IRC log for #asterisk on 20070921

00:02.19*** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au)
00:04.04*** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt)
00:07.53*** join/#asterisk loompek (n=NoName@noname.rula.net)
00:07.55loompekmorning
00:09.03loompeki've got a little ol' question.. how come asterisk doesn't register at some other sip server even though i have all the necesary stuff in sip.conf (register => ... and [server]...)
00:10.13loompeki'm looking the tcpdump output for port 5060... asterisk just sends OPTIONS to all of the serververs in sip.conf with register command and qualify !=no
00:11.25JT~pb
00:11.26jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
00:11.33JTpastebin sip.conf minus passwords
00:12.51*** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br)
00:13.32litageis it possible to determine whether ``fxotune -s'' actually worked, or what changes it made?
00:13.36*** join/#asterisk PepOSX (n=pepOSX@190.72.153.233)
00:16.32bungalowhi: im trying to get asterisk to reinvite after dial, to bridge two sip channels... but the media stream is continuing to go through asterisk...
00:16.39bungalowany idea on how to debug this
00:16.40bungalow?
00:18.03fujin_are both sides of the conversation set up with canreinvite=yes?
00:18.54*** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net)
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00:21.43bungalowfujin_: yes
00:22.57fujin_is there any reason for the audio to still be routing through asterisk? meetme.. monitor?
00:23.41bungalowI was running mixmonitor before the dial, but I've turned that off... so now nothing... the call comes in, processed by agi, and then dial (w/ g option)
00:24.15bungalowwith sip debug I see some INVITES, but not sure what I should be looking for exactly
00:27.01loompekJT hope i didn't miss anything
00:27.01loompekhttp://rula.net/121
00:27.37ManxPowerbungalow: HOW do you know the media stream is still going thru Asterisk??
00:28.22bungalowManxPower: I see rtp traffic going to and from asterisk with rtp debug... is this a valid way of checking?
00:29.22bungalowManxPower: also, earlier when I had mixmonitor on before the dial I also was recording audio (not sure if this causes asterisk to not re-invite, though)
00:29.52ManxPoweryes, but rtp debug is a new feature and not everyone knows about it.
00:29.59loompekJT any ideas?
00:30.11ManxPowerbungalow: ANYTHING that causes asterisk to listen to the audio will make it not reinvite.
00:30.27ManxPowerbungalow: I assume you are forcing both legs of the call to use the same codec?
00:30.47fujin_that'll do it
00:30.49bungalowManxPower: yes, both ulaw... but canreinvite=yes.
00:31.00bungalowManxPower: rtp debug on is included in that ANYTHING?
00:31.14*** join/#asterisk jsidhu2 (n=atomik@66.206.163.184)
00:31.23[TK]D-Fenderbungalow, No.
00:31.59[TK]D-Fenderbungalow, any of these Dial options : "tTwWr", Use of "Monitor", etc.
00:32.17jsidhu2aight, i need some help. I have a sip trunk from voipvoip, i can make outgoing calls, but inbound route isnt working.. i create a new any DID/any CID route to goto an IVR, but it doesnt do anything, just hangs up.. anyone help?
00:32.17[TK]D-Fenderbungalow, Basically antyhing you tell * to do that requires it to snoop in.
00:32.51[TK]D-Fenderjsidhu2, enable SIP debug.  Place another call to test.  PASTEBIN the *entire* attempt.
00:32.52[TK]D-Fender~pb
00:32.52jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
00:32.53[TK]D-Fender^^^^^^^^^^^^^^^^^^^^^^^^^
00:33.02bungalow[TK]D-Fender:  I should be ok then.... the only dial option I have set is g
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00:33.29[TK]D-Fenderbungalow, that is non-invasive.  Another thing that precludes most reinvites : NAT <-
00:34.29bungalowI have nat=no
00:35.34loompekJT didya check the pastebin?
00:35.48bungalowasterisk is reporting 'native bridging <channelid1> with <channelid2>'
00:35.56bungalowis that something I should be seeing?
00:36.16[TK]D-Fenderbungalow, that can be fine
00:37.05z001hi;  I'm looking to make a Queue that will ring phones (A+B+C) with 'ringall', and at the same time ring mobiles (M+N) with 'rrmemory', so the first ring will be (A,B,C,M), the second (A,B,C,N)...
00:37.08[TK]D-Fenderbungalow, a native bridge is where TRIES to let the channels connect directly.  It does not imply that it actually succeeding though.  I presum that'd show up on RTP debug
00:37.09z001This doesn't seem possible with a single queue... Can I put a call in multiple queues, or put some kind of combined/alias extension in a queue?
00:37.50jsidhu2[TK]D-Fender: http://pastebin.com/d77ae3d7f
00:37.51[TK]D-Fenderz001, You can do that.
00:38.47[TK]D-Fenderz001, What you need to do for "M+N" though is dial a LOCAL CHANNEL in which you'll use a "toggle" stored value to select which to ring.  Check which, change the flag, dial the guy.
00:38.47loompekmaybe anybody else?
00:38.48bungalow[Tk]D-Fender: so is it an indication that it's trying the reinvite?  what sort of INVITE sequence should I expect?  I assume once the call is connected it performs a series of SIP invites and the RTP stream would vanish from the server...
00:39.17[TK]D-Fenderbungalow, I can't tell you any more on the final detail unfortunately.  That is the limit of my experience
00:39.42[TK]D-Fenderbungalow, I presume you could do a port dump against those IP's to config RTP is going direct
00:39.55[TK]D-Fenderconfirm*
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00:40.41bungalow[TK]D-Fender:  rtp debug should do the same, no?
00:40.47bungalowshow the same...
00:41.02[TK]D-Fenderbungalow, Possibly, I've never actually used it myself, though I have heard of it
00:41.10z001[TK]D-Fender: so my queue members will be SIP/A, SIP/B, SIP/C and {something}/123   where 123 is an extension in the dialplan with a global variable and some toggling logic?
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00:41.45[TK]D-Fenderz001, "Local/123@context/n" (yes, keep the /n)
00:42.06[TK]D-Fenderz001, Now there is a CATCH, and a big one....
00:42.13bungalow[TK]D-Fender: ok thanks
00:42.17bungalowManxPower: still there?
00:42.20[TK]D-Fenderz001, What kind of interface are you going to call those Cell's on?
00:42.27z001[TK]D-Fender: thanks - I was looking for an example like that in the queues.conf.sample and the voip-info wiki, but they only cover ZAP/SIP and Agent lines.
00:42.42z001[TK]D-Fender: via an outgoing SIP trunk
00:43.00[TK]D-Fenderz001, remember those samples were showing you how to use a give channel, and LOCAL is a channel type too...
00:43.31[TK]D-Fenderz001, Ok, be warned that if your ITSP considers the call "answered" the moment you PLACE it your entire plan goes out the window
00:44.02[TK]D-Fenderz001, and the other thing : If a cell has voicemail and is busy, etc, VM could pick up iNSTANTLY thus answering the call.  Not a good thing.
00:44.23z001ah, it might just connect the caller to the first mobile while it's ringing and that's it?    That's quite a catch
00:44.25[TK]D-Fenderz001, I typically advise against using Cell's as queue members
00:44.34[TK]D-Fenderz001, it is indeed.
00:44.49z001but what alternative is there?
00:44.58z001handling it entirely in dialplan logic?
00:46.41jsidhu2aight, i need some help. I have a sip trunk from voipvoip, i can make outgoing calls, but inbound route isnt working.. i create a new any DID/any CID route to goto an IVR, but it doesnt do anything, just hangs up.. anyone help?  SIP DEBUG: http://pastebin.com/d4f6d2b78  (The inbound route is set to forward all calls to a queue with an announcement)..
00:47.15litagewhen calling using an FXO port, i hear echo but the other person doesn't. when i hook up an analog phone to the same POTS line, there's a little bit of echo. this leads me to believe that there's echo being generated somewhere in the building's wiring, or between the building and the exchange. what would you recommend to reduce echo?
00:48.01[TK]D-Fenderz001, No, the catch is the "answering" possibilities that come up in trying to call a cell.  This could happen automatically because of your ITSP, or becase the cell is out of range and VM's, etc.
00:48.31[TK]D-Fenderz001, to this means you had better have control and confidence in how you call that you know you will actually get to cyctle around.
00:48.59*** part/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net)
00:50.17*** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net)
00:50.42CrazyTuxHey guys, I'm working on voicemail, how much configuration can I do as far as email templates go?  Can I pipe information from external source for example into the emails that are sent out as well?
00:50.50z001[TK]D-Fender: ah. Well, this is for a night service option - a ring-all queue, but after hours ring-all and some mobiles.   So there isn't really an option to not call mobiles.   I'll have to experiment, I guess.
00:51.03z001[TK]D-Fender: Thanks for the help
00:51.44[TK]D-Fenderjsidhu2, The call is being answered and the dialplan is processing I see it calling apps and reaching the end of that context.  This is FreePBX and you are at the end of the help you should expect in this channel.  FreePBX is *not* supported here.
00:52.26[TK]D-Fenderz001, What I MIGHT suggest for this : Queue the inter ones and only let them rotae around a few times, then maybe TIMEOUT to *dump* them to the cells....
