IRC log for #asterisk on 20111230

00:22.07*** join/#asterisk zerohalo (~zerohalo@74.60.136.128)
00:27.43*** join/#asterisk [Outcast] (~anonymous@pool-96-252-45-211.bstnma.fios.verizon.net)
00:41.44*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
00:44.01atanHas anyone managed to get a sidecar working with a Cisco IP phone running SIP firmware?
00:44.10*** join/#asterisk uofiguy (0caf924a@gateway/web/freenode/ip.12.175.146.74)
00:51.20uofiguya
01:07.47*** join/#asterisk SeRi (~wtf@c-76-31-169-54.hsd1.tx.comcast.net)
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01:25.51*** join/#asterisk vinhdizzo (~vinh@pool-74-100-182-10.lsanca.fios.verizon.net)
01:35.14carraratan you might try http://chan-sccp-b.sourceforge.net/
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03:43.22F2KnightQ: Has anyone using dahdi seen the channel report as DADHDI/i1 with the 'i'?
03:43.33F2Knightvs the normal g/G
03:48.25*** join/#asterisk JanSch (~JanSch__@p4FD78E82.dip0.t-ipconnect.de)
03:51.35JanSchI am trying to use asterisk to proxy SIP to JACK. I register with a SIP provider, create an extension that Accepts and then calls the JACK application. unfortunately, what I get on the JACK client is noise.
03:53.53JanSchthe other direction works
03:54.06JanSchi.e. other jack application -> phone calling the SIP number
04:09.45*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
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05:53.48*** join/#asterisk asteriskmonkey1 (~darlek@74-51-38-204.telnetcommunications.com)
05:54.01asteriskmonkey1why no res_fax_digium for freebsd?
06:04.50ChannelZracists!
06:11.01coppiceasteriskmonkey1: why is that a problem? use spandsp
06:23.24*** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl)
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07:48.35*** join/#asterisk IsUp (~nocturne@unaffiliated/isup)
07:48.52IsUphello
07:49.05*** join/#asterisk rhce7320 (~rhce7320@59.167.200.141)
08:07.50*** part/#asterisk ayrjola (~ayrjola@89.18.236.11)
08:09.23*** join/#asterisk jkroon (~jkroon@dsl-242-2-75.telkomadsl.co.za)
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08:32.09*** join/#asterisk krotos (~d3v1l@host146-98-dynamic.11-79-r.retail.telecomitalia.it)
08:32.12krotoshi all :)
08:33.56IsUphi
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08:34.54*** join/#asterisk Nasga (~Nasga@112.4.118.78.rev.sfr.net)
08:36.22krotosi'have a curiosity about the command core show translation..
08:36.30krotosspecially after migration from 1.4 to 1.8
08:38.03krotosthe conversion timing is higher than in 1.4, i think that the codec on my asterisk is not the right
08:38.27krotosfor "the codec" i intend the codec_g729.so module that i'm using is not the right one
08:39.13jkroonkrotos, use the benchg729 tool to figure out what the best variant for your system is
08:39.50jkroonalso note that the translations are only used in cases where you need to bridge between different codec types, on a (imho) properly configured setup this should be almost never.
08:46.47*** join/#asterisk ayrjola (~ayrjola@89.18.236.11)
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09:09.31*** part/#asterisk e7e5 (~rudenko@188.134.2.33)
09:11.33*** join/#asterisk jermey_g (~test@gw-sthlm01.rebtel.com)
09:13.54jermey_gHi, How can I validate whether dhadi_dummy is loaded and performing properly? I am using 1.4.43. I have seen some posts that loading dhadi_dummy is not necessary anymore as timings now is generated internlly by the dhadi driver itself. In that case, how do I know which kernel module is being used as timing reference (rtc or?)
09:14.11jermey_gs/timings/timing
09:32.21*** join/#asterisk irroot (~gregory@197.110.170.161)
09:33.00*** part/#asterisk irroot (~gregory@197.110.170.161)
09:36.30*** part/#asterisk Poincare (~jefffnode@2001:470:cb24:4::2)
09:37.54krotosthankyou jkroon
09:39.17krotosjkroon: in my config i set only alaw / g729 / gsm, same config on ata's
09:39.33jkroonkrotos, why not pick just one?
09:40.10jkroonyou want to avoid having to transcode at very near all cost.  transcoding == bad for quality
09:41.02jkroonpersonally uses g729 only, and enable t38 with a few patches from irroot for those rare cases that he needs to bridge with t.30 going into dahdi.
09:41.59jkroonand have g729 licenses obviously for bridging between voip channels and dahdi :p
09:42.27krotosactually, i use too irroot patch ( WaitFAX ) for faxes
09:43.06krotosin some case, the client say that it not hear so good
09:43.13krotosand i switch from g729 to alaw
09:48.12jkroonWaitFAX doesn't work for me.
09:48.52jkroonit leaves the channel in a state from where T.38 negotiation fails once entering into Send/ReceiveFAX (can't remember which one).
09:49.18jkroonand no, when using WaitFAX with anything other than ulaw/alaw, don't expect it to work (noise distortion)
09:50.13jkroonyou need to let the end-points (ie, either the Send/ReceiveFAX applications, or the remote SIP endpoints) initiate the t38 renegotiation
09:51.18jkroonfax sometimes even works on a VERY low jitter+latency LAN using t30 (audio) mode with alaw/ulaw.  sometimes.
09:56.12florzlatency is irrelevant
09:57.07jkroonflorz, that's interesting.
09:59.40florzyou can fax over intercontinental PSTN connections, so why should IP induced latency be any different?
10:01.58krotosjkroon: what branch of iroot are you using?
10:02.36jkroonthe stuff he back-ported to 1.8
10:02.46jkroonthe gateway code.
10:04.06krotosok, me too..another bug found
10:04.17krotoson my asterisk box
10:04.38krotosnow i start to prepare the new one..but benchg does not start without licence key
10:04.42krotosi've to call directly digium
10:05.41jermey_ghow would i know if i have configured dhadi right. i donnt have any voip hardware and i can use dhadi commands from asterisk CLI. i need dhadi only for timing source. i am using 1.4.43 and my dahdi-linux complete-2.5.0.2 + 2.5.0.2. In asterisk cli> dahdi show status, shows nothing, not even dhadi_dummy.
10:05.52krotosfor re-enable my g729 licence key ? My asterisk box is on VMWare ESXi, and i have to move on another server (more performance), but i'm sure that change mac address of network interface card..
10:05.57krotosand so the lic key is not valid :(
10:09.08jkroonjermey_g, if you only need dahdi for timing consider rather using timerfd
10:09.27jkroonkrotos, correct.
10:09.38jkroonand benchg729 does not function without a license key either.
10:09.52jkrooni've inquired and digium said that they won't allow it.
10:10.12krotosit's only a bench..not the codec :(
10:11.02jkroonstill requires a license.  i had this arguement with them, and i can understand where they are coming from, but honestly, as you say, it's only a benchmark tool, not a real coder.
10:12.27*** join/#asterisk tzafrir (~tzafrir@212.179.75.202)
10:18.07jermey_gtzafrir:do you work with timerfd as well? i am just trying to validate if my dhadi install/config is correct
10:19.14jermey_gtzafrir: i am using 1.4.43 and my dahdi-linux complete-2.5.0.2 + 2.5.0.2. In asterisk cli> dahdi show status, works but shows nothing. dhadi_test generates numbers. is my dhadi install correctly? is there a way to verify this.
10:20.00*** join/#asterisk master_of_master (~master_of@p57B55733.dip.t-dialin.net)
10:21.18jkroonjermey_g, is there a reason you're using asterisk 1.4 and not 1.8/10 ?
10:21.29*** join/#asterisk benlangfeld (~ben@unaffiliated/benlangfeld)
10:28.55jermey_gjkroon:yes.
10:29.15jkroonok, let's see, dahdi_dummy manages to load?
10:29.33jkroonit's integrated into dahdi "core" module now, so lsmod should only show "dahdi"
10:31.50jermey_gjkroon:then the install is correct. because i saw tzafrir's post that he posted a year ago on reviewboard in which he suggested using the internal dhadi_timer making dhadi_dummy obsolete and therefore it won't be compiled at all. what i wonder is that which kernel timer does this dahdi use. is it rtc,hi_res? i can't see a kernel module. i am on 2.6.18 kernel which doesn't have HIRES
10:32.55jkrooninternal HZ clocking at 1000HZ perhaps.  no idea really.  not sure if 1.4 has this command, but in the cli, can you run "timing test" ?
10:33.37kaldemarthere is no separate dahdi_dummy in 2.5.0.2. the timing feature was merged to the core module in an earlier version.
