and have been pretty happy with their service
18:47.36techwerkz[TK]D-Fender: Basically just want to make sure this will be ok: http://pastebin.com/WtbcpC3L or if I need to add something to stop the dial plan after the Macro commands.
18:48.37*** join/#asterisk harovali1 (~harovali@r190-134-146-155.dialup.adsl.anteldata.net.uy)
18:51.44harovali1hi, my call center provider uses asterisk to manage calls, and he lets me download the audio files of the incoming calls so that I can hear them, I do that thru a web of his, where the audio files can be downloaded. The problem is that I can hear the audo files, posibly because I'm missing the correct codec of file type. I couldn't determine which audio files are these. I tested renaming the files as .mp3, .avi, .wav, an
18:52.44harovali1I used the 'file '  command in linux to see if the audio format could be guessed, but it just says it's binary data
18:52.50harovali1any hint ?
18:56.07autofsckkryduh: thanks ill take a look
18:56.56vltharovali1: Assuming you *can’t* hear them: Asterisk’s native format is 8bit 8kHz PCM data. I thought nearly everything could play that. Maybe your provider uses some codec like OGG or weird things like AMR …
18:57.21*** join/#asterisk jdoe (jdoe@falseprophet.ca)
19:00.04SupYoshihello
19:00.11SupYoshiwhen i run my make and make install commands
19:00.15SupYoshiIt breaks off with KILL
19:00.23SupYoshiThere is no way I can install asterisk on CentOS 5
19:00.27SupYoshiI have tried 5 times now
19:00.31SupYoshiFollowing different guides
19:00.52*** join/#asterisk GVolkmann (~administr@rrcs-24-213-178-60.nyc.biz.rr.com)
19:00.52harovali1vit: now he told me they are in GSM format
19:01.14*** join/#asterisk micols (~t@rlogin.dk)
19:02.14vltharovali1: Aah, gsm. Ok. Can your asterisk play the files? (That’s what I’d try first.)
19:03.09harovali1vlt: I don't have asterisk installed, do you think installing asterisk to hear the files is my best bet under linux ?
19:04.25*** join/#asterisk bmg505 (~leon@196-209-152-75.dynamic.isadsl.co.za)
19:05.20*** join/#asterisk bmg505 (~leon@196-209-152-75.dynamic.isadsl.co.za)
19:06.08*** join/#asterisk gonewage (~gonewage@72.2.130.205)
19:06.17*** part/#asterisk gonewage (~gonewage@72.2.130.205)
19:08.46*** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt)
19:09.00[sr]hellou my friends
19:10.37jayteeanyone have any good links for setting up just the Buddy Watch feature on Polycoms with Asterisk 1.6.2. I'm setting up a client with one Polycom 560 as the receptionist phone and another 19 Polycom 331s for everyone else. I've found a few articles but one is out of date and the others aren't walkthroughs so much as postings for people trying to figure out their own issues. I want to use the
19:10.37jayteeBuddy Watch feature with a softkey and not use up line keys for BLF.
19:10.39vltharovali1: I just tried to play tt-weasels.gsm on Ubuntu with vlc. Worked. mplayer doesn’t.
19:12.12jayteeSupYoshi, installing asterisk on CentOS is fairly straightforward. What version of CentOS are you trying to use?
19:12.44*** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
19:13.42Naikrovekjaytee: never done it with a softkey.  efk is insanely frustrating.
19:13.56Naikroveklike "i NEED to kill someone right now" frustrating
19:14.11*** join/#asterisk wudles (~wudles@gateway.secureinstrument.com)
19:25.12*** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d)
19:28.54mr_peteo_O
19:28.58mr_petejust defined a class called [none] in musiconhold.conf
19:29.08*** join/#asterisk bmg505 (~leon@196-209-152-75.dynamic.isadsl.co.za)
19:29.12mr_petewhich uses a directory of /dev/null
19:54.08*** join/#asterisk infobot (~infobot@rikers.org)
19:54.08*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.2.1 (2012/03/15), 1.8.10.1 (2012/03/15), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org
19:54.16[TK]D-Fenderdanmuniz, As you said, you're new.  Let our eyes filter it for you
19:55.02danmunizthanks, have to run an errand for a couple of hours will you be on around 3:00 pst?