00:53.10[TK]D-Fenderz001, All jsut ideas.  As long as you understand the cirmstances you are dealing with you'll know which was best servers your employees and your callers
00:56.58z001I'll get them to run VPNs and SIP clients over GPRS on their phones and pretend they're internal then.  ;-)
00:57.08z001ugh
01:00.40[TK]D-Fenderz001, Go test how your ITSP reacts, etc, and make sure not to ring them too long so that they fall to VM.  I did a bit of this just for basic OOO forwarding here
01:01.05*** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net)
01:02.26z001it's a bit late now (I can't IRC from work), but I will tomorrow
01:04.39JerJerz001:  setup a proxy at home  :D
01:05.38z001oh technically I could, but politically I can't.
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01:05.42fujin_ssh tunnel
01:06.47z001on another note, if I want to query an SQL database for caller ID, that's going to have to be an AGI script, isn't it?
01:07.01Qwellz001: func_odbc
01:07.21Qwellfunc_odbc is pretty much one of the most awesome dialplan functions ever
01:07.23z001ah... I have caller ID going to an odbc database, and was wondering if I could piggy-back on that
01:07.42z001I thought I'd seen odbc related functions somewhere, but haven't spotted it since wanting to find it
01:08.02rob0fujin_: an ssh tunnel can't do VoIP, because it's TCP, and voice protocols are UDP.
01:08.25fujin_that's a pretty stupid assumption
01:08.29fujin_but yeah, sure, if you say
01:09.14rob0Stupid assumption?
01:09.28rob0Maybe you can explain it to me then.
01:09.38fujin_my comment was referring to him not being able to not IRC from work
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01:10.06rob0oh ok :)
01:10.08fujin_you do realise you can tunnel both udp and tcp over a SSH tunnel
01:10.16fujin_using netcat to redirect the udp
01:10.20rob0no I did not / do not
01:10.20fujin_I've done it, works fine.
01:10.22litagewhen changing echo cancelers in /usr/src/zaptel/zconfig.h , is it necessary to reboot the box, or just unload then load all zaptel-related modules?
01:10.26rob0hmmm
01:10.33Qwelllitage: just unload the modules
01:10.38fujin_perhaps even later revisions of openssh allow udp forwarding through ssh
01:10.44rob0I'd still think openvpn would be better and easier.
01:10.56z001pingtunnel is one of the neater tunnels I've heard of
01:11.17fujin_dnstunnel is awesome too :)
01:11.24fujin_slow, though;
01:11.43fujin_rob0: openvpn is an abstraction layer which is usually a pain in the ass, I'd generally go with PPTP vs. openvpn.
01:11.55z001ha, nice
01:11.56rob0yikes, not me. BTDT.
01:11.56litageQwell: thanks
01:12.20rob0<== retired pptpd admin
01:12.21Qwelllitage: while you're playing with echo cans, try the jpah "echo can"
01:12.24Qwellit's pretty funny :D
01:12.35litageQwell: yeah, i've heard it's a bit strange
01:12.45Qwell(I wrote it because I actually needed something like that to test something)
01:13.15Qwellit drops 2 out of every 3 frames of audio, heh
01:13.22litageafter reloading the zap-related modules, dmesg says "Zaptel Echo Canceller: MG2". however, MG2 is commented in zconfig.h , and KB1 is uncommented
01:13.26Qwellit sounds *horrible*
01:13.37Qwelllitage: is this zaptel 1.4.5?
01:13.39Qwellif so, use 1.4.5.1
01:13.47litageQwell: this is zap 1.2.20.1
01:14.00Qwelllemme check something...
01:16.07JTfujin_: can you get ssh to run over udp?
01:16.21RyushinOkay, I'm at a bit of a loss here.  Do polycom phones by default register themselves with the server after a reboot, or only when they make their first call?
01:16.29litageJT: not that i know of..
01:16.39Qwellgrr
01:17.10litageQwell: ?
01:17.36JTlitage: i didn't think so either, i wonder if fujin_ knows otherwise
01:18.20Qwelllitage: get svn branch 1.2
01:18.34QwellI thought those changes got released...  they never did though
01:18.53litageQwell: which revision?
01:18.57Qwelllatest
01:19.50litageQwell: what are the differences between MG2 and KB1?
01:19.55Qwellno idea
01:20.14Qwellwhat hardware do you have?
01:20.49litageQwell: TDM400P
01:20.52Qwellhow old?
01:20.59litageQwell: couple of weeks
01:21.08Qwellcall Digium sales tomorrow, tell them you want some HPEC licenses. :)
01:21.23Qwellsay you bought a card a few weeks ago, and that MG2 isn't cutting it
01:21.32litageQwell: how much are HPEC licences?
01:21.47QwellFor Digium customers with certain cards, $0 :)
01:21.58Qwellotherwise it's $10 per port I think
01:22.23Qwellit's *far* better than any of the open source ones in zaptel
01:22.39bungalowhi...trying to do an external rtp bridge via sip reinvite -- I see that asterisk attempts the re-invite via notice in the X-asterisk-info header, and it appears the INVITE attempt is properly acked, but the RTP stream continues to go through my server.  Any idea how to debug this?
01:23.07Qwellbut call them up tomorrow, they'll hook you up
01:23.28litageQwell: should i get one HPEC license for each FXO port?
01:23.28QwellI'm not sure what you'll need - maybe just the cards serial number
01:23.32Qwellyep
01:23.46Qwellshouldn't cost you anything
01:23.56litageQwell: so should i bother with the zaptel svn trunk?
01:24.04Qwellyou still should, yes
01:24.11Qwellthe same bug is going to affect HPEC
01:24.43litageah gotcha
01:25.27Qwellsvn branch 1.2 that is, not trunk
01:25.57litageQwell: ie?:   svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2
01:26.00Qwellyep
01:26.05litageawesome possum
01:26.52rob0hmmm, this definitely sounds like an Alabammy channel. :)
01:27.03Qwellrob0: howso? O.o
01:27.14rob001:26 < litage> awesome possum
01:27.20litageQwell: btw, if i reduce the rxgain in zapata.conf to -20.0 , most echo is gone, but i can barely hear the other person. i'm assuming the echo mostly disppears simply because echo's produced by a combo of delay and loudness, and the loudness makes the echo inaudible, but doesn't eliminate the delay?
01:27.32litagerob0: hah, i'm in .au . nowhere near .us
01:27.44Qwelllitage: yeah, if the audio isn't received on the far end, it can't echo back ;)
01:27.47rob0Aha! Even MORE marsupials down under!
01:28.06litagerob0: yup, we've got a few. we even have the world's only monotremes!
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01:34.56litageQwell: this zaptel-1.2 branch that i'm exporting...is it considered 1.2.20.1 r3055, or just r3055, or ...?
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01:49.50litageQwell: alive?
01:52.08russellbi killed him
01:52.10russellbsorry
01:53.34[TK]D-Fenderrussellb, Don't just sit there!  Go clean up the mess!
01:54.02russellbi pay people to handle that part for me.
01:54.17[TK]D-Fenderrussellb, outsourcing is entirely acceptable
01:54.26russellbthanks for the approval :)
01:55.29litagehahah
01:55.37litagecya guys! thanks for your help, Qwell
01:56.11RyushinI'm trying to work on some polycom phones remotely.  430's and a 601.  When I reboot them, they don't seem to register themselves with the sip server automatically after they reboot.  Am I missing something?
01:56.38RyushinIf they initiate a call, then they register themselves, but not before then.
02:00.19[TK]D-FenderRyushin, without seeing your configs we can't know
02:01.18RyushinWhich configs do you need to see?  The sip.cfg file?
02:04.17[TK]D-FenderRyushin, Everything applicable, plus logs, and CLI SIP debug on reboot
02:07.36Ryushinhttp://www.pastebin.ca/705368  This is the app log from the phone.
02:08.45Ryushinphone boot log: http://www.pastebin.ca/705371
02:11.04[TK]D-Fendererver 'voipdenver.hq.xpulseusa.com' said 'x7072/0004f214dc21-phone.cfg' is not present
02:11.17[TK]D-FenderuBLFCompressed: File /ffs0/local/0004f214dc21-phone_cfg.zzz does not exist or is empty
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02:12.19RyushinWell, these phones were in another state and the vpn was down between the server for a couple of days.
02:12.35RyushinI'll kick that phone again and look at the log to see if it does it again.
02:17.49RyushinTo reboot the phone I'm logging into the web server on the phone, going to sip, choosing the last setting and without changing anything, clicking submit to reboot the phone.  The the phone truly reboot using this method?
02:20.21[TK]D-Fendershould
02:21.31jarrodhmm, has anyone used the asterisk appliance?
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02:22.19RyushinOh well, I'm burnt out on this.  I'll pick it up in the morning.  This just bugs the crap out of me since this stuff was working fine before.
02:23.08RyushinThanks [TK]D-Fender.  You awesome to have around here.
02:23.18Ryushinbe back in 10 hours.  :(
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03:18.47yidiyuehanhi, any one can tell me the website that I can report a IAX channel bug?
03:20.58[TK]D-Fenderyidiyuehan, on the Bug Tracker
03:21.43yidiyuehanhi, D-Fender,where is it
03:21.56[TK]D-Fenderyidiyuehan, its linked from www.asterisk.org
03:22.08[TK]D-Fenderunder support I believe
03:22.09yidiyuehanok. thanks man.
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03:25.34Math`is it possible than OPT_CALLER_HANGUP doesnt work if the call is packet-to-packet bridged?
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03:43.59perdare there any good web interfaces for voicemail out there?
03:45.48perdis ari considered 'the best' i guess?