10:34.18*** join/#asterisk timahvo1 (~rogue@197.177.18.211)
10:38.28*** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net)
10:39.34jermey_gkaldemar:what is it using from the kernel as clock reference - usb hpet, rtc...?
10:39.57*** join/#asterisk prasanthprabalan (~root@218.248.24.19)
10:43.18jermey_ghas someone experienced this bug in asterisk 1.4 that when it transcodes ulaw to ilbc, the rtp timestamps get screwed up and as a result the call quality gets real bad.
10:44.05jermey_gthe timestamps are wrong. it'll be great if someone can relate that to a bug and when was it fixed. i can't leave 1.4 for certain reasons.
10:44.57rhce7320I have set up kiax on 6 PCs - Win, Mac, Linux faultlessly.  This one has me stumped.  An Acer netbook.  Kiax only emits registration packets.  It emits ___no___ packets when placing a call.  Same result in Wireshark with firewall & AV off.   Asterisk logs & debug are silent while the Kiax call appears to be "ringing".  After 2 unsucessful nights, I installed X-lite,.. config in sip.conf & it works fine. :/
10:46.06jkroonjermey_g, may i inquire as to those reasons?
10:46.11jermey_gwill this ilbc transcoding bug be fixed if i start using dahdi. currently i dont use any timing generator.
10:46.28jermey_gjkroon:its a customer requirement :)
10:46.57jkroondoubt it ... that doesn't actually have anything to do with timing per sé, rtp (unless you force jitter buffers) will simply shuffle the RTP as quickly onwards as it can.
10:47.06kaldemarjermey_g: afaik, hpet.
10:47.07jkroonjermey_g, i hate those arb requirements.
10:47.27jermey_gjkroon:you are not alone.
10:47.45jkroonbut yea, setting internal timing = yes is still a good idea for things like Playback() and so forth.
10:48.22jkroondo you know WHY the client is adamant about 1.4?
10:48.46kaldemarrhce7320: sounds like a mere Kiax issue.
10:49.32rhce7320kaldemar: Time to tickle the kiax forms...
10:49.55*** join/#asterisk s[X] (~mark@ppp59-167-157-96.static.internode.on.net)
10:50.08jermey_gjkroon:not yet. i am wondering if the dahdi timing and rtp timestamps in the transcoding have a connection. may be using dahdi means that asterisk gets a more accurate clock reference so it timestamps more accurately. Currently, the delta between the rtp packets vary considerably.
10:50.51jermey_gjkroon:timerfd might not work with 1.4. i just got this impression from little googling.
10:51.01jkroonjermey_g, it won't.
10:51.44jkroonjermey_g, that's strange.  so the timestamp increment in successive rtp packets doesn't use a constant delta?
10:53.48*** join/#asterisk timahvo1 (~rogue@197.178.209.144)
10:58.44*** join/#asterisk ltd (~z@zx.io)
11:06.13prasanthprabalanHi all..
11:06.15prasanthprabalanI have connected my analog phone in port number 1(FXS) and PSTN line into port number 4(FXO) of Digium TDM410P card. But not getting the dial tone on my analog phone. I have tested with two TDM410P card and getting the same message on CLI. So i think not the error with the card. This is the message am getting in the CLI> :
11:06.19prasanthprabalan== Using SIP RTP CoS mark 5
11:06.21prasanthprabalan-- Executing [6003@users:1] Dial("SIP/sai-00000003", "DAHDI/g1/6003,20") in new stack
11:06.23prasanthprabalan-- Called g1/6003
11:06.25prasanthprabalan-- DAHDI/1-1 is ringing
11:06.35prasanthprabalan-- Nobody picked up in 20000 ms
11:06.39prasanthprabalan-- Hungup 'DAHDI/1-1'
11:06.41prasanthprabalan-- Auto fallthrough, channel 'SIP/sai-00000003' status is 'NOANSWER'
11:06.43prasanthprabalanThese are my configurations:
11:06.45prasanthprabalan;chan_dahdi.conf
11:06.47prasanthprabalan[channels]
11:06.49prasanthprabalan;
11:06.51prasanthprabalanusecallerid=yes
11:06.53prasanthprabalanhidecallerid=no
11:06.55prasanthprabalancallwaiting=yes
11:06.57prasanthprabalanthreewaycalling=yes
11:06.59prasanthprabalantransfer=yes
11:07.01prasanthprabalanechocancel = yes
11:07.03prasanthprabalanechocancelwhenbridged = yes
11:07.05prasanthprabalanrxgain = 0.0
11:07.09prasanthprabalantxgain = 0.0
11:07.11prasanthprabalangroup=1
11:07.13prasanthprabalancontext=users
11:07.15prasanthprabalansignalling=fxo_ks
11:07.17prasanthprabalancallerid="Bob Analog" <6003>
11:07.19prasanthprabalanmailbox="6003"
11:07.21*** join/#asterisk AmirBehzad (~behzad@79.127.53.248)
11:07.22prasanthprabalancallwaiting=yes
11:07.23prasanthprabalanthreewaycalling=yes
11:07.25prasanthprabalantransfer=yes
11:07.27prasanthprabalanchannel => 1
11:07.29prasanthprabalangroup=2
11:07.31prasanthprabalancontext=from_outside
11:07.33prasanthprabalansignalling=fxs_ks
11:07.35prasanthprabalanechocancel=yes
11:07.40prasanthprabalancallerid=asreceived
11:07.41prasanthprabalancallwaiting=no
11:07.43prasanthprabalanchannel=>4
11:07.45prasanthprabalan--------------------------
11:07.47prasanthprabalan;extensions.conf
11:07.49prasanthprabalan;
11:07.51prasanthprabalan[general]
11:07.53prasanthprabalanstatic=yes
11:07.55prasanthprabalanwriteprotect=no
11:07.57prasanthprabalanautofallthrough=yes
11:07.59prasanthprabalanclearglobalvars=no
11:08.01prasanthprabalanpriorityjumping=no
11:08.03prasanthprabalan[globals]
11:08.05prasanthprabalan[users]
11:08.09prasanthprabalanexten => 5001,1,Dial(SIP/sai,20)
11:08.11prasanthprabalanexten => 6003,1,Dial(DAHDI/g1/${EXTEN},20)
11:08.13prasanthprabalan[from_outside]
11:08.15prasanthprabalanexten => 243,1,Answer()
11:08.17prasanthprabalanexten => 243,2,Dial(DAHDI/g1,20,rt)
11:08.19prasanthprabalan-------------------------------------------------
11:08.21prasanthprabalan;system.conf
11:08.23prasanthprabalan# Autogenerated by /usr/sbin/dahdi_genconf on Fri Dec 30 16:23:47 2011
11:08.25prasanthprabalan# If you edit this file and execute /usr/sbin/dahdi_genconf again,
11:08.27prasanthprabalan# your manual changes will be LOST.
11:08.29prasanthprabalan# Dahdi Configuration File
11:08.31prasanthprabalan#
11:08.33prasanthprabalan# This file is parsed by the Dahdi Configurator, dahdi_cfg
11:08.35prasanthprabalan#
11:08.39prasanthprabalan# Span 1: WCTDM/0 "Wildcard TDM410P Board 1" (MASTER)
11:08.41prasanthprabalanfxoks=1
11:08.43prasanthprabalanechocanceller=mg2,1
11:08.45prasanthprabalan# channel 2, WCTDM/0/1, no module.
11:08.47prasanthprabalan# channel 3, WCTDM/0/2, no module.
11:08.49prasanthprabalanfxsks=4
11:08.51prasanthprabalanechocanceller=mg2,4
11:08.53prasanthprabalan# Global data
11:08.55prasanthprabalanloadzone = us
11:08.57prasanthprabalandefaultzone = us
11:08.59prasanthprabalanIf ha
11:09.01prasanthprabalanIf having any idea in this problem plz help
11:10.44*** join/#asterisk kaushal (~kaushal@14.97.27.82)
11:11.39*** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com)
11:12.11EmleyMoorAsterisk itself is presumably fine on IPv6 - but what about typical IP phones? Can they use it?
11:13.36kaushalHi
11:14.50prasanthprabalanHello Plz check the configurations above and plz give suggestions
11:17.19kaldemarprasanthprabalan: please use pastebin instead of pasting stuff on the channel.
11:17.22kaldemar~ps
11:17.22infobotps can be found in the procps ipkg
11:17.27kaldemar~pb
11:17.27infobot[~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.
11:18.39kaushalkaldemar: hi
11:19.22kaushalCLI> cdr mysql status says No such command 'cdr mysql status' (type 'core show help cdr mysql' for other possible commands)
11:20.12kaushalAny clue ?
11:20.20kaldemarkaushal: you don't have cdr_mysql.so loaded
11:20.28kaushalok
11:21.18kaushalkaldemar: is it module load cdr_mysql.so ?