19:55.21[TK]D-Fenderdanmuniz, Share whatever you can, whenever you can, and whoever is available to help you will.
19:56.30mr_peteright
19:56.32mr_petesort of working
19:56.42mr_petelocally I have a silent gsm file, and set the format in the moh.conf to gsm
19:56.46mr_peteit plays silence locally
19:57.26mr_peteyep, working on trunks too
19:57.26*** join/#asterisk jsjc (~Adium@199.Red-79-150-67.dynamicIP.rima-tde.net)
19:57.34mr_peteI need to point to a dir with a file (no files doesn't work)
19:57.44mr_peteand given the silent file I used is in gsm format, I had to set format=gsm too :)
20:00.15*** join/#asterisk ke-esc (~ke-esc@155.229.209.170)
20:01.02ke-escDump question- is there a rule that a * cannot appear at the beginning of an extension? _*XXX always gives me a fast busy, but if I change it to _XXX* and reload the dialplan it works fine
20:02.43leifmadsenke-esc: no such rule exists -- most likely your phone is the problem
20:03.10leifmadsenke-esc: change the phones dialplan to permit your 3 digit starcode to be sent to asterisk correctly
20:04.00*** join/#asterisk jero (~boo@mtl.savoirfairelinux.net)
20:05.48ke-escleifmadsen, hmmm...okay, thanks :)
20:06.49jerohi
20:08.23Delido192123Please, i have this problem since few weeks, i want to get ccbs working, without call-limit 1 (because the Phoneuser can enable or disable callwaiting over an intranet) code http://pastebin.com/hqpk1CQ3
20:12.18jerois it possible to disable presence status updates from the dialplan? I have a bunch of extensions and a polycom with lots of BLF hints, who don't like Pages on the mid/longterm because they cause too many simultaneous updates
20:19.02*** join/#asterisk AutoStatic (~jeremy@5ED0E47D.cm-7-1d.dynamic.ziggo.nl)
20:20.49Delido192123i use the latest asterisk LTS Version and Polycom IP 320 Phones with the latest SIP and Bootrom. used Dialplan is copied in pastebin ( http://pastebin.com/hqpk1CQ3 ) i used an phpagi get the status from callwaiting and others.. so if DEVSTATE INUSE the Caller become directly an CALLWAITING menu
20:21.07*** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith)
20:21.26Delido192123can i used asterisk Function CALLCOMPLETION to get it to work?
20:21.32leifmadsenDelido192123: with 1.8 you don't use call-limit -- you use callcounter=yes
20:21.42leifmadsencall-limit is deprecated
20:23.38*** join/#asterisk tzanger (tzanger@wallace.mixdown.ca)
20:24.02*** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66)
20:24.10Delido192123leifmadsen: thanks for answer. callcounter=yes and  limitonpeers=yes is set in sip.conf [general]
20:24.59leifmadsenok
20:26.11*** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za)
20:26.34bent_screwdriveris it best to use iax or sip for asterisk box to asterisk box calling? i have been rolling out with iax then using sip as a second option if iax fails for some reason but just wondering if it's worth the trouble to even setup the sip part...
20:27.45vltbent_screwdriver: I have better experience with IAX (and trunking) when bandwidth is limited.
20:34.49dijibwould anybody in here have any use for a sangoma a-200 ? im looking for a trade for ram or an hdd or something
20:35.25dijibata ... smething
20:35.29leifmadsenbent_screwdriver: I always just use SIP everywhere and don't bother with IAX2
20:42.27bent_screwdriverleifmadsen: thanks. particular reason or just preference?