03:49.13litage|wperd: "good", "best" etc are all relative to your needs
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03:50.59perdwell, i am a simpleton and i like features and flashy looks
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03:57.58[TK]D-Fenderperd, Go by an Avaya
03:58.02[TK]D-Fenderbuy*
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03:59.27markgreeneDoes anyone have any advice to offer for this situation: When I want to record a call I go into the asterisk CLI and type "mixmonitor start SIP/EXTEN filename.wav" and it records the call just fine. Once the call is hung up or I manually stop the recording asterisk restarts, disconnecting any current calls. I am running asterisk 1.2
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04:07.17Math`I think its worth mentioning in app_dial's documentation that the 'H' option will use the Disconnect feature code once the call is established
04:13.10russellbMath`: yeah, there is a bug report open for exactly that ... to clarify when features become available
04:20.46Math`oh, I'll look it up
04:20.53Math`and add a note to the wiki
04:23.48tzangerweird
04:24.00tzangerzttest reports 100% consistently
04:24.08tzangerbut coppice's sliptest program is all over teh map
04:34.38russellbtzanger: have you been writing a zaptel driver lately or something?
04:34.38tzangerrussellb: I've got a 288 channel zaptel driver to a proprietary pbx
04:34.38russellbinteresting
04:34.38tzangerit works fine so long as I'm taking timing from the PBX
04:34.38tzangersince teh PBX won't let me slave it, I have the DSP implementing elastic buffers and decoupling the TDM and TDMoE sides
04:34.38tzangerthink one PC with 4 288-span TDMoE connections to 4 different PBXes
04:34.38tzangerbut something is fucking my timing right in the ear
04:34.38luke-jrbtw, why doesn't zaptel use the Linux telephony interface?
04:34.38tzangerthe elastic buffers seem to be working just fine
04:34.38russellblol..
04:34.42luke-jrwtf
04:34.42tzangerztdummy was teh first suspect, even though the elastic buffer is sized for 5 frames
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04:34.46tzangerbut periodically I get a flurry of underrruns/overruns, depending on which eleastic buffer you're looking at
04:34.46tzangerthere's a tdm400p in there right now doing nothing but timing
04:35.01tzangerzttest shows perfect consistent results
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04:35.23tzangerbut sliptest (sends awgn out an unconnected zaptel port and listens for the echo, trying to correlate it to determine loop length)
04:35.30tzangerbut sliptest is ALL OVER the map
04:35.37tzangersliptest on my home machine's tdm400p is consistent
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04:35.53tzangerso at the moment I don't know who to trust :-)
04:37.30rob0You can trust your car to the man who wears the Star.
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05:09.05JTluke-jr: what are you wtfing at?
05:11.48luke-jrJT: the netsplit ☺
05:12.11JTyeah, they happen.
05:12.36luke-jr☺
05:17.14TJNIITo include another file into extensions.com I use #include [filename] correct?
05:17.31TJNIIs/extensions.com/extensions.conf/
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05:19.11ManxPowerTJNII: correct
05:19.56TJNIIManxPower: ty
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05:21.28ManxPowerTo be safe, use the fulll path and don't use quotes unless you understand what you are doing.
05:21.47TJNIIRoger
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05:25.46luke-jrso is Voxee dead?
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05:37.27TJNIIWith the background command, if I want it to look for file xyz in subdirectory abc of the sounds directory should I do Background(/var.../sounds/abc/xyz) or is this done another way?
05:40.13TJNIIsurvey says Background(abc/xyz) works.  Is that the correct way, though?
05:41.00QwellTJNII: yes
05:41.05TJNIICool.  Thanks
05:41.07Qwellif the files are in /var/lib/asterisk/sounds/
05:41.15Qwellif not, you'll need to use full path
05:41.27QwellI wonder if backwards relative paths would work...
05:41.32TJNIIRight.  Just wanted to make sure what was working for me was actually supposed to work.
05:41.39Qwellmv myprompt.gsm /var/lib/asterisk/blah
05:41.46QwellBackground(../blah/myprompt)
05:41.49QwellI should try that some time
05:41.58luke-jrmkdir /var/lib/asterisk/blah first
05:42.05luke-jr☺
05:42.06Qwellyeah, yeah
05:42.18Qwellsomebody test that for me :p
05:42.29luke-jrso any good VoIP providers yet? :P
05:42.34Qwell~itsp
05:42.34jbotAn ITSP (Internet Telephony Service Provider.) is a "VoIP Phone Company". Example : Vonage, Broadvoice, Teliax, VoicePulse, etc. "All ITSPs suck. Some suck less than others. Teliax seems to suck less than most.." (tm) (c) 2007 ManxPower
05:42.41Qwellsurvey says - nope
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05:42.45luke-jrheh
05:43.05luke-jrVoipjet should do origination
05:43.07luke-jrthey seem stable
05:43.24kiscokidVoicepulse seems fine
05:43.29luke-jrVoicepulse?
05:43.47luke-jrVoxee seemed fine when I signed up
05:44.20kiscokidhttp://connect.voicepulse.com/
05:44.33luke-jrerr
05:44.39luke-jrVoicePulse looks like a ripoff
05:44.41luke-jr☺
05:44.52kiscokidwhy is that?
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05:46.10luke-jr$15 for the cheapest thing
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05:49.59kiscokidthat's the consumer side.  Look at the asterisk page
05:50.19luke-jraha
05:50.37luke-jrpfft, $50 minimum deposit
05:51.12kiscokidwhat's wrong with that?
05:51.45luke-jrit's not a trivial amount to bet on an unproven service
05:51.55kiscokidits proven to me
05:52.01luke-jrI've seen too many ITSPs die to risk that
05:52.52luke-jrlet alone having to sign up just to see their full rate table
05:54.00kiscokidwhich itsp do you like, if any?
05:55.51luke-jrVoipjet works great for outbound calls
05:56.11luke-jrMyPhoneCompany has pretty much always worked, but their site is Flash so I can't really do anything there if I had to
05:56.45luke-jriConnectHere *usually* works, but has total crap for support personel
05:57.08luke-jrVoiceStick usually works and actually added a "feature" when asked by enough people
05:57.16kiscokidI tried Voipjet a few months ago.  It didn't work
05:57.26luke-jrSellVoip was nice, but unreliable and absolutely no support
05:57.30luke-jrdidn't work? O.o
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06:22.28loompeki've got a little ol' question.. how come asterisk doesn't register at some other sip server even though i have all the necesary stuff in sip.conf (register => ... and [server]...)
06:22.51loompeki was checking the tcpdump output for port 5060... asterisk just sends OPTIONS to all of the serververs in sip.conf with register command and qualify !=no
06:22.56loompekhere's my sip.conf
06:22.56loompekhttp://rula.net/121
06:24.42loompekeven though 'sip show peers shows status ok for all
06:29.45zeeeshinstalling asterisk-addons-1.4.2 getting error, This content is stored as http://sial.org/pbot/27618.?
06:30.51loompekany ideas?
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06:37.37tengulrehi,all
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06:37.59tengulrewhat's different between asterisk-1.4 and asterisk-1.2?
06:39.05JTwhatever it says in UPGRADE.txt
06:39.26loompektengulre 1.4 has more features but it's prolly not so stable because the code is quite fresh
06:39.59loompekJT awsome.. you're here! did you happen to spend some time with my sip.conf? :)
06:40.50hmmhesaysin mother russia your sip.conf spends time with you
06:40.52JTsorry i was pretty busy
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06:44.25loompekJT did you even look at the conf file? i don't think there are any syntax errors and i guess a trained eye (like yours) should probably find the 'gremlin' in a matter of minutes :) here's my sip.conf again... any help would be appreciated - http://rula.net/121
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06:45.58r00tlzhi
06:47.22loompekmorning to you too
06:50.56luke-jrloompek: posting the error might be a good idea
06:51.52loompekluke-jr asterisk doesn't REGISTER but only sends OPTIONS
06:52.08loompekso i can call through a peer
06:52.13loompekbut i can't receive a call
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07:00.16kaldemarloompek: have you checked that with sip set debug?
07:01.10loompekkaldemar don't know how to use sip set debug
07:01.23loompeki've enabled it just right now in console
07:01.27loompekwhat next?
07:02.30kaldemarit prints all the SIP traffic that asterisk sends or receives. look for the REGISTER messages.
07:02.44kaldemarthat's a way to confirm that asterisk tries to register.
07:02.58loompekthere aren't any REGISTER
07:03.02loompekthat's the whole point!
07:06.57luke-jrloompek: why do you have 2 [general] headers?
07:07.38loompekluke-jr just because.. is it wrong?
07:08.59luke-jrcould be
07:10.06loompekno it isn't
07:10.23loompekdeleted the second [general] and still no luck
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07:11.25loompeki can see only OPTIONS in sip set debug... and in tcpdump
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07:13.39loompeki mean.. combinations of OPTIONS and 404 Not found (on the remote asterisk) and OPTIONS and 200 Ok (on all the other)
07:14.47loompekany mode ideas?
07:15.11kaldemarloompek: did you remove the first or the second [general]? what did you do after that? sip reload, restarted asterisk?
07:15.44kaldemarumm. i assume you deleted the second. :P
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07:17.40loompekrestart asterisk
07:17.57loompekof course i deleted the one in between, not the one in the beginning :p
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07:21.06kaldemarloompek: comment out registerattempts=10 and try to debug it again.
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07:23.07loompekokay.. sip reload... still only options
07:24.49loompeklike i said.. asterisk sends ONLY options request... there is no REGISTER anywhere (except from the internal clients)
07:24.55loompekthe phones 1,2,3,4
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07:25.46HarisHello people
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08:04.45awkhrm, anyone advise on these issues?