11:21.33prasanthprabalanHi all
11:21.59kaushalError loading module 'cdr_mysql.so': /usr/lib/asterisk/modules/cdr_mysql.so: cannot open shared object file: No such file or directory
11:22.48kaldemarkaushal: how did you install asterisk and what version are you using?
11:23.12kaushalAsterisk 1.8.5.0
11:23.26kaushalusing yum install asterisk on centos 5.6
11:23.47prasanthprabalanI have a problem in getting dial tone in analog phone connected to Digium TDM410 module
11:24.31kaldemarkaushal: install a package that has the cdr_mysql module then. i don't know which exact package has it in centos but it might be packaged to addons since that's where it is in source.
11:24.32*** join/#asterisk s[X] (~mark@ppp59-167-157-96.static.internode.on.net)
11:25.42prasanthprabalanThis is the message am getting in CLI while dialing to the analog phone
11:25.45prasanthprabalanhttp://pastebin.com/DPkQF8sL
11:25.46kaldemarprasanthprabalan: your output has nothing to do with dial tone.
11:25.52kaushalkaldemar: Thanks
11:26.04kaldemarprasanthprabalan: are you trying to say that the phone is not ringing?
11:26.30prasanthprabalanYes phone is not ringing and dead.
11:26.39prasanthprabalanNot giving any response
11:26.58kaldemarprasanthprabalan: is it silent when you lift the handset?
11:27.35prasanthprabalanYes it is silent
11:27.44kaldemarmaybe the phone or the cable is broken.
11:27.50kaushalkaldemar: it worked
11:28.02kaushalkaldemar: Thanks
11:28.15prasanthprabalanNo i have checked all those hardware stuffs.
11:28.30prasanthprabalanNo problem with the connections and the phone
11:29.12kaushalkaldemar: I have still not made any progress on exten => h,4,Set(CDR(userfield)=${DIALSTATUS}) Hangupcause:${HANGUPCAUSE})
11:29.21kaushalAny clue please ?
11:30.33*** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl)
11:31.47jacc0I want to send you all the best wishes for 2012. The asterisk dev team realy did a great job in 2011 making 1.8 stable as it is now
11:32.12kaushaluserfield is empty
11:32.17prasanthprabalanThese are my configurations.
11:32.20prasanthprabalanhttp://pastebin.com/uqnjtCFt
11:32.26prasanthprabalanPlease check this
11:32.46kaushaljacc0: wish you the same
11:34.56kaldemarkaushal: Set(CDR(userfield)=${DIALSTATUS}) Hangupcause:${HANGUPCAUSE}) <--- that is garbage. what is the problem?
11:35.15kaldemarkaushal: and pastebin your whole dialplan and a CLI output of a call.
11:35.50kaldemarkaushal: if userfield does not get set, your h,4 does not get executed.
11:36.36kaushalkaldemar: sure
11:36.42kaushalkaldemar: please give me a moment
11:38.46prasanthprabalanhttp://pastebin.com/v33HLqmU
11:39.01prasanthprabalanmy dial plan and CLI out put
11:40.05kaldemarprasanthprabalan: how have you verified that the phone and the cable works and is properly connected?
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11:46.15prasanthprabalanAm having another pstn port here. Have connected the phone and the cable with that
11:46.22prasanthprabalanand it is working fine with that
11:46.52aberriosprasanthprabalan, and you've connected the phone to the correct port on the card?
11:48.01prasanthprabalanYa I have connected my analog phone in port number 1(FXS) and PSTN line into port number 4(FXO) of Digium TDM410P card.
11:51.35kaushalkaldemar: http://fpaste.org/nenf/
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11:54.23kaldemarkaushal: that's not your whole dialplan
11:55.07aberrioskaushal, line 4 should be like h,1 and what is the Hangupcause:${HANGUPCAUSE}) all about???? And you've not shown your context names in the dial plan part
11:55.40kaushalok
11:55.56kaushalkaldemar: please give me a moment
11:56.01kaushalaberrios: just a sec
11:56.36aberrioskaushal, anyway the Disposition field in CDR records what you're after, you'd end up with Disposition Fields and Userfield being the same for each record.
11:58.05kaushalok
11:58.47kaushalaberrios: as per http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
11:59.09kaushalI am actually interested in CANCEL status
11:59.17kaushalCall is cancelled. The dial command reached its number but the caller hung up before the callee picked up.
11:59.30kaushalso i get a NO ANSWER instead of CANCEL
11:59.55kaushalbasically i need to know what was the exact cause behing "NO ANSWER" ?
12:00.00aberrioskaushal, yes but the syntax for that line makes no sense.
12:00.01kaushalbehind*
12:00.08kaushalok
12:00.20kaushalaberrios: let me pastebin the entire dialplan
12:03.21aberrioskaushal, so you're after something like this: h,1,Set(CDR(userfield)=${HANGUPCAUSE})
12:03.40kaushalok
12:04.08aberrioskaushal, or you could add in another field to your CDR table called hangupcause and use h,1,Set(CDR(hangupcause)=${HANGUPCAUSE})
12:04.35aberriosproviding you put it in the from-pstn context
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12:06.58prasanthprabalanHello
12:07.09prasanthprabalanI have connected the link correctly
12:07.16prasanthprabalanAny idead wat is the problem
12:08.38aberrioskaushal, the hangup cause is actually there in the CLI output Span 2: Channel 0/1 got hangup request, cause 16.... but you're wanting to record this rather than just seeing it for debugging purposes?
12:09.10aberriosprasanthprabalan, do inbound calls from pstn work okay on the card?
12:09.38kaushalaberrios: please give me a moment
12:11.52prasanthprabalanActually phone is even not detecting.
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12:15.43prasanthprabalanHi
12:16.30prasanthprabalanThis is the error am getting for inbound calls
12:16.31prasanthprabalanhttp://pastebin.com/rt1ARXRR
12:16.37prasanthprabalanPlz check this
12:17.47aberrioshas dahdi been configured correctly for your countries pstn? iirc it looks like its not. Long time since i've been using dahdi.
12:18.08jacc0@prasanthprabalan, please do a : ' set verbose 10'
12:18.14jacc0before pasting the output
12:18.39aberrioscountry's *
12:19.09jacc0sorry : 'core set verbose 10'
12:19.41jacc0or you could create a extension = s,1,application()
12:19.46jacc0in your default context
12:24.07prasanthprabalanverboswe is already 10
12:24.32prasanthprabalanplz check the cli output while call come from outside
12:24.39prasanthprabalanhttp://pastebin.com/rt1ARXRR
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12:27.34kaushalaberrios: http://fpaste.org/2UQ4/
12:27.38kaushalkaldemar: http://fpaste.org/2UQ4/
12:28.37kaushalaberrios: also what does cause 16 mean ?
12:28.57kaushali get the same cause for couple of different tests
12:30.34aberrioskaushal, cause 16 is normal clearing
12:31.53kaushalaberrios: is there a wiki or documentation which says about various cause status ?
12:32.01aberriosyes somewhere
12:32.17aberrioswell 16 is normal clearing on ISDN lines here in the UK anyway
12:32.34aberriosie. "they hung up"
12:33.36kaushalaberrios: so in short i will have only disposition - ANSWERED, NO ANSWER, BUSY, FAILED. as call status ?
12:33.57aberrioskaushal, in the disposition field yes.
12:34.11kaushaland is there a way to set CANCEL as per http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS ?
12:34.22aberriosoh
12:34.43kaushalbasically like the exact cause of "NO ANSWER" ?
12:35.12kaushali mean in a more acceptable format
12:35.17aberrioskaushal, the exact cause of NO ANSWER should always be that the dial timeout was  reached
12:35.20aberriosi think
12:36.07aberriosare you hanging up before your dial timeout?
12:36.36kaushalyes
12:36.50aberriosshow me the cli output again?
12:38.35aberriosTry putting a timeout in your Dial functions of 30 seconds. Then ring the extension and hang up after 10.
12:38.47aberriosSee if that gives you a disposition of CANCEL
12:43.35jermey_gdid i miss anything regarding my DAHDI question and codec issue
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12:44.55kaldemarkaushal: cause 16 is normal clearing
12:45.59kaldemarkaushal: and there is no way to set DIALSTATUS to CANCEL.
12:47.24kaldemardisregard that, i have too much going on at the moment...
12:47.57aberrioskaushal, also you're best getting your info from https://wiki.asterisk.org/ rather than voip-info.org
12:48.13kaldemarkaushal: your hangup extension does not set userfield. you only have "exten => h,1,Set(CDR(hangupcause)=${HANGUPCAUSE})"
12:48.37kaldemarkaushal: add exten => h,n,Set(CDR(userfield)=${DIALSTATUS}) below it.