20:43.30leifmadsenbent_screwdriver: never saw a reason to change technologies between locations as it just causes problems with features
20:43.46leifmadsenkeeping it all on the same protocol tends to make things a lot easier to implement
20:49.34Delido192123no one can help? what i need: The callwaiting can ondemand changed over webinterface; All User can do more then one Call (only for callforwarding); If is callwaiting off, and the called phone in use ccbs should work (this is my problem)
20:52.48*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
20:53.27*** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za)
21:06.06Delido192123it is posible to set an variable in dialplan, that let the dial command not called the called phonenumber? and get causecode back? i think ccbs only work in my dialplan if the dialcommand is executed
21:17.01*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
21:26.38*** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
21:26.40*** join/#asterisk Neptu (~Neptu@213-67-244-97-no100.tbcn.telia.com)
21:30.11*** part/#asterisk danmuniz (~danmuniz@ip72-197-14-232.sd.sd.cox.net)
21:40.14*** join/#asterisk smps (~smps@brk.ehet.net)
21:43.04*** join/#asterisk gonewage (~gonewage@72.2.130.205)
21:48.48*** join/#asterisk lanning_ (~lanning@32.157.143.31)
21:50.49*** join/#asterisk lanning- (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
21:51.19*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
21:55.12*** join/#asterisk serafie (~erin@nat/digium/x-zqgxdhskedqjszvc)
21:58.00*** part/#asterisk serafie (~erin@nat/digium/x-zqgxdhskedqjszvc)
22:03.45*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net)
22:09.22*** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za)
22:13.24*** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za)
22:25.24*** part/#asterisk mjordan (~mjordan@nat/digium/x-kvgmzrsthzwsaduc)
22:27.57*** join/#asterisk pdtpatrick (~ptaylor@12.249.4.226)
22:40.07*** join/#asterisk nosaj (~jbarinas@gwb.anditel.com.co)
22:43.31*** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za)
22:44.32*** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za)
22:45.10*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
22:52.04*** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com)
22:52.49SupYoshiHi
22:52.58SupYoshican anyone tell me whats going on here now :)
22:52.58SupYoshihttp://pastebin.com/bnuuRbcc
22:53.11SupYoshiI end up with this everytime :) Its so confusing..... annoying frustrating et.c
22:53.17SupYoshimake: *** [pbx] Killed
22:54.00*** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d)
22:57.18*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
22:57.47*** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za)
23:00.54*** join/#asterisk Antonjo (~Antonjo@46.183.121.39)
23:01.18Antonjohello to all
23:01.53*** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za)
23:04.20Antonjocan somebody help
23:04.22Antonjo?
23:06.04*** join/#asterisk neurosys_ (~neurosys@c-98-254-216-32.hsd1.fl.comcast.net)
23:10.17navaismoAntonjo, ask an you will see
23:12.31Antonjoi have a problem
23:12.47Antonjowhen i do a call from a sip
23:12.47Antonjoi have 2 asterisk
23:13.06Antonjowhen i do the call and send to sip 1 wich is asterisk 1
23:13.10Antonjoi got these message
23:13.19*** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za)
23:14.46*** join/#asterisk Antonjo (~Antonjo@46.183.121.39)
23:14.58Antonjosorry but i disconect
23:17.23navaismonp
23:18.57navaismoso the error is...
23:19.40AntonjoNOTICE[7483]: chan_sip.c:21870 handle_request_invite: Sending                                            fake auth rejection for device
23:19.58Antonjowhen i do a call
23:20.03Antonjolet me explain
23:20.08Antonjoi have to asterisk
23:20.14Antonjoasterisk 1 and asterisk 2
23:20.35Antonjoi add a sip trunk to asterisk 2
23:20.50Antonjothe user is of sip i create in asterisk1
23:21.08Antonjowhen i call from asterisk 2 the call go to the asterisk1 and i got the message
23:21.12AntonjoNOTICE[7483]: chan_sip.c:21870 handle_request_invite: Sending                                            fake auth rejection for device
23:21.13navaismothat is a notice not an error
23:22.04Antonjoah ok
23:22.07Antonjoany idea?
23:22.13[TK]D-FenderIdea about what?
23:22.19[TK]D-FenderWe dont see a problem yet.
23:22.32Antonjobut  the call is not execudet
23:22.48Antonjoand i drop the call
23:22.49[TK]D-FenderAntonjo: PASTEBIN an actual failed call for us to look at.