08:05.24awk~pbb
08:05.26awk~pb
08:05.27jbotA Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
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08:06.08awkhttp://paste.debian.net/37653
08:06.15awkI can't see what has changed since 1.4 zaptel
08:06.27awkbut it doesn't like these things for some reason, i'm pasting zapata.conf now
08:07.06awkhttp://paste.debian.net/37654
08:07.08awkthat is my zapata
08:07.31awkany sugestions would be great, i'm getting major issues with dtmf now with this upgrade and it is not due to echo training
08:08.02awki'm sure if I work this zapata issue out i will have resolved it, I cant find naything on voip-info telling me what I have is wrong, nor does google have replys to these warnings.
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08:10.27AstNewbieHi everyone, I have a TE120P T1 card connected to my Asterisk server. It works flawlessly since it started to provide service.
08:10.36defsworkbut ?
08:11.51AstNewbieHowever, with the latest version, 1.4.11, our users will hear some noise occasionally when dialing out ...
08:12.20*** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com)
08:12.26Uatechi there
08:12.29Uateci'm getting Sep 21 09:12:01 WARNING[13087]: file.c:229 ast_writestream: Natural write failed
08:12.30UatecSep 21 09:12:01 WARNING[13087]: format_sln.c:166 slinear_write: Bad write (256/320): File too large
08:12.33Uatecevery single second
08:12.36Uatecwhat the hell does it mean?
08:12.42Uatecwhat file is it failing to write to?
08:12.46AstNewbieMore specifically, the problem exists since
08:12.58AstNewbiearound 1.4.9 ...
08:14.16AstNewbieIt seems the problem relating to the Zap channels .... cos we dont hear any noise when calling internal extension (all are SIP channels)
08:16.34Strom_Cwhat kind of noise?
08:17.08AstNewbieSili Sala .....
08:17.27UatecAsterisk is writing about 5 log files a second
08:17.32Uatecunder the name event_log.XXX
08:17.32Strom_Ci don't know what a "sili sala" noise is
08:17.36Uatecand it's always empty
08:17.52Uatecthere are 6000 files in this directory and counting
08:17.59Uatecand they're all empty
08:18.10Uatecthe total size of the directory is like 500k
08:18.41luke-jrlol
08:19.22Uatecit's not funny because after a while asterisk slows right down
08:19.24Uatecthis is not right
08:19.36Uateclet me show you some of the stuff from my cli
08:19.50luke-jrout of disk space?
08:20.17Uatechttp://rafb.net/p/sUYo6D98.html
08:20.19Uatecno
08:20.19Uatecit's not
08:20.33Uatecwhy is it creating all these files?
08:20.55AstNewbieHmmm ... the noise is squelched ...
08:20.57luke-jrread the message
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08:21.01Uateci am reading the message
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08:21.03luke-jryour files are too large ☺
08:21.07Uatecbut it doesn't tell me which file
08:21.16Uateci've moved all my log files
08:21.17luke-jrthe event_logs I'd presume
08:21.19Uateceffectively deleted them
08:21.28luke-jrit seems to think you have a 0 byte limit
08:21.36luke-jrwhich could be disk space restrictions
08:21.38luke-jror quota
08:21.46luke-jror perhaps a log size limit
08:21.49luke-jrmaybe a ulimit
08:22.27Uateci emptied the logs
08:22.32Uatecbut the error continued
08:22.42luke-jrcheck the other possibilities I mentioned
08:23.18Uatecand i've got 46 gig of free space
08:23.24Strom_MUatec: it's the slin format driver, which means it's trying to write a sound file
08:24.00Uatecand ulimit is unlimited
08:24.12Uatecahhhhh
08:24.21UatecWTF?
08:24.40Uatecthere is a 2 gig wav file in /var/spool/asterisk/monitor/
08:24.45luke-jrLOL
08:24.52Uatecwhy the hell?
08:25.14Strom_Mare you recording any calls?
08:26.15Uatecyes
08:26.17Uatecall of them
08:26.22Uatecbut no 2.5 gig long calls
08:26.26Uatecor supposedly not
08:27.16Strom_Mperhaps you have a call up that never cleared properly
08:32.39luke-jrit*
08:33.48AstNewbieHi everyone, I have a TE120P T1 card connected to my Asterisk server. It works flawlessly since it started to provide service.
08:33.49AstNewbieHowever, since the last few version, 1.4.9 - 1.4.11, our users heard some buzz noise occasionally when dialing out while the called parties didn't ...
08:33.51AstNewbieIt seems the problem is relating to the Zap channels .... cos we dont hear any noise when calling internal extension which are SIP channels only.
08:37.43AstNewbieThe underlying OS is Debian.
08:37.44AstNewbieAny settings would be related to such situation ??
08:37.46AstNewbieThanks in advance.
08:44.48AstNewbieI have also called the telecom ... but telecom seems not having any clues also ....
08:44.49AstNewbieI am not so sure which part causing the problem ....
08:44.51AstNewbieDo you have any idea ??
08:46.05luke-jrno
08:46.35Strom_MAstNewbie: I'd suggest waiting until the start of business hours in Alabama, and then call Digium for support
08:46.50awkgrr, anyone know why the lights are reversed now with parking
08:46.56awkwhen a call is not parked lights are on the phone
08:47.01awkbut when the call goes onto park it goes off?
08:47.06awkwhat couldbe the reason for this
08:47.13AstNewbieThanks Strom_M ...
08:47.14AstNewbie^_^
08:47.15awkthis is some bug with asterisk, any way to resolve this
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09:08.54yidiyuehanHi, Any one can give me a sample configuration file for ISDN card with bristuff driver?
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09:23.18pimoussanyone here could help me on asterisk-gui ?
09:25.04awk@*$%(*$U&@^*$!@*^T*#$^T$*#T$UQ#*$T#(*#$
09:25.12awkanyone know of any issue with hints on 1.4
09:25.18awkmy lights are reversed
09:25.28awkit doesn't show the lights on the snom when a call is parked
09:26.24loompekwell
09:26.29loompekany asterisk gurus here?
09:26.42loompekor is there really noone that could help me
09:26.42loompek:S
09:27.39thewiizledo you have your register string enabled?
09:27.45thewiizlewhat happens when you type sip show registry
09:30.09pimoussit seems to be dead
09:30.13pimoussnobody answers us on this chan
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09:35.42loompeksip show registry
09:35.42loompekHost Username Refresh State Reg.Time
09:35.42loompek*CLI>
09:35.43loompekblank
09:35.51awkgrrr no hints what so ever
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09:38.09loompekso ... how do i enable register strings?
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09:39.27KpoHwhy "realtime mysql status" causes asterisk crash?!
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09:44.49pimoussanyone to help please, guys ?
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09:47.19tzafrir_homehelp about what? I didn't you you actually asking anything
09:47.55awktzafrir what is the issue with hints in 1.4
09:48.02awkmy parked is all messed up
09:48.09awkit doesn't show a parked call, the lights dont show on
09:48.17awkis there something i should know about?
09:49.20pimoussyes, i had a problem about using asterisk-hui
09:49.21pimoussgui
09:50.21tzafrir_homepimouss, you don't seem to want others to help you with your problems
09:50.41tzafrir_homeOr are otherwise quite shy at mentioning them
09:50.58tzafrir_homeUnless you call asterisk-gui your problem ;-)
09:51.14tzafrir_homeThis is the only piece of information we have lerned about you so far
09:51.54tzafrir_homeawk, what have you configured? what do you expect to happen? what actually happens?
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10:07.52yidiyuehanHi, Any one can give me a sample configuration file for ISDN card with bristuff driver?
10:08.06awktzafrir: I have setup a parked group 701 right
10:08.15awknow I call into an extension right
10:08.23awkI have added on the snom to monitor 701
10:08.37awknow I put the call on park but it does not show the light on the phone saying their is a call parked
10:08.39awkyet if
10:08.47awki hit that function key it goes to that parked call
10:08.52awkso its not showing the parked calls
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10:09.00awkif I do s a show hints its just unavialble
10:09.24awkand we using metermaid to assign blf to parking bays
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10:10.05awk701@uditelco-local      : Local/701@parkedcall  State:Unavailable     Watchers  0
10:11.11bintuttzafrir: faidon forwarded already the http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=438702 to http://bugs.digium.com/bug_view_page.php?bug_id=10780
10:11.32awkin extensions.conf I have exten           => 701,hint,Local/701@parkedcalls
10:11.42awkwhat am I missing, it has a hint
10:12.53awkand I have set in a include extensions-app exten           => 701,1,ParkedCall(701)
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10:37.44disposableis it possible to have a ringgroup among extensions defined in a ringgroup?
10:41.29thieumsHello, i would need some help regarding mysql realtime. I would like to link a context from the sip table to the extension table (ie without using the switch statment in extensions.conf). Do you know if it's possible ?
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10:49.44ai-a[out]Question about Faxing on Asterisk... how come we can fax from uk -> auz. guessing its going over internet and other stuff but we cant fax from one fax machine to asterisk on the same switch with no other data ?
10:53.48thewiizleanyone pretty up to speed on SPA dialplans?
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10:56.15nDuffai-a: you generally can't fax over the internet, at all.
10:56.26nDuffai-a: even faxing over LANs is iffy, unless you're using T.37 or T.38.
10:57.08nDuff...in either of those cases, faxing over the Internet works fine, but Asterisk doesn't natively support those protocols, and (in the US, at least) using either of them gives up legal protections which faxes otherwise enjoy.