12:49.21kaldemarby DIALSTATUS and CANCEL i meant to say that there is no way of setting CDR(disposition) to CANCEL.
12:50.34kaushalkaldemar: ok
12:51.17kaushalkaldemar: Thanks
12:53.01kaushalaberrios: sure
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13:09.40jermey_gwhat entried in /dev would imply that dahdi is generating the timing (dahdi_dummy).
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13:12.34jermey_gdoes asterisk support iSAC?
13:12.47jermey_gits an audio codec supported by web-rtc ;)
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13:14.53kaldemarjermey_g: /dev/dahdi/timer and /dev/dahdi/pseudo
13:15.50kaldemarand iSAC is not supported by asterisk.
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13:29.49jermey_gyes, the /dev has those entries.
13:35.20kaushalaberrios: hi again
13:37.19kaushalaberrios: as per http://fpaste.org/a7Ab/
13:38.01kaushalline 9 as userfield as 16
13:38.13kaushalbut i am unable to reproduce anymore
13:38.27kaushalshall i pastebin the dialplan ?
13:41.31kaushalhttp://fpaste.org/rXMI/
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13:44.32jermey_gkaushal:some folks pastebin there sip.conf with passwords.
13:44.42aberrioskaushal, so its recorded the hang up cause in the userfield. If you want the Dialstatus change the CDR(userfield)=${HANGUPCAUSE} to CDR(userfield)=${DIALSTATUS} or you could record both CDR(userfield)=${HANGUPCAUSE}${DIALSTATUS}
13:48.10kaushalaberrios: ok
13:48.29kaushalaberrios: no luck
13:48.32kaushal:(
13:48.43kaushalexten => 6120,1,Set(CALLERID(num)=61816120)
13:48.43kaushalexten => 6120,2,Set(CDR(accountcode)=2000071214-test)
13:48.43kaushalexten => 6120,3,Dial(DAHDI/g1/09930963298)
13:48.43kaushalexten => h,4,Set(CDR(userfield)=${HANGUPCAUSE}${DIALSTATUS})
13:50.28aberriosshouldnt be h,4
13:51.19aberriosYou should read up on: context->extension->priority
13:51.29aberrioswould probably help you save a lot of time in the long run
13:51.41kaushalaberrios: sure
13:51.44aberriosit should be h,1
13:55.19kaushalaberrios: -- Span 2: Extension 6120@from-pstn does not exist.  Rejecting call from '9172980160'.
13:55.48kaushalhttp://fpaste.org/wg3s/
13:55.51kaushalweird
13:57.32aberrioswo wo..hang on there
13:57.42aberriosit should go
13:57.50aberrios6120,1
13:57.52aberrios6120,2
13:57.56aberrios6120,n etc tec
13:57.58aberriosthen
13:58.06aberriosh,1 h,n etc
13:58.26aberriosyou always need to have an extension with a 1 priority
13:59.35aberriosn means next priority
13:59.48aberriosmakes it easier when editing dial plan
14:05.07kaushalaberrios: got it now
14:05.08kaushalThanks
14:05.15kaushalit worked like a charm
14:05.24*** join/#asterisk serafie (~erin@nat/digium/x-kzjirmbbtsujdpxk)
14:05.28kaushalalso when i see the mysql query
14:06.18kaushalhttp://fpaste.org/98kZ/
14:06.29kaushalcan i set fieldname ?
14:07.08kaushalunder userfield as HANGUPCAUSE and DIALSTATUS
14:07.26*** join/#asterisk mjordan (~mjordan@nat/digium/x-kxngjowohylfmjxw)
14:07.34aberrioshow do you mean? like add another column to the Table?
14:08.04kaushalyes
14:08.25aberriosyeah that should work. jus tbe sure to change your dialplan to CDR(myfieldname) or whatever you call it
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14:15.15kaushalaberrios: great
14:15.27kaushalaberrios: as per http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause
14:15.51kaushaldo i need to refer ISUP Cause value SIP response in that link ?
14:16.17kaushali mean for hangupcause
14:17.48kaushalor as you said asterisk wiki ?
14:17.54kaushalbut i dont see it there
14:18.22aberriosthat just says you can have SIP response codes if you want them but only in 1.8, you're on 1.6 arent you~?
14:18.35kaushalaberrios: i am on 1.8
14:18.48aberriosah
14:19.07aberriosYou don't have to. its an option.
14:19.15kaushalaberrios: ok
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14:29.40WintermeWhi guys..got a question little off-topic, concerning SER..when using SER nathelper , do the clients behind NAT still need to use STUN to find their public ip + mapped port, or are they two complementary methods ?
14:31.35kaushalaberrios: any clue about [Dec 30 19:59:20] WARNING[17666]: app_dial.c:1353 wait_for_answer: Unable to write frametype: 2 ?
14:31.41kaushalI get this on CLI >
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14:40.56kaushalaberrios: anyways
14:41.00kaushalaberrios: Thanks a lot
14:41.05kaushalmuch appreciated
14:41.14kaushalkaldemar: Thanks again
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15:08.00mchounot so much an asterisk question, but I'm trying to build a debian pkg of * using checkinstall, but the build always fails complaining about sound files missing.  Anyone know how to get around this?
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15:46.17mchoumn, figured it out
15:46.47EmleyMoorDo IP phones generally support IPv6?
15:47.24*** join/#asterisk Sean-Der (~Sean-Der@108-90-184-35.lightspeed.toldoh.sbcglobal.net)
15:48.04Sean-DerWhat is the correct syntax to use DIAL() to play an announcement that doesn't interfere with the current call?
15:49.34EmleyMoorSean-Der: Why do you want to use Dial() to do it?
15:49.55*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
15:50.16Sean-DerI don't have to use dial its the only way I know how to call and play Announcements to other extensions
15:50.31Sean-DerSo if 2078 is called 2071 gets a call that plays an announcement
15:51.41EmleyMoorOh, you want to play the announcement to somewhere else... during or after the call?
15:52.13Sean-DerDuring without interfering with the current call being made. The only issue is the way I due it interrupts the current call path
15:57.06Sean-DerI guess the question is how can I btanch the event so it doesn't interupt the call flow?
16:00.53Sean-DerOr is it possible to execute two lines simultaneously, I know that isn't logical, but it is just an idea
16:01.42Sean-Derhttp://pastebin.com/s2U6yJxA
16:01.50Sean-DerHere is my stuff so far
16:04.47*** join/#asterisk brezular (~brezular@adsl-dyn219.91-127-179.t-com.sk)
16:10.49pabelangerEmleyMoor: I don't know any ATM
16:10.58pabelangerhard phones that is
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16:12.06leifmadsenSean-Der: not sure if my msg's came though... looks like I got disconnected
16:13.53EmleyMoorpabelanger: That's a shame... but I'll no doubt be able to come up with some kind of workaround if/when I need it...
16:14.31WIMPyEmleyMoor, pabelanger: Snom do since 7.3.14, or fully since 7.3.17 and above.
16:14.47pabelangerthe more you know
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16:17.16Sean-Derleifmadsen, Sorry I didn't get any of your messages
16:17.33leifmadsen<leifmadsen> Sean-Der: check out Cooking With Asterisk here:  http://leifmadsen.com/node/5
16:17.33leifmadsen<leifmadsen> Sean-Der: it's the last recipe in the slides
16:17.57Sean-DerThank you very much!
16:18.01mchouumm, I'm encountering a very puzzling issue with asterisk
16:18.37mchouContact: <<sip:1234@a.b.c.d:5060>▒>;expires=120
16:19.11mchouseems like there are some funky characters mucking things up
16:19.54mchouand ending up with "<<.*>>"
16:20.07*** join/#asterisk [Outcast] (~anonymous@pool-96-252-45-211.bstnma.fios.verizon.net)
16:20.14mchouanybody ever seen this behavior before?
16:20.19*** join/#asterisk pdeschen (~Adium@modemcable160.98-81-70.mc.videotron.ca)
16:21.01mchouincidently this is 1.8.8.0
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16:33.24Sean-Derleifmadsen, so I write an additional dialplan that Dials and Announce what I need. This dialplan is then executed by my matching context so it won't mess up the call flow?
16:33.51leifmadsenSean-Der: I'm unsure what you're asking
16:33.55leifmadsennot enough context
16:34.00Sean-DerOk sorry
16:35.17Sean-DerMy goal is that when someone calls an extension (2071) this call operates as normal, but Asterisk makes a new call to 2069 and plays an audio file
16:35.30Sean-DerYou sent me to the cooking with asterisk pdf on your website
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16:42.13Sean-Derleifmadsen, What I should be using is originate and I am reading up on it now to understand it
16:42.24Sean-DerJust a little slow have a couple things going on
16:44.05leifmadsenSean-Der: right, so when you place your call, you're going to need to use Local channels likely, and Originate() another call to the first call, and play audio back to it. I'm still not entirely sure what you're trying to do though, and what the scenario is. When are do you need to play audio? Who are you playing it to? There are several different ways to approach.