23:22.56[TK]D-Fender~pb
23:22.56infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
23:22.57[TK]D-Fender^^^
23:23.09[TK]D-FenderAntonjo: "sip set debug on" <----------
23:23.22[TK]D-FenderAntonjo: Make sure to have SIP DEBUG enabled
23:24.10Antonjothis is from asterisk 2 when i connect my phone
23:24.25Antonjo<PROTECTED>
23:24.25Antonjo<PROTECTED>
23:24.25Antonjo<PROTECTED>
23:24.25Antonjo<PROTECTED>
23:24.25Antonjo<PROTECTED>
23:24.26Antonjo<PROTECTED>
23:24.26Antonjo<PROTECTED>
23:24.34Antonjothis is from asterisk when i have the trunk
23:24.38Antonjo[Mar 27 01:28:53] NOTICE[7483]: chan_sip.c:21870 handle_request_invite: Sending fake auth rejection for device "Laxon 1001" <sip:1001@46.183.121.39:1075>;tag=as2557bebc
23:24.38Antonjodc-asterisk01*CLI>
23:25.21navaismoAntonjo, use PB
23:25.39Antonjowhat is PB
23:25.39navaismoand your trunk seems dead
23:25.39Antonjo?
23:25.47navaismo~pb
23:25.47infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
23:26.14navaismoshow us the output of: sip show peers and use PB
23:27.13Antonjo*****3007                ******                               N      5060     OK (2 ms)
23:27.20Antonjothis is the trunk
23:27.51navaismobut you are using "mc"
23:28.40Antonjoues mc is the trunkname
23:28.41[TK]D-FenderAntonjo: I asked you to show us the actual call attempt
23:28.56Antonjoyes this is the actual call atempt
23:29.00[TK]D-FenderAntonjo: Antonjo That does not mean anything will work.
23:29.18*** join/#asterisk dijib (~root@bas10-kitchener06-1176145006.dsl.bell.ca)
23:29.19[TK]D-FenderNo, it isn't.  There is no pastebin, in which there should be DOZENS of lines of debug for us to see
23:30.02*** join/#asterisk Bullmoose (~Bullmoose@71-33-30-40.bois.qwest.net)
23:32.16Antonjook this is the log of the asterisk who i have a user to trunk to asterisk 2
23:32.24Antonjo[Mar 27 01:36:35] NOTICE[7483]: chan_sip.c:21870 handle_request_invite: Sending fake auth rejection for device "Laxon 1001" <sip:1001@46.183.121.39:1075>;tag=as6eb16fa0
23:32.51Antonjothis is the asterisk 2 logs when i try to make a call
23:33.52Antonjo<PROTECTED>
23:33.52Antonjo<PROTECTED>
23:33.52Antonjo<PROTECTED>
23:33.52Antonjo<PROTECTED>
23:33.52Antonjo<PROTECTED>
23:33.52Antonjo<PROTECTED>
23:33.52Antonjo<PROTECTED>
23:33.53Antonjo<PROTECTED>
23:33.53Antonjo<PROTECTED>
23:34.12*** join/#asterisk brian98 (~brian98@188.141.12.34)
23:34.57navaismoUSE PB
23:35.08navaismoor you will be banned by the admins
23:35.40navaismoand still going to MC trunk wich appear offline
23:35.47navaismoshow us your sip show peers
23:36.00navaismoand use PB
23:36.20*** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net)
23:36.20*** mode/#asterisk [+o mjordan] by ChanServ
23:36.39Antonjoi translate whith pb but is the same
23:36.43Antonjo:)
23:37.20navaismoi know but: " A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel."
23:37.44Antonjo== Using SIP RTP TOS bits 184
23:37.44Antonjo<PROTECTED>
23:37.45Antonjo<PROTECTED>
23:37.45Antonjo<PROTECTED>
23:37.45Antonjo<PROTECTED>
23:37.45Antonjo<PROTECTED>
23:37.45Antonjo<PROTECTED>
23:37.46Antonjo<PROTECTED>
23:37.46Antonjo<PROTECTED>
23:37.52Antonjothis is the translate of PB
23:38.30navaismofacepalms
23:40.32*** join/#asterisk bmg505 (~leon@196-209-10-231.dynamic.isadsl.co.za)
23:40.45Antonjoany idea of this message
23:40.46Antonjo?
23:41.22[TK]D-FenderAntonjo: Meaningless because you did not follow any of the directions you were gien on this.
23:41.41[TK]D-FenderAntonjo: Enable SIP DEBUG as I told you.  and then PASTEBIN it.  www.pastebin.com <---
23:42.08[TK]D-FenderAntonjo: give us the LINK to go read what you submit.  Do not ever flood in here like you just did twice.
23:51.35Antonjowhat links

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.