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10:58.13masushi all, which ports use asterisk anybody know ?
10:58.15nDuffai-a: faxing is best done as a 100%-zaptel affair, or using something like iaxmodem where the only VoIP span is internal to the server. (Something like TDMoE which guarantees timing is fine too, of course)
10:58.23nDuffmasus: depends, what are you doing with it?
10:58.47nDuffmasus: Asterisk has support a wide variety of protocols; what ports it needs depends on what protocols you use.
10:59.02nDuffmasus: if you want to know which ports your specific installation is using, use a tool like netstat to tell you.
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11:03.57TUplink_i keep getting a funny error....... [Sep 21 06:47:01] WARNING[78048]: chan_sip.c:8128 check_auth: username mismatch, have <20001>, digest has <>
11:03.57TUplink_[Sep 21 06:47:01] NOTICE[78048]: chan_sip.c:13388 handle_request_invite: Failed to authenticate user "Tommy Huff"<sip:20001@24.126.34.203>;tag=cb227e18-2710-3d7f27c1-8605-597a0dac
11:03.57TUplink_<PROTECTED>
11:04.12TUplink_it use to work... then my ATA lost its config... and now its all FOBAR
11:06.29nDuffTUplink_: check the "auth user" setting in the ATA's settings.
11:06.47TUplink_ok... i will but im sure its right
11:06.48nDuffTUplink_: should match the regular username.
11:06.58nDuffTUplink_: are you sure there's an entry in that field at all?
11:07.02TUplink_have to get on the winblows comp to check the xml then upload it
11:07.07nDuffTUplink_: the error you gave reads like "auth user" is blank.
11:07.08thieumsany realtime specialist here ?
11:07.36TUplink_when i paste that error in a webbrowser it comes up as a funnt box
11:08.03TUplink_its noty even registering
11:09.13TUplink_let me go check
11:11.19masusnDuff: what i have done is a outgoing with 2 internet connections thrue ADSL + GSHDSL , it's working but atADSL i hav NAT problems so i'll do port forwarding
11:11.49masusso i need , which ports are used or which range of ports
11:12.42nDuffmasus: you still haven't told me what protocols you're using.
11:13.08nDuffIAX? SIP?
11:13.08masusi haven't told u have ask ?
11:13.15masusSIP
11:13.32masussorry my english is not good
11:13.42thieums5060 udp
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11:23.00HarisGuys, can I configure myself, Vonage inbound on a linksys RTP300 ?
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11:28.23ai-anDuff: well we have a softfax on this server that works quite well.  However we've got only the pbx and ata box on a switch with nothing else. and its 100% failure so far.
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11:31.34TUplink_nDuff..... not sure what i did i removed the proxy and changed the caller id name to match the extension and now it works
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11:35.21hieunm_vipshi all, I want to change volume of sound when playback to user, how could I do this with "Playback" application?
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11:35.58hieunm_vipsOr is there any other applications support my case ?
11:47.48Uatechey if i monitor() a call to the same file every time, does it concatenate the file?
11:53.30hwthey. i am looking for a way to tap into the INCOMING voice to a meetme conference, to kick users that are making noise.
11:53.33hwtany suggestions?
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11:57.51lirakismorning
11:59.48hwtis it at all possible?
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12:45.48_x86_anyone use OctWare SoftEcho with Sangoma cards that already have HWEC?
12:46.23Uatechey there
12:46.31Uatecwell i'm about to buy an a500
12:46.31_x86_morning ;)
12:46.38_x86_a500?
12:46.42_x86_whats that?
12:46.43Uatecthe sangoma BRI
12:46.46_x86_ah
12:46.55Uatecand i want to decide if i want to get the echo cancellation module
12:46.57Uatecit's £150
12:47.19_x86_dunno about with BRI/PRI, because those are digital interfaces anyway
12:47.43_x86_POTS in .us are analog, and you almost always need HWEC
12:48.17UatecHWEC?
12:48.22UatecHard white enveolope cuttings?
12:48.27_x86_hardware echo cancellation
12:48.32Uatecahh
12:48.50Uatec£150, hmmm
12:48.53Uateci think we do want it
12:49.33tzafrir_homeUatec, is it actually supported? do current Sangoma drivers support it?
12:50.59Uatecwell, the sangoma cards come with an optional echo cancellation module
12:51.12Uatecit would seem pointless if the drivers didn't support it
12:59.10_x86_tzafrir_home: yeah, sangoma drivers support it
12:59.41_x86_Uatec: it's not enabled by default, you have to use the wancfg utility (included with the drivers) to manually enable it on each span
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13:01.18tzafrir_homeright, I see now
13:01.26tzafrir_homehttp://wiki.sangoma.com/sangoma-wanpipe-smg-asterisk-bri-installation
13:01.38tzafrir_homenice of them to use bristuff
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13:05.01pimousshello
13:05.04pimoussI have a problem trying to get GUI to work
13:05.04pimoussasterisk's http is responding on 8080  but answers File not found on /
13:05.04pimoussany idea please ?
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13:09.45tzafrir_homepimouss, I don't think that / should have a file
13:09.58tzafrir_homeA redirection from / as an optional setup, IIRC
13:11.53Kurin-Is there any reason a polycom phone would break a router or a network connection?
13:12.10pimoussyes, indeed, but none of the expected URLs work
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13:12.16Kurin-We're installing these phones and three times now when we've plugged the phone in the router's have died
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13:12.39deeperrorKurin: are phone lines plugged into network jacks?
13:12.44Kurin-and one was a fbsd box, and when the nic died ifconfig said "no carrier" but the nic showed a link light
13:12.47*** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca)
13:13.06Kurin-The patch cord goes from the phone's "LAN" jack to the switch
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13:14.29nacerhey hey
13:14.49nacersomeone know a good documentation for make update of asterisknow ?
13:15.38lolscorruptionofconary update conary
13:15.52lolscorruptionofconary update asterisk zaptel libpri
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13:16.03naceroki
13:16.08nacerhow to have a shell ?
13:16.30lolscorruptionofChange to another virtual term
13:17.15keulinwhy my genzaptelconf script is displaying for about on hour "Generating '/etc/zaptel.conf'"
13:17.18keulin?
13:18.06nacerok
13:18.51modujust a question about IAX/RSA : there is no way to sign IAX paquet to prevent man in the middle attacks ?
13:19.07[TK]D-Fendernacer: You're in the wrong channel, please read the topic.
13:19.17nacerok
13:19.44nacer[TK]D-Fender, tks i am in the good one now :)
13:20.05nacertks lolscorruptionof  for your help
13:20.53keulinno idea about this issue ?
13:22.22[TK]D-Fenderkeulin: just hit enter.  if you're not at aprompt, just ctrl-c out
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13:24.07keulin[TK]D-Fender, yes, i've allready done that but in this case /etc/zaptel.conf is not generated
13:24.17keulinI don't understand why
13:24.29[TK]D-Fenderjust build them yourself.  its a handful of lines...
13:26.10rob0./gen[TK]D-Fender.conf && echo it worked
13:27.33keulinyes i know how to do that, but i would like to make this work
13:27.37moduis IAX encryption is stable now ?
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13:32.51VJFROMGTi have an outbound route with matching pattern but evertime i try to make a call i get "no rotue to destination"
13:36.51[TK]D-FenderVJFROMGT: pastebin the call CLI with channel debug enabled.
13:42.03Kurin-So no one's ever had a polycom phone break the switch it's plugged into?
13:42.39[TK]D-FenderKurin-: nope....I have seen 2 Polycoms simply "die" however
13:42.54Kurin-No, this is definitely the switch
13:43.10Kurin-He plugs his phone in and his PC gets knocked offline and can't ping the gateway
13:43.19Kurin-but when he brings his phone in it still works fine
13:43.26JTusername=jesse
13:43.26JTsecret=jghb572
13:43.30JToops
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13:43.39keulinlol
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13:43.58Kurin-We all have the same password
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13:44.07Kurin-I wish you could put in your SIP credentials via the phone
13:44.18[TK]D-FenderKurin-: ?
13:44.28Kurin-Just musing at JT's misspaste
13:44.31Kurin-mispaste
13:44.55[TK]D-FenderKurin-: No, wondering about the "sip credentials on phone" bit...
13:44.56Kurin-Yeah so I'm pretty sure the phone is somehow breaking the switch, since when I plugged it into my fbsd machine it took the NIC offline
13:45.15[TK]D-FenderKurin-: What model?
13:45.16deeperrorit probably has a hard coded ip?
13:45.17Kurin-Well the polycom downloads its config file, which has the sip username/password in it
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13:45.28Kurin-Soundpoint 550 and 330
13:45.45Kurin-No it gets its IP from DHCP
13:45.52[TK]D-FenderKurin-: You can enter auth infor directly in the phones LCD menus, in the web admin, through provisioning files, and even through DHCP.
13:45.58JTheh you have no sip hostname anyway, mwuahaha
13:46.07Kurin-And in fact it's on the network for a few seconds
13:46.12Kurin-Since I can see packets going through
13:46.16Kurin-But then everything just stops
13:46.17JTi meant to paste http://youtube.com/watch?v=AmCc6MEhNGM rofl!
13:46.25Kurin-I'm not really concerned about auth info
13:46.26[TK]D-FenderKurin-: Any chance he doesn't have PoE at home and doesn't have an injector?
13:47.03Kurin-Well he doesn't have PoE, no, but these phones all use regular power cords
13:47.55bintutgtg..