16:44.38leifmadsenOne of them could be to place a call, and drop all participants into a conference and play audio to the conference. Or you can use ChanSpy() like I've shown and whisper audio tot he channel you need to play audio to.
16:45.40Sean-Derleifmadsen, I am trying to implement that if one of my customers calls 911 it calls me and will play back an audio file saying "911 has been called" without interrupting the call to 911
16:46.24carrarAre you in the USA?
16:46.30Sean-DerYes
16:46.51carrarWhy not parse the cdr?
16:47.03leifmadsenthen ya, just use a pair of Local channels to execute two sets of dialplan -- one to the 911 service, another to your notification service (dialplan) and originate the other call using Originate()
16:47.04carraror have it fireoff a script that pages you
16:47.09leifmadsenthat too
16:47.27leifmadsenuse the M() or U() options potentially the triggers the script separate of the call
16:48.21mchouhow does one write an * regex that corresponds to this? "\+?[0-9]*@.*"
16:48.42mchou* complains that's not valid
16:48.53*** join/#asterisk bbourdage (~BBourdage@ip67-91-166-45.z166-91-67.customer.algx.net)
16:49.56mchoubasically match +12125551212@... or match 12125551212@...
16:51.15jkroonmchou, [+]?[0-9]+@.* possibly?  or double escape the \
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16:55.32Sean-Dercarrar, leifmadsen sorry for the stupid question, but two sets of Dialpan? I though the one that comes first procedurally is executed THEN the second?
17:00.26mchoujkroon: looks like \\ works.  thanks
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17:06.19LordRainwell
17:06.33LordRainim pleased to announce that upgrading from asterisk 1.0.08cvs to 1.8.7 went well lol
17:06.41LordRainnot really upgrading, but moving from, more so.
17:06.52LordRainA LOT of syntax changes
17:07.37LordRainand thank you guys for being helpful and nudging me in the right direction with some of those syntax changes
17:07.54LordRainsaved me a TON of reading - reading through the changelogs from 1.0 to 1.8 would have severely sucked. :)
17:08.37Sean-Derleifmadsen, carrar I am having a hard time figuring out what you guys were trying to tell me
17:08.38Sean-Derhttp://pastebin.com/ZdRhh2F5
17:08.41WIMPyThe UPGRADE*.txt are not that long.
17:09.10LordRainWIMPy - I'm not going to lie - I run out in to the dark a lot, so I rarely read changelogs or readmes lol ;x
17:09.42LordRainbut you helped big time by informing me that flags are now a paremeter for the VoiceMail application lol
17:10.05LordRainused to be <flag>box, now box,flag
17:10.16LordRainthat was the biggest stumper to me for some silly reason ;x
17:10.19WIMPyGood to leave the old stuff behind.
17:10.28LordRainim just baked
17:10.32LordRainits friday at 11:10 am
17:10.37LordRainand im up to 67 hours this week
17:10.39LordRainlol
17:10.46LordRainim leaving in 45 mins :P
17:10.59LordRaini did the upgrade in just one day, though
17:11.05WIMPyhad a 16h day yesterday.
17:11.15LordRainyeah, i feel your pain!
17:11.31LordRainthese guys were running asterisk 1.0.08cvs on a redhat 6 machine....
17:11.36LordRainlol
17:12.03LordRaini've got 1.8.7 running on a gentoo box now
17:12.07WIMPyIt wasn't that bad. At least 4h were just sitting in the car.
17:12.24LordRainlol, that would have made me nuts...
17:13.05WIMPyYes, I hate that as well, so better to have one long day thatn to drive twice.
17:13.30LordRainwell, im used to traveling for work - i've had three day drives because my old company was too cheap to pay for a plane ticket
17:13.36LordRaini hate sitting in cars...
17:13.40LordRainespecially mine.
17:13.55LordRaini drive an 02 mustang gt... looks great, but uncomfortable as HELL on long rides
17:14.21LordRaindriving three days in that loud monster will make anyones eyes and ears bleed and back feel like they were hit by a train
17:14.31WIMPyAudi A8 with driver is ok :-)
17:14.38LordRainnice :)
17:14.52LordRainim a mustang fanatic...
17:15.03LordRaini sold my 89 which i wanna slap myself for
17:15.09WIMPyYes, that thing is loud as well. Could almost feel my ears when I came home.
17:16.49LordRaini yanked the cats off the mustang and put ravin 1150cfm exhaust on - combined with the bbk long tubes? that thing gives me a headache sometimes
17:17.00LordRaini have 1100 watts  of sound in the car, and the exhaust actually drowns the music out
17:17.16LordRainim getting old now, so i might end up getting rid of the car for something a little more... quiet
17:17.17LordRainlol
17:17.24LordRaini want speed, but quiet speed
17:17.50LordRainbut most of all, i want torque with that speed...
17:17.56LordRainso looks like im going to have to go back to shopping again
17:18.02WIMPyOuch
17:18.09WIMPy(for the noise)
17:18.24LordRainim sure that audi rides SMOOTH though, too
17:18.36LordRaini lowered my mustang 2" with eibach lowering springs and put all steeda suspension
17:18.45LordRainhandles like a dream, rides like a dilapitated tank
17:19.03LordRainyou feel every pebble on the road... right in your neck lol
17:19.04WIMPythe suspension was very hard, yes.
17:19.55WIMPyYes, even on the Autobahn there were parts where you get a wobbely feeling.
17:20.15LordRainoh god i want to drive on the autobahn so bad...
17:20.24LordRainis there an actual speed limit on it?
17:20.24WIMPyBut the seats were very good. Lots of space.
17:20.48WIMPyGenerelly no, but often there is.
17:22.05WIMPyWe had a 120km/h limit for some period because of rain.
17:22.26WIMPyLuckily this is the area without many limits.
17:22.47LordRainso, when there is no speed limit, you could legally run full force?!
17:22.55WIMPyyes
17:23.17LordRainyep, gotta go there...
17:23.18LordRainLOL
17:23.24WIMPyThe general rule is jsut a _recommended_ speed of 130km/h.
17:24.12WIMPyOr as they said on the Channel Travellers Radio: "Please note the word 'recommendecd'."
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18:21.22Sean-DerCan I originate a call between An App and the receiver?
18:21.50Sean-DerI went to work on something else with out ever figuring out how to get this right
18:23.46[TK]D-FenderSean-Der: You Originate a call to a device (as you would Dial) and dump them into the dialplan on answer, or to a single application
18:24.29Sean-Der[TK]D-Fender, ok that makes sense. Will the originate stop the current dialplan until there is a hangup?
18:25.09[TK]D-FenderSean-Der: Once answered, there is no more "Originate".  It's work is done
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18:25.56[TK]D-FenderSean-Der: Wait... you're asking if the dialplan version of that is blocking?  Is that it?
18:26.30[TK]D-FenderSean-Der: To which I believe it was found to be "yes".  * CLI and AMI Originates can be used and are non-blocking
18:26.30Sean-Der[TK]D-Fender, I am trying to create two channels simultaneously that is my ultimate endgoal
18:26.55[TK]D-FenderSean-Der: remember that Originated channels have no relation to any other channels
18:27.33Sean-Der[TK]D-Fender, Thank you so much for your help! I think I have it all figure out. Just going to use originate to call an application which will contain all my logic
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18:30.26[TK]D-FenderSean-Der: Not making sense of what you're looking to do exactly...
18:30.55[TK]D-FenderSean-Der: What are you looking to continue doing after this starts off, and what is this other process supposed to do?
18:31.34Sean-DerWhen some calls the extension 2069 that channel is uninterrupted. A new channel is then created that calls 2070 an uses Playback() to play an audio file
18:32.43*** join/#asterisk jsdallas (~js@cpe-72-181-162-220.tx.res.rr.com)
18:35.58jsdallasQuestion: i am trying to setup my extentions.conf to answer all inbound calls is there any way that I can get that setup, I really do not want to setup 500 DIDs in the exten file.
18:36.18Sean-DerWildcard?
18:36.36WIMPyPatterns
18:36.50jsdallasI was reading and they do not seem to be working right or i am just not setting them up right.
18:36.55Sean-Derhttp://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
18:37.00Sean-DerI have this bookmarked
18:37.53[TK]D-FenderSean-Der: ok, don't use the dialplan app for that, use SHELL or System to launch it so you don't stall the primary channel
18:38.15jsdallasso dialplan and inbound are the same setup.  beucase I have tried XXXX for the blocks and i have tried ! and nothing seemed to work right.