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13:49.18Kurin-Could it be that the phone is doing some PoE stuff out the LAN interface and needs the patch cable that came with it?
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13:52.54tzangermorning
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13:56.18tzangerhe's here!!
13:56.21tzangergood morning, coppice
13:56.28tzangeror evening for you I suspect
13:57.14coppicewell, I feel sleepy. either its bedtime, or I'm in an important meeting
13:57.20tzangerhaha
13:57.21hwthey. i am looking for a way to tap into the INCOMING voice to a meetme conference, to kick users that are making noise. any suggestions?
13:57.33tzangerI have a datapoint and question for you, if you have time
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13:58.35tzangerI have access to a system with a tdm400 in which the zaptel-provided zttest program consistently reports 100% for timing accuracy (8192 samples in 8192 sample periods), yet sliptest returns results all over the map (inconsistent correlation between emitted sound and detected audio)
13:58.57tzangerreplaced the card and sliptest is now reporting what I'd expect (about 508/516)
13:59.18Kurin-Yeah that must be it
13:59.30Kurin-I plugged the phone on my desk in with a different cable and made a call
13:59.38Kurin-Within about a second it died
13:59.53tzangermy question is how would the two (zttest and sliptest) be at such huge odds with each other?  I understand they're measuring timing differently (one strictly time, the other measuring audio path loop) but it seems strange
13:59.55Kurin-though it recovered
14:00.13tzangersliptest appears to work equally well on FXS and FXO ports, at least on my TDM400P at home here
14:07.21[TK]D-Fenderhwt : why not jsut MUTE them out instead?
14:09.06[TK]D-Fenderhwt: And you could always parse out "show meetme X" to see who's talking...
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14:25.09MindTheGaphello all, i need to answer a zap channen call and if I get an specific sound i want * to drop the call. so far i'm having 60% success at this using backgrouddetect because sometimes it detects this sound as a "DTMF 2". but its not consistent. is a pre recorded call from the telco warning me that this is a collect call and I should hangup if i do not wish to pay for the call. As you may have notice, the telco is not willing to block collect calls at the
14:25.10MindTheGapir end, so im stuck. Its a E1 ISDN.
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14:27.54tzafrir_homeMindTheGap, "drop the call if I get a specific sound": that's busydetect?
14:28.04tzafrir_homeBut what if htat sound occours randomly?
14:31.12MindTheGaptzafrir_home, it is a pre recorded message, its the same all the time, but theres music in it and sometimes its recognized as DTMF
14:31.58tzafrir_homeMindTheGap, if it is ISDN, I suspect that there are smarter ways to know you should hang up
14:33.24MindTheGaptzafrir_home, please enlighten me :)
14:33.58Uatechey, is there any hardware i can use to simulate an ISDN line?
14:34.15JTan isdn simulator
14:34.25russellbasterisk with a T1/E1 card in it?  :)
14:34.54tzafrir_homeUatec, hardware? sure. A dual ISDN adapter. Did you ask about software?
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14:35.33Uatechardware
14:35.34MindTheGaptzafrir_home, i know it is possible on a E1/R2 using unicall, but im not aware of any methosd on ISDN.
14:35.39Uateci have just bought a sangoma a500
14:35.51tzafrir_homethat's ISDN BRI, not PRI
14:35.54Uatecbut i don't have an ISDN BRI line to test it on
14:36.04Uateci know, i didn't say PRI
14:36.37tzafrir_homeUatec, Asterisk with a cheapo HFC-s card?
14:37.30UatecHFC-s card?
14:37.33Uatechmm
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14:38.37Uatecand hfc-s doesn't appear too useful for finding cards on google
14:38.50tzafrir_homecologne HFC-s is the name of the chipset
14:39.45Uatecknow any cards which use it?
14:40.29tzafrir_homeBillion , hmm
14:40.47[TK]D-Fendertzafrir_home: No, I'm sure he jsut needs one ;)
14:42.27UatecWAIT
14:42.37nacer?fxo fxs
14:42.54tzangerNT = network termination, TE = terminal endpoint?  I can't remember
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14:44.00[TK]D-Fenderouch
14:44.18*** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)
14:44.36santibioticoTE = Terminal Equipment
14:45.40Uatecso i could actually set one of my ports of my b410p to TE and one of them to NT and have them connect to each other ?
14:45.51*** join/#asterisk tristanbob (n=tristan@oalug/member/tristanbob)
14:45.58Uatecthat sounds too simple
14:46.06JTUatec: correct
14:46.08tzafrir_homeUatec, right
14:46.10JTi've done it before
14:46.43tzafrir_homeUatec, just make sure you use a cable with all 8 wires
14:46.47tzafrir_homenon-crossed
14:47.13tzafrir_home(all 8 wires: ISDN BRI uses just 4, but not exactly the same 4 as ethernet)
14:47.28Uatecok, well all our cables here have all 8
14:47.30Uatecawesome
14:47.38Uateci just became very very very happy and excited
14:47.41tzafrir_home~fxsfxo
14:47.42jbotwell, fxsfxo is an FXO port expects to receive dialtone and receive ring voltage. You can connect it to a PSTN line from the telco. An FXS port expects to provide dialtone and provide ring voltage. You can connect a phone or a fax to it.
14:49.18tzangercoppice: any ideas about my question regarding HUGE difference between zttest and sliptest?
14:50.19coppicetzanger maybe it matches so fantastically well, there is no echo
14:50.57tzangercoppice: an unterminated line on a TDM400 acting as a perfectly terminated line?  surely you are joking with me :-)
14:51.40coppicehow does it sound? is the card creating a horrible distorted mess that won't correlate?
14:52.18tzangerI must admit I have not listened to a thing, I'm only using the card as a source of timing for tdmoe
14:52.34*** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)
14:53.26tzangersliptest should work equally well with FXO and FXS ports, right?  I think it would (but I'm not the brightest, etiher)
14:53.41*** join/#asterisk KpoH (n=AID@host-89-41-66-8.moldtelecom.md)
14:54.04tzangerit seems to work on my home * box anyway on both types of ports, and with digitla lines that are looped back it also seems to respond as I'd expect it to
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14:57.02tzafrir_hometzanger, I commited some fixes to zttest a week ago or so. With them as well?
14:57.42tzafrir_homeMake sure you use zttest -v, and also look at the final result after you press ctrl-C
14:57.55tzangerno this is a 1.4.3 pull of zaptel (i'm locked to thsi version for the client)
14:58.13tzangertzafrir_home: yeah I am getting what I expect to be normal results with the new tdm400
14:58.26tzafrir_homelocked version? just pull it from somewhere as source  / binary
14:58.33tzangerI know :-)
14:58.43tzangerjust haven't got to it yet,a nd they're all off for Yom Kipur now anyway
14:58.46tzangerso I have to wait for Monday
14:59.19*** join/#asterisk mtaht4 (n=m@172-110-62-200.enitel.net.ni)
14:59.45tzafrir_homeWhat is sliptest?
15:00.10*** part/#asterisk mtaht4 (n=m@172-110-62-200.enitel.net.ni)
15:00.39*** join/#asterisk Lawbringer (n=Lawbring@212.183.134.130)
15:04.29tzangertzafrir_home: it's a little utility the dsp guru coppice came up with which opens a zap channel and spews AWGN on the line.  it then listens to the line and auto-correlates the transmitted audio to the received audio
15:04.50tzangeressentially it determines the loop length, which for any TDM circuit should be pretty much consistent at *some* value
15:04.58tzafrir_home~AWGN
15:05.19tzangeradditive gaussian white nose
15:05.27coppiceA White Guy's Noise
15:05.36tzangeryou mean like Eminem?
15:05.41tzafrir_homelooks like my toolkit is missing some pretty useful tools from spandsp
15:05.47[TK]D-Fender"He didn't have Ice Cube, so he brought Vanilla Ice"
15:05.56tzangerI didn't realize you could write his music mathematically :-)
15:07.17tzangertzafrir_home: yeah I love this app
15:07.45tzangerit tells you in one quick test whether a) timing is consistent b) path is consistent and c) audio should work
15:09.10nDuffai-a: what codec is the ata using?
15:09.38tzafrir_homecoppice, is it included in spandsp?
15:09.48*** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net)
15:10.06tzangertzafrir_home: no
15:10.35tzangerhttp://www.soft-switch.org/downloads/sliptest.c
15:10.38nDuffai-a: you should be using alaw or ulaw.
15:11.15tzangerer sli8ptest
15:11.19tzangerdammit
15:11.19tzangersliptest
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15:14.27tzafrir_homecoppice, mind you that you use <linux/zaptel.h> which has changed to <zaptel/zaptel.h> in zaptel 1.4
15:14.44AeudianQuestion: Is there a way to stop extensions.ael from being parsed on reload?  my guess would be add a noload line in modules, but i am not sure what.
15:17.24nDuffAeudian: pbx_ael
15:18.19coppicetzafrir_home: 1.4 didn't exists when I wrote that code
15:18.27AeudiannDuff: noload => pbx_ael in modules.conf right?
15:18.52nDuffAeudian: sounds about right, yes.
15:19.22AeudiannDuff: still getting notifications from pbx_load_module
15:19.26outtoluncrename it <G>
15:19.29[TK]D-Fendernoloca => pbx_ael.so
15:19.41[TK]D-Fendernoload => pbx_ael.so
15:20.46tzafrir_homeAnyway, now zttest (in svn) should count only the time it actually spends reading. I sitll suspect it is a bit inaccurte
15:22.26coppicewhat does zttest actully do? it used to be something entirely meaningless
15:22.40AeudianTK: i did that but pbx_load_module still attempts to find the file, my guess cause i see an auto load all?
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15:23.22nDuffcoppice: counts percentage of interrupts serviced, if I understand correctly (!= guaranteed)
15:23.47coppicenDuff: that is awfully vague
15:23.59nDuffcoppice: that *is* useful -- if for no other reason for telling people for whom the percentage is too low that they need to go fix their systems before faxing will wore correctly.
15:24.16tzangercoppice: my understanding is that it waits 8192 sample periods, and then counts the actual number of samples receieved in that period
15:24.17coppiceit tells them nothing of the sort
15:24.18[TK]D-Fender<PROTECTED>
15:24.45tzafrir_homecompare time as messured by the clock generated by Zaptel to the system clock. One thing it does very well is detecting when the Zaptel clock is very bad: non-existant, badly lagging, etc.
15:25.02AeudianTK: ya it works, i forgot i had to restart asterisk service a simple reload wouldn't do it, thanks
15:25.27coppicetzafrir_home: if that is what it does, it is entirely useless
15:25.30tzangertzafrir_home: I've found sliptest to be far far more useful in that regard
15:25.55nDuffcoppice: there's a well-established relationship between zttest numbers being below 99.98% and faxing being iffy.
15:26.06tzangernDuff: I don't believe that
15:26.10*** join/#asterisk ManxPower (n=manxpowe@202.sub-70-220-216.myvzw.com)
15:26.19tzangerI have several systems where zttest tells me that faxing will never work, where it in fact works very well
15:26.30tzafrir_hometzanger, well, does sliptest need an actual Zaptel channel?
15:26.31tzangerzttest ~95% but consistent results
15:26.34tzangertzafrir_home: yes
15:26.50tzafrir_homeor can it use a single pseudo channel?
15:27.09etfonhomey_[TK]D-Fender Which Polycom phone do you recommend for the secretary/power user?
15:27.11tzangertzafrir_home: it requires an actual channel, as it actually spits audio out and listens for its reflection
15:27.40coppicenDuff: its telling you something completely bogus
15:27.44tzafrir_homewell, you don't always have a handy channel. In fact, when Asterisk runs, you don't
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15:27.52tzangertzafrir_home: agreed
15:28.02tzangerbut without an actual channel, how do you measure actual path timing?
15:28.02coppicefaxing is *not* dependent on the system clock at all
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15:29.09nDuffcoppice: *shrug*. I'm relaying what's quite typically painted as a scapegoat on the iaxmodem list
15:29.33coppiceif there is an echo anywheer down the line, sliptest will tell you is the loop length is stable. that is a meaningful test of what will work for faxing
15:29.33nDuffcoppice: ...including by folks who should know better if it's not true.
15:29.53FXOLanyone got some time to help w/ a basic php problem?
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15:30.33tzafrir_homecoppice, right. But if the zaptel clock is not very close to the system clock, chances are that the Zaptel clock is bogus. This is why zttest is very good at detecting simple problems with it
15:32.47|NexT|8192 zaptel samples in 8192.217 system clock sample intervals (100.003%)
15:32.53|NexT|this is normal?
15:33.13tzafrir_homenDuff, I am actually yyet to messure a clear 100% on systems where Zaptel is clocked by our devices. But they pass faxes pretty well
15:33.28tzafrir_homeWe do have other indications of where things slip
15:33.32jsmithFXOL: Well, I could probably help you out, but this isn't really the right channel for it... what's your question, and then we'll take this off the channel
15:34.04nDufftzafrir: I've actually gotten 100%, but only on multicore systems using Sangoma's PCI-E boards.
15:34.08tzafrir_homezttest is not entierely accurate. And its display of multiple digits of precision just makes it appear "accurate"
15:34.36FXOL(jsmith): I got help... fixed, thanks ;P
15:35.06FXOL(jsmith): Had to modify my PHP script to help get Cepstral working right, and apparently screwed up one little thing :P
15:35.40[TK]D-Fenderetfonhomey_: Power users don't normally factor in, receptionists = IP 650
15:35.57etfonhomey_Thanks!
15:36.15ManxPowerCharter Cablemodem tech support - when you need a good laugh.
15:36.44etfonhomey_What artifacts would you hear in the audio of a phone call if there were issues with jitter?
15:36.49rob0"What version of Windows are you running?"
15:36.55[TK]D-Fenderetfonhomey_: Basically its not a big premium over the 601 any more and its a backlit screen, HD (like that matters....), USb expansion, etc...
15:37.06ManxPoweretfonhomey_: think bad cellphone connection
15:37.08[TK]D-Fenderetfonhomey_: Mayan <-
15:37.12jsmithetfonhomey_: Pops, crackles, something that sounds like a bad cell phone
15:37.32jsmith[TK]D-Fender: I think they're Aztec, not Mayan
15:37.34jsmith:-)
15:37.44[TK]D-Fenderjsmith: The jury is out..
15:37.57ManxPoweryou don't get pops and crackles bad cell connections unless you are using ANALOG
15:39.45Corydon76-digor eating rice krispies on the phone
15:40.17FXOL(jsmith): wanna test? :P
15:40.58jsmithFXOL: What exactly would I be testing?
15:41.07FXOLnaw.. it's ok ;P
15:41.35FXOLjust part of my IVR
15:42.12*** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br)
15:42.20flujanhi all.
15:42.42*** join/#asterisk swombat (n=KDan@87-194-122-30.bethere.co.uk)
15:42.43flujanis it possible to asterisk record a extension state on a postgresql database?
15:43.02flujanswombat: here I am ... :D
15:43.12flujan[TK]D-Fender: hi man... how are you doing?
15:43.19swombatflujan: I think your question should be: "Is it possible to get an AGI script to be called every time the extension state changes?
15:43.22swombat"
15:43.35swombatif you can get an AGI script called, you can do whatever you want to the db from there
15:43.38flujanswombat: yeap.. It will also solve the probem... :D
15:43.46flujanswombat: for sure...
15:44.02swombatso. now we await the answers from all these lovely people here :-)
15:44.07FXOLwow
15:44.11[TK]D-Fenderswombat: It would not be AGI you'd be wanting to do, and yes.  You could have a script monitoring AMI for the state change notification messages and act accordingly.
15:44.12FXOLmultiport Cepstral is $
15:44.40swombat[TK]D-Fender: flujan is trying to resolve a performance issue because monitoring 500 extensions gets a bit slow
15:45.02swombat[TK]D-Fender: so i suggested he should update the status in the db when it changes, rather than polling it every 5 seconds across hundreds or even thousands of users
15:45.30[TK]D-Fenderswombat / flujan :elaborate on how you expect to monitor them.
15:46.01swombatI was suggesting that he has some sort of AGI call whenever the state changes, and that that call then updates the db
15:47.19flujan[TK]D-Fender: yeap... Actually I am using AJAM to do this... But I run a loop on each extension, parsing the results from each on them and updating each user on the database.
15:47.41[TK]D-FenderAJAM?
15:48.15outtoluncits the backend asterisk-gui uses to talk to manager
15:48.53[TK]D-Fender:/
15:48.54flujan[TK]D-Fender: yeap... the manager api that I can acess like a web-service.
15:48.57*** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr)
15:49.20flujanhttp://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+%28AJAM%29
15:49.40ManxPowerusing an AGI everytime an extension state changes will
15:49.50ManxPowerNOT "improve performance"
15:50.07swombatManxPower: not on the asterisk side
15:50.27swombatbut the bit where performance is dying is where he loops through a thousand states every 5 seconds to keep the db up to date
15:50.29flujanManxPower: I run a script that collects the extensions state every 5 seconds and stores it on a db.
15:50.48flujanthe script takes about 5 seconds to run...
15:51.24flujanManxPower: how much performance I will loose using this behavior of asteirsk calling a agi script ?
15:51.49ManxPowerflujan: I don't know, but there are several forks and execs involved.
15:52.15swombatthat's irrelevant
15:52.26swombati presume there are not 500 state changes per second
15:52.32flujanswombat: the forks and execs?
15:52.33swombator per 5 seconds even
15:52.48swombatflujan: assumption: state changes much less than 500 times per 5 seconds
15:52.50swombatcorrect?
15:52.59flujanswombat: for sure...
15:53.03swombat(either that, or you have 500 users online all the time and switching like crazy)
15:53.21swombattherefore, even if the AGI takes 10x as long as the AJAM, it doesn't matter because it's called much more rarely
15:53.40flujanswombat: nops they not swich states like crazys...
15:53.50flujanyeap.
15:53.57twistedi've got the world on a string
15:53.58swombatagi calls are near-instantaneous on my server. Compared to your 5 second loop, you might as well consider them free
15:54.14flujanManxPower [TK]D-Fender how can I put in on the dialplan?
15:54.34ManxPowerflujan: Did any say you CAN run an AGI everytime an extension state changes?
15:54.42ManxPowerI'm not aware of that feature
15:55.48*** join/#asterisk bmd (n=bmd@72.54.252.34)
15:55.49flujanManxPower: ... [TK]D-Fender suggest a AMI that will notify asterisk on each state change...
15:56.07CCFL_Man2is Mark here?
15:56.15[TK]D-Fenderflujan: What place does this have in the DIALPLAN?
15:56.59[TK]D-Fenderflujan: What do you want to do on state change exactly?