18:38.57WIMPyyou must have done something wrong then.
18:40.35[TK]D-Fenderjsdallas: Then you didn't do it properly.  Show us what you did and the failed call
18:40.40[TK]D-Fender~pb
18:40.40infobot[~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.
18:40.42[TK]D-Fender^^^^^
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18:46.03Sean-Der[TK]D-Fender, http://pastebin.com/wmUd9iHs
18:46.09Sean-DerCrap I didn't see your last message
18:46.25jsdallashttp://pastebin.com/CGvBYqVY
18:46.29Sean-DerSo exec a bash script that then does the originate?
18:46.54[TK]D-FenderSystem(asterisk -rx "originate .....")
18:47.14[TK]D-Fenderjsdallas: Patterns have to start with a _ in front
18:47.21WIMPyjsdallas: You are missing the underscore.
18:47.40jsdallasah, I see now, knew it would be something simple.
18:48.45jsdallasThat seems to have done it, thank you.
18:53.01Sean-DerIf my extension is numeric does asterisk see it as an INT? Or is it still a string? I was wondering if I could do mathematical comparisons.
18:56.29[TK]D-FenderSean-Der: * doesn't have data "types"
18:56.49[TK]D-FenderSean-Der: expressions using math treat as numbers
18:57.25Sean-DerThats wonderfully nice :D
18:57.40[TK]D-FenderCraptastic!
18:59.33jim_ec2Sean-Der: most any language you can cast a numeric string to an int
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19:02.17[TK]D-Fender* doesn't have "types" so there is nothing to type-cast
19:02.38saxahello, anybody good with network wireing in here ? I have troubles when i connect my voip phone to the wall socket, its a grandstream gxp285 phone, and dirctly connected to the switch it works, so when i put my phone cable to the wall socket and connect this cable to the switch the phone does not get the ip and cant connect, any idea of how can I do some tests ?
19:03.01[TK]D-Fender* treats everything as a "string" and I suppose you could say it casts them as numeric (mix int/float) where math is required
19:03.14saxai know this is probably off topic, but if somebody can point me the direction, i would really apreciate it.
19:03.20Sean-DerYou have functions that do toFloat and toInt or whatever, but I was worried that asterisk ONLY saw things as strings and math would be impossible with CDR
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19:04.10Sean-Dersaxa: Do other devices receive an IP from the DHCP server?
19:04.24Sean-DerIt sounds like an issue at the physical layer
19:04.37[TK]D-Fendersaxa: "phone cable" ?  "wall socket"?  What is this "phone cable?  And wall socket?
19:04.53Sean-DerI think he means Ethernet ... RJ-45?
19:05.02Sean-DerOr ATA doubt it though
19:05.25WIMPyRJ-45 is a two wire data line.
19:08.48[TK]D-Fender"wall socket" doesn't say anything about what wiring is behind it, what the jack-type is, or where it leads
19:09.16WIMPyMight be Ehterkiller V2.
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19:21.25Sean-Derasterisk -rx "originate SIP/2071 extension 1234@911-playback" will hold up the dialplan
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19:22.01Sean-Der[TK]D-Fender, the System() call to a script with that in it sits and waits for it
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19:24.26Sean-DerI just && it stupid me
19:26.56saxaSean-Der: yes, dhcp works , it is a cable issue
19:27.18saxa[TK]D-Fender: phone cable = network cable 4 pairs
19:27.29saxawall socket is a network wall socket
19:27.31saxasorry
19:27.39*** join/#asterisk envisean (~envisean@unaffiliated/envisean)
19:27.46mchouI'm attempting to register to my sip provider, and apparently I can only use "register => user:passwd@provider/user", and not "register => user:passed@provider/mylocalexten"  What am I screwing up?
19:28.05saxabut it seem i inverted the colors
19:28.21saxabasically i turned updown the rj-45
19:28.55enviseanmchou: do you need to include your extension?
19:29.01enviseanmchou: just try it without the extension at the end
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19:29.46mchouthe whole point of including my local exten is it makes the dialplan a bit more readable....
19:29.54mchouenvisean: ^^
19:29.59S1lv3RCan i manage incomming calls from my UPC Cablecom Modem with Asterisk?
19:30.09mchouenvisean: I'd like to retain that format
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19:30.37enviseanmchou: it doesn't make sense though if you're connecting to a provider
19:30.46S1lv3Rore i need Special device?
19:30.56mchouenvisean: that's why I'm so puzzled why the part after / would even matter to registration upstream
19:30.57enviseanmchou: your extension will not hold on their system unless you're explicitly including your DID
19:31.04enviseanmchou: it doesn't
19:31.06[TK]D-FendermchouI'm attempting to register to my sip provider, and apparently I can only use "register => user:passwd@provider/user", and not "register => user:passed@provider/mylocalexten" What am I screwing up? <- you can put whatever you want after the /
19:31.07QwellS1lv3R: If it's analog, you'll need a device like an ATA.
19:31.14enviseanmchou: however, it matters if you're connecting different systems together or even clients
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19:31.40[TK]D-FenderQwell: 15th gen phtotcopy for add the D>A>D>A conversions :)
19:31.41S1lv3RQwell likes from Cisco?
19:31.47[TK]D-Fenderall*
19:32.25mchou[TK]D-Fender: that's what I thought, but apparently it's not true
19:32.25[TK]D-FenderS1lv3R: Your cable modem is effectively an ATA as it is, you wouldn't not plug another ATA into it, but rather an FXO device.
19:32.28Sean-Der[TK]D-Fender, my script if run from bash forks to the background 'asterisk -rx "originate SIP/2069 extension 1234@911-playback" &'
19:32.49Sean-DerBUT if I use System to run it Asterisk doesn't move onto the next item in my dialplan
19:32.52[TK]D-Fendermchou: It is.  However that thing you enter after the / is what they should dial back and you have to match in your dialplan just the same
19:33.24mchou[TK]D-Fender: I have match in my dialplan
19:33.32enviseanmchou: it's dependent on your sip provider and what format they want to see, it should be ok though for most providers
19:33.39mchou[TK]D-Fender: that's the point
19:33.43[TK]D-FenderSean-Der: I did not tell you to run a script and using RX is not a blocking process so unless you are doing something else that would tie it up then System should execute really fast and not effectively delay your call
19:33.43enviseanmchou: but not everyone conforms to the SIP RFC perfectly
19:33.58[TK]D-Fendermchou: Show us the call and your dialplan and the resgieration attempt
19:34.04QwellSean-Der: Do this.  Open a terminal, sleep 60s&, exit.
19:34.11QwellSean-Der: You'll see why there's a problem.
19:34.21S1lv3R[TK]D-Fender; my problem is i cant configure my UBEE Modem from Cablecom
19:34.31mchou[TK]D-Fender: right now you're just going to have to take my word for it
19:34.53mchou[TK]D-Fender: cause I'd have to scrub too much info
19:35.02enviseanmchou: you can grab a pcap or even use ngrep to see what's going to / from
19:35.14[TK]D-Fendermchou: Right now I have no reason to take your word for it.
19:35.16enviseanmchou: just add some filters for your userID or something so you can limit the scope of that trace
19:35.32mchou[TK]D-Fender: I assure you what I'm reporting is correct and not due to misconfiguration
19:35.43[TK]D-Fendermchou: password it and PM the link & PW to me if you like, but there's no wy I'm taking a claim like that at face value without backup....
19:35.57enviseanhahaha
19:36.09[TK]D-Fendermchou: And the more people protest they did everything right and refuse to show the less I believe them.
19:36.29mchou[TK]D-Fender: well, right now envisean thinks differently, so I have more respect for his opinion :)
19:36.37[TK]D-FenderWhen you're sure you did everything right is exactly the point where something slips past you
19:36.39*** mode/#asterisk [+o mjordan] by ChanServ
19:36.47Qwellmjordan: You can do it!
19:36.48S1lv3RAlso i need this strukture? UPC CC MODEM -> Cisco-SB-VOIP-SIP-ATA-Adapter -> Asterisk (Debian)
19:37.05mchouI didn't just start running asterisk yesterday
19:37.16enviseanmchou: I never said the format is wrong, I'm just saying it's not necessary with all sip providers :)
19:37.36mchouenvisean: yes, I hear you loud and clear
19:37.41enviseanmchou: we have tons of customers that don't include their DID or localext in their registration string w/ their * config
19:37.47[TK]D-FenderS1lv3R: No. ATA's by definition is an FXS device.  Your cable-model already has an FXS device built in.
19:37.58[TK]D-FenderS1lv3R: Linksys SPA-3102 would do the job.
19:38.17S1lv3Rokies thank you [TK]D-Fender
19:38.19[TK]D-FenderS1lv3R: But you are wasting conversions in this process and will lose call functionality, etc
19:38.27[TK]D-FenderS1lv3R: And quality
19:38.51[TK]D-FenderS1lv3R: Ask your provider if they will allow you to configure your own endpoint instead of the CableModem
19:39.34[TK]D-Fender14:37]enviseanmchou: we have tons of customers that don't include their DID or localext in their registration string w/ their * config <- yes, and when * does this, * sends "s" which looks even more like "shit" to your provider as a suggestion of what to dial back to you
19:39.49*** topic/#asterisk by mjordan -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0 (2011/12/15), 1.8.8.1 (2011/12/30), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org
19:40.05[TK]D-Fenderenvisean: register => user:pass@host/ifyoudon'ttellthemwhattodialherethenasteriskwillsay"s"foryou
19:40.14saxaok it worked now , many thanks, i just inverted upside down the RJ45.
19:40.25[TK]D-Fendermjordan: HAPPY PALINDOMRE VERSION DAY!
19:40.53[TK]D-Fenderpalindrome*
19:40.55[TK]D-Fendergah
19:41.00envisean[TK]D-Fender: sure, but we have put in place rules to handle that on our end
19:42.02[TK]D-Fenderenvisean: Any provider that actually cares beyond just accepting and using it or ignoring it and sending what they will is pretty retarded.  This would be the FIRST case of its kind I'd have ever heard of in my 7 years of being in here
19:43.16[TK]D-Fender'Joe says to dial "HAPPYHAPPYJOYJOY" when I send calls to him?  Okay....'
19:43.17envisean[TK]D-Fender: that's your opinion, and I thank you for your input, but we chose to move in a different direction as our platform is different from most of the other providers out there.
19:43.43[TK]D-Fenderenvisean: Yes, "special" people get a bus of their own too... it's just "little" :p
19:44.04mjordan[TK]D-Fender: I totally missed that.  Awesome.
19:44.09mjordanthose don't come around often :-)
19:45.19[TK]D-FenderTrying to abuse the "Contact:" header is a bastardization of the already beleagered SIP spec.
19:46.04envisean[TK]D-Fender: not too sure how this turned into a flame war, but I was simply stating the facts about how we allow our customers to interconnect. Our company can be small in your eyes, and you're free to hold on to your belief that we're "special", but our take is to make our platform as easy to connect to for any type of customer.
19:46.57[TK]D-Fenderenvisean: It wasn't your size.  its that you are twisting something that no other provider has any issue over.
19:47.33envisean[TK]D-Fender: I never said that we don't accept it either. I was simply stating that not all providers require it.
19:48.21[TK]D-Fenderenvisean: Somehow standards-compliance on this aspect makes the client conform to you, not the other way around.  it isn't as easy for them to connect to you.  Please note who is catering to whom with this practice.
19:48.48[TK]D-Fenderenvisean: None "require" it.  Your's seems to say if it isn't something specific we'll refuse you.
19:49.02[TK]D-Fenderenvisean: That is inappropriate.
19:49.04envisean[TK]D-Fender: I never said that
19:49.31[TK]D-Fender[14:27]mchouI'm attempting to register to my sip provider, and apparently I can only use "register => user:passwd@provider/user", and not "register => user:passed@provider/mylocalexten" What am I screwing up?
19:49.31envisean[TK]D-Fender: I said that we take it if it's there and we also don't require it. We do this so that it's easier for our customers to interconnect with us.
19:49.37*** part/#asterisk fireman_biff (~biff@65.48.133.103)
19:49.59envisean[TK]D-Fender: yes, and I offered a suggestion for him to try
19:50.09envisean[TK]D-Fender: he's not one of our customers, but I was simply giving him an option
19:50.16[TK]D-Fenderenvisean: "register => user:passed@provider/mylocalexten" <- if this is unacceptable, then hos is the "S" * sends if you put nothing acceptable?
19:50.59[TK]D-Fenderenvisean: sorry, the wording made it seem that you were his provider BTW.
19:51.16[TK]D-Fenderenvisean: Apologies if that was not the case.  But the points do stand on their own merit.
19:51.21envisean[TK]D-Fender: not at all, I'm just trying to help someone who isn't connecting w/ their provider by offering some solutions to try
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19:51.35[TK]D-Fenderenvisean: Well if his does this... then bollocks to them :p
19:51.48envisean[TK]D-Fender: absolutely, but you also thought that I was arguing against the /DID shouldn't be there, and I never said that
19:52.35envisean[TK]D-Fender: yeah, if he was one of our customers, one of our techs would be on the phone by now to assist and take a look through his/her configs
19:52.53[TK]D-Fenderenvisean: well the /whatever is jsut what gets put in the contact header and no provider should care about it's content.  It should either be respected qwhen they send you a call, or ignored.  It should never cause him an issue with a provider refusing to ACK the register or sending him calls.
19:53.10[TK]D-Fenderenvisean: Which was his claim.
19:53.15envisean[TK]D-Fender: yeah, and that's what we do.
19:53.57[TK]D-Fenderenvisean: So if I do "register john:pass@yourserver/somethingIwantHere" you'd refuse it?
19:54.02envisean[TK]D-Fender: no
19:54.07envisean[TK]D-Fender: we'd accept it, and it would register
19:54.11[TK]D-Fenderenvisean: as in reject the register?
19:54.25[TK]D-Fenderenvisean: Then good.  His was "wouldn't work".
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19:54.49[TK]D-Fenderenvisean: If you chose not to actually send that in the INVITE back to him... well so be it...
19:54.52envisean[TK]D-Fender: yes, and the confusion was that you thought we were his provider, so already there was a conflict in what I was saying vs what he was doing
19:55.19[TK]D-Fenderenvisean: Yeah... minor.  No excess harshness here... all good...
19:55.42envisean[TK]D-Fender: it's IRC, what else can you expect :P
19:56.46[TK]D-Fendermchou: Then again... coming in, asking about a problem and showing nothing... that usually demands a pretty quick swat upside the head :p
19:57.01S1lv3R[TK]D-Fender can i use Fritzbox 7141 as ATA ?
19:57.13[TK]D-Fenderenvisean: I expect a variable level of twitdom :)
19:57.21[TK]D-Fenderenvisean: And hope that it is set on "low"
19:57.53[TK]D-FenderS1lv3R: Does it say that you plug a PHONE into it?  Or does it say that you plug a LINE into it?
19:58.04[TK]D-FenderS1lv3R: You need one where you plug a LINE into.
19:58.28WIMPyS1lv3R: Depends on the firmware version, I guess.
19:59.30S1lv3R[TK]D-Fender i think i can modyfie 1 port with special software/firmware for Fritzbox
19:59.35[TK]D-FenderS1lv3R: Looking at the specs, no, it is an ATA with FXS interfaces
19:59.51[TK]D-FenderS1lv3R: Software won't change the circuitry on-board
20:00.17S1lv3Rahh okay
20:00.17[TK]D-FenderS1lv3R: It is an FXS device.  Nothing more to say.  Not usable
20:00.45WIMPy[TK]D-Fender: It is both.
20:01.04WIMPy1 FXO and 2 FXS
20:01.17S1lv3Rjop
20:01.44S1lv3Ri look likes DSL/TEL
20:01.52WIMPyThe FXO is probably paired with an S0.
20:01.58S1lv3Rand FON1 - FON2
20:02.35WIMPyYes.
20:02.36S1lv3RWIMPy u know Fritzbox works with DD-WRT?
20:02.52WIMPyAFAIK no
20:03.00S1lv3Rore i need an costum fw for Fritzbox
20:03.18WIMPyBut you can find all sorts of modifications for them.
20:03.32[TK]D-FenderWIMPy: Now is that a "fail-over-only" as most devices like that are, or is it an actual usable interface?
20:03.49WIMPyThat is ddWrt probaly runs, but doesn't support the telephony hardware.
20:03.59S1lv3Rahh okay
20:04.16WIMPy[TK]D-Fender: Depends on firmware version.
20:04.28S1lv3Rcan u give me any helpful links for Fritzbox Experimentals?
20:04.46[TK]D-FenderWIMPy: Firmware doesn't make a dumb-switch into an interface with actual DSP's behnind it <-
20:04.59WIMPyI asked google.
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20:05.18S1lv3Rthe question? (o;
20:05.44WIMPy[TK]D-Fender: It has four interfaces connected via software.
20:05.44S1lv3RFritzbox SIP Router
20:06.16QwellWIMPy: An FXO needs to be able to provide ring voltage.  There is hardware required for that.
20:06.42Qwelland an FXS needs to be prepared to handle such voltage.  Unless it is designed to do so, it will be fried.
20:06.59WIMPyWasn;t it the other way round?
20:07.24Qwellumm, yes
20:09.16WIMPyAll old Fritxboxen offer an FXO and an S0 to connect to the PSTN and 2 FXS for phones. Some have a 3rd FXS or a 2nd S0 in NT mode.
20:09.41QwellI didn't read backlog, I just saw something about firmware changing it
20:09.45WIMPyThe new models also include a DECT base and the latest versions don't offer PSTN connectivity any more.
20:11.47WIMPyThey also offer to use SIP phones.
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20:17.13mattsqzso i need to get our phone server recording all calls on all 10 workstations. im sure i can figure out how to enable this, however i am concerned about hardware.
20:18.45mattsqzthe machine in question is a pentium dual 2ghz E2180 (core2 based, not the older p4 crap but still with a tiny L2) and 512mb ram, with a single 7200rpm sata disk
20:19.25mattsqzcertainly a memory upgrade is in its future, but would the cpu and disk be sufficient for constant simultaneous recording of 10 lines?
20:19.43Qwellfor 10 calls?  easily
20:20.13[TK]D-Fendermattsqz: My watch can do this ... and it's ANALOG
20:21.29mattsqzmy thought was to have the recordings saved to a second physical disk, to avoid a reinstall of the server software and hopefully have no downtime
20:22.36mattsqzwell thats good to know
20:22.59mattsqzwe had an issue awhile back where it would spawn 20+ rasterisk processes and slow down quite a bit
20:23.44mattsqzso..simultaneously recording 10 lines 8 hours a day would be no problem then eh
20:23.52[TK]D-Fendermattsqz: I don't see how saving recordings to another disk helps you avoid a reinstall.  You reinstall if you choose to or have to.  And saving recodrdings to another disk has no impact on the uptime of your server
20:25.35mattsqzthe present disk hasnt enough room for recordings
20:26.03[TK]D-Fendermattsqz: Also, those aren't 8 solid hours of recording are they?  Or rather just a full work-day of multiple calls per channel?
20:26.25[TK]D-Fendermattsqz: At the end of your call, move the files to another server
20:26.30[TK]D-Fender(or drive/etc)
20:29.02mattsqzthe client insists these recordings be available for a certain amount of time. i thought 1tb raid-1 would be proper for 6months of calls. not 8 solid hours of recording, but nearly so. only a few seconds between calls.
20:29.58[TK]D-Fendermattsqz: No, solid means ONE recording.
20:30.09mattsqzok, no. many back to back recordings
20:30.12[TK]D-Fendermattsqz: If there are 50 calls with only 1 second between, that doesn't matter.
20:30.24[TK]D-Fendermattsqz: Not huge individual files, just many
20:30.34[TK]D-Fendermattsqz: So record, move, next.
20:30.51mattsqzi see
20:31.12mattsqzi saw suggested somewhere online that i create a ramdisk to record to then offload to disk after call
20:31.30WIMPyYes, make sure you don;t shove them all in to only one dir.
20:31.38[TK]D-Fendermattsqz: If you had more channels I certainly suggest that
20:31.49[TK]D-Fendermattsqz: Reduces actual disk use to 0
20:32.11WIMPyramdisk? How much RAM do you want to use? And creating bursts is usually not a good idea.
20:32.57WIMPy(although Linux has a certain tendency of doing it by itself)
20:33.14mattsqzram is cheap, 8 gigs wouldnt be unacceptable
20:33.54[TK]D-Fendermattsqz: 28.8 megabytes / hour @ G.711
20:34.04WIMPyYes, it's cheap, but I don't see any benefit there.
20:34.11[TK]D-Fendermattsqz: that is how much you need to float before committing to disk
20:35.11WIMPy28.8M if you mix both directions.
20:35.13[TK]D-Fendermattsqz: If you're taking the heavies codec and recording as-is
20:35.38[TK]D-Fender28.8 per direcetion.
20:35.52[TK]D-Fenderso technically lets say 57.6
20:36.00mattsqzok. so this is peanuts. the last call center i worked in had 600 seats.
20:36.04WIMPyyes
20:36.31[TK]D-Fendermattsqz: that is on the thoery us using pure-ram storage for immediate off-load and never hitting the local physical disk.
20:36.51[TK]D-Fenderon a worst-case size scenario
20:38.50mattsqzi think im still going to go with raided disks and more memory, once this is set up i dont want to have to touch it again except to add more extensions
20:39.24[TK]D-FenderRAID is always a good idea.
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20:42.48mattsqzalso this is a pbxtra cce server. they want mucho $$$ to enable call recording to their offsite servers. we dont need that, so im attempting to enable local recording.
20:43.30QwellPeople actually use pbxtra?
20:43.40mattsqzheh yeah
20:43.47ChannelZI use PB&J
20:44.13mattsqzwhen they set this place up, its what they (management) found that fit all their needs in a more or less turn-key package
20:45.37mattsqzits just bastardised (monetized) asterisk really.
20:46.52mattsqzim in the process of de-crapifying the place
20:47.20mattsqzexample: they were using windows home server boxes for file servers, along with microsoft groove
20:47.38mattsqzthey are now using simple embedded linux 2.6 NAS boxes
20:48.10mattsqzthey were using a clunky windows based cctv setup that barely worked, now its a slick zoneminder implimentation
20:49.11mattsqznow im slowly moving them away from pay-lots-for-not-much phone services
21:02.00*** join/#asterisk rayk_sland (~rklassen@mail2.communitascare.com)
21:04.15rayk_slandchan_dahdi seems to ignore all config files and set the channels to default context. I have removed every reference to any context=default and replaced it with context=from-pstn and dahdi show channels still shows my dahdi card as the default context
21:04.24rayk_slandhelp
21:04.52WIMPyDid you specify the parameters befor the channels?
21:05.08rayk_slandyes
21:05.43WIMPyShow us
21:11.25rayk_sland<PROTECTED>
21:12.00WIMPy~freepbx
21:12.00infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
21:12.28WIMPyAnd as you can clearly see, taht file doesn;t define anything.
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21:13.55mattsqzwhen i see a config on a website that looks like "exten => s,1,Answer()" for example, is that in asterisk.conf?
21:14.13rayk_slandno as you can clearly see it includes other files where the context directive used to be but I got so frustrated with having it ignored moved it just below the [channels] section in this file
21:14.21WIMPyextensions.conf
21:14.25bbourdagein the extconfig.conf, which family name would you use for the user table ?
21:14.26mattsqzthx
21:14.34rayk_slandto make sure it was actually read
21:15.38WIMPyrayk_sland: You have to edit the right file, but if you do, that might not go down too well with FreePBX, so better ask in their channel.
21:17.49rayk_slandI've asked there and done all sorts of backflips with the set up. Whatever I do my context= directive is ignored. (I watch the logs for chan_dahdi and it loads all the files and never complains Which file do you recommend.  recommend
21:21.22*** part/#asterisk rayk_sland (~rklassen@mail2.communitascare.com)
21:23.04mattsqzit appears this pbxtra is based on an older asterisk. what was features.conf previously called?
21:23.53woleiumIs there any way to get asterisk to show me which handsets are subscribed to mwi?
21:26.34woleiumor to display mwi status?
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21:27.35WIMPyI don't think there is anythign but 'voicemail show users'.
21:28.20*** part/#asterisk rhce7320 (~rhce7320@59.167.200.141)
21:35.38*** part/#asterisk asteriskmonkey1 (~darlek@74-51-38-204.telnetcommunications.com)
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22:55.40mattsqzwell, it looks like enabling incoming call recording is going to be easier than i thought
22:56.22[TK]D-Fender"enabling" is a rather glorifying term.  It's 1 line of dialplan....
22:56.30mattsqzit appears the hardest part will be whipping up an sh script to organize the recordings into a usable dir tree
22:56.55mattsqzhah if you saw what fonality quoted for that 1 line..
22:57.49mattsqzit was in the thousands of dollars
22:58.12mattsqzi think im probably in the wrong business
22:58.34[TK]D-FenderUse OSS and save lots of money ... you can blindly throw at consulting because you don't want to learn the basics....
23:00.16mattsqzwhat ive really liked lately is the maturity and stability of alot of OSS solutions.
23:01.08mattsqzive had alot of fun with embedded systems this year. routers, cameras, NAS, phones, tablets
23:01.38mattsqzall of it simple and foolproof compared to available proprietary offerings
23:01.56mattsqzwe havent had any of the NAS' i set up go down even once
23:02.15mattsqzbut..its time to start a 4 day weekend.
23:02.18mattsqzis outta here
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23:20.56carrarenables the enabler
23:25.05S1lv3Ri will remove asterisk from my debian system. The Error msg http://paste.debian.net/150594/
23:25.19S1lv3Raviable other way to remove the package?
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