15:57.26swombathe wants to update a single field in a pgsql db
15:57.42flujan[TK]D-Fender: just have the control of which users are on the phone, which is not logged and so on... I will display it on a report.
15:58.05flujan[TK]D-Fender: to know if the users are on the phone... Dialing and so on.
15:58.15[TK]D-Fenderflujan: the just make a COMPLETELY seperate script (no AGI or dialplan involved), which will poll AMI for state change messages and do your DB work.
15:58.27coppicetzafrir_home: sure, its some kind of sanity check, but people try reading significance into 99% vs 99.9%, and are told that above some figure is perfect. its complete fiction
15:58.33swombat[TK]D-Fender: that is SLOW.
15:58.45flujan[TK]D-Fender: I already did it...
15:59.17flujan[TK]D-Fender: this is working right now... but it is slow... for each extension I need to update a row on the database...
15:59.29flujanthis update have a serious cost...
15:59.33[TK]D-Fenderhow is it slow?  Its REALTIME
15:59.52[TK]D-Fenderyou don't poll EVERYBODY, you just bloody well wait for INDIVIDUAL events!
16:00.03[TK]D-Fenderpsychos....
16:00.04swombatah well. that would work.
16:00.11swombatif it's possible
16:00.13[TK]D-Fender:p
16:00.14flujan[TK]D-Fender: hum...
16:00.17tzafrir_homecoppice, I agree with you there
16:00.32flujan[TK]D-Fender: yeap it will solve the issue, but how to wait for individual events from the AMI?
16:00.33swombatflujan: that's another possible solution then - if you can make the poll only return rows for users which have state changes
16:00.48swombatand with this, i'm out. have fun :-)
16:00.50flujanswombat: for sure...
16:00.58[TK]D-Fenderflujan: Open a tcp socket and sit around waiting for events.  Its called PROGRAMMING <-
16:01.18flujan[TK]D-Fender: I am currently using the extensionstate command...
16:01.27coppicetzafrir_home: but people do report cases where that test gives lousy results, but there is not evidence of actual data loss
16:01.43[TK]D-Fenderflujan: STOP
16:01.46CCFL_Man2crap, this guy doesn't take paypal
16:02.00[TK]D-Fenderflujan: that YOU asking for it.  this is the opposite of what I am advising.
16:02.23flujan[TK]D-Fender: do you recommend creating a socket on the ami port and just process the output?
16:02.40tzafrir_homecoppice, a short test will not catch an occasional slip
16:02.53[TK]D-Fenderflujan: Yes, watch the events go by in real-time.  Your script will stay open the WHOLE TIME.
16:02.57tzafrir_homemay or may not catch
16:03.00CCFL_Man2[TK]D-Fender: the Mark who collects western electric phones, whats his nick?
16:03.19flujan[TK]D-Fender: thanks for the tip... I will meditate on that... :D
16:03.40[TK]D-FenderCCFL_Man2: Don't recall for certain.  Go check a channel archive
16:03.41JTprogramming > prayers
16:03.43coppicetzafrir_home: but why would people get a lousy result when they see no errors at all when faxinging without ECM
16:03.51flujanswombat: thanks for the help too... I will see what can I do with the socket solution
16:04.01JTcoppice: either strom or his friend
16:04.02swombatgood stuff. good luck!
16:04.11[TK]D-FenderJT : I keep a private reserve of buddist monks aside in case of urgencies regardless ;)
16:04.12JTs/coppice/CCFL_Man2/
16:04.15coppiceit certainly won't catch the occassional slip. its bogus when you use it for that fine detail
16:04.25CCFL_Man2ahh
16:04.27JT[TK]D-Fender: always useful
16:04.48[TK]D-FenderJT : My karma ran over your dogma :p
16:05.10CCFL_Man2JT: it's Strom_M's friend
16:05.18JTheh
16:05.26davevg-btwtechflujan: If you are somewhat ok in perl, try POE::Component::Client::Asterisk::Manager
16:05.36Strom_Mwhich friend?
16:05.42Strom_Moh, Mark
16:05.44CCFL_Man2Strom_M: Mark
16:05.44tzafrir_homeor Asterisk::Manager
16:05.50Strom_MCCFL_Man2: his handle is rudholm
16:05.55tzafrir_homewhich is probably not that good
16:06.10flujanthanks  davevg-btwtech :)
16:06.13CCFL_Man2Strom_M: ahh, ok, i think i talked to him before
16:06.59*** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com)
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16:10.56CCFL_Man2Strom_M: i'm glad you're here, i need advice
16:11.26Strom_Mok?
16:11.56CCFL_Man2i need a ringer for my green imperial WE202
16:12.00pots_lineD-Fender:  What do you do to monitor state and presence for 100s of phones?
16:12.48*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
16:13.04CCFL_Man2Strom_M: there is a 684A on ebay, i'm guessing it's better for authenticity because it's newer than the 543 ringers?
16:13.42Strom_Mi dont know a damn thing about 202s
16:13.48[TK]D-Fenderpots_line: Depends on poll-frequency, etc
16:13.56pots_linefor BLF
16:14.00CCFL_Man2Strom_M: neither do i :P
16:14.10pots_linewould need to be pretty often
16:14.31CCFL_Man2Strom_M: i'm guessing since mine is dated 50s i should get a newer ringer box with network
16:15.31CCFL_Man2apparently the network provides sidetone
16:15.43[TK]D-Fenderpots_line: Never had an install for someone to monitor that many.  I might use an Aastra 5i series phone + 2 LCD consoles.  Or a web script on 5s refresh.
16:17.16CCFL_Man2Strom_M: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&rd=1&item=170149395255&ssPageName=STRK:MEWA:IT&ih=007
16:17.17pots_lineJust curious . . . We have several 100+ phone installations that require it.  Presence stuff kind of whigs out the phones when you are monitoring that many.
16:17.22pots_lineGoing to have to go to an application instead of using the BLF on the phones
16:18.18[TK]D-Fenderpots_line: I've only jsut recently heard of some Polycom setup whigging out on a mass-page, but no details on model/firware combo.
16:18.21defswork[TK]D-Fender: is there existing web scripts to show that ? I could do with one
16:18.31JTmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmmm
16:18.33[TK]D-Fenderpots_line: But franly a web setup is jsut so much more readable.
16:18.44pots_lineI agree
16:18.57[TK]D-Fenderdefswork: Dunno, never looked.  I jsut code my own
16:19.07defswork[TK]D-Fender: open source it ! ;)
16:19.08JToops
16:19.19CCFL_Man2Strom_M: i'm not sure what i should do :P
16:19.22*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
16:19.36pots_linePolycom actually called and wanted to have us generate packet traces and debug to fix the problem.  But, they didn't call it a bug because they don't officially support Asterisk.
16:19.51pots_lineAt least they are being helpful.
16:20.01ZeeekHey yall, join #voip-users-conference for the um voip users conference at http://voipusersconference.org
16:20.05[TK]D-FenderMy polycoms monitor 2 queues, 4 agents login/pause/call status, and I have a general MB script for full company presence.
16:20.49pots_lineNone of the consoles 601s monitor less than 40 phones. . . . And, they make a ton of state change noise.
16:20.50defswork[TK]D-Fender: would you care to give me a copy so I can see how you go about it?
16:20.59CCFL_Man2pots_line: ahh, my favorite kind of phone line
16:21.12pots_line:-)
16:21.15GoRKdoes anyone have polycom's technical bulletin 25751 that explains the SRTP options in 2.2.0 firmware?
16:21.25[TK]D-Fenderdefswork: I just dump "show hints" via AMI and parse away nice & dirty like :)
16:21.34defsworkoh :)
16:21.38*** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net)
16:21.41flujan[TK]D-Fender: I am creating that script that connects to ami via socket an grab the information...
16:21.43pots_lineThat'll do it.
16:21.49defsworkI did something similar last night to create phonebooks for aastra phones
16:21.58flujanDoes AMI echos all state changes to the port or I need to ask for it?
16:22.00[TK]D-Fender<- the Red Green of coding.....
16:22.14flujan[TK]D-Fender: my program is connecting to it put just outputs nothing... :(
16:22.37[TK]D-Fenderflujan: Your code-fu is weak :|
16:22.56defsworkideally you'd have a monitor script that gets the ifno only once - save multiple clients getting it multiple times
16:23.01pots_lineAMI requires you to make a request right . . .
16:23.19[TK]D-Fenderpots_line: My normal way, yes.
16:23.21pots_lineSo, if you loop through every 5 secs or so
16:23.26flujan[TK]D-Fender: but I thought I was a Jedi... :(
16:23.32pots_lineyou can pretty well keep track of state
16:23.35Zeeekrussellb is anyone available today?
16:23.51flujan[TK]D-Fender: any way.. I need to send something to ami to get the new changes right?
16:24.10[TK]D-Fenderdefswork: For my polycom CSR polling I have *1* process poll, and it then generates a STATIC page for the others to load.
16:24.17fileZeeek: hrm!
16:24.32[TK]D-Fenderflujan: nope.  Go WIKI up the AMI to learn how it sends those events
16:24.58defswork[TK]D-Fender: yeah thats the right way to do it
16:25.17russellbZeeek: not sure, i'm not ...
16:25.23[TK]D-Fenderdefswork: My code is ugly but functional :)
16:25.29defswork[TK]D-Fender: and only poll when someone is watching too :)
16:25.29russellbit's going to be crazy around here for the next few weeks
16:25.49[TK]D-Fenderdefswork: Mine are always watching, its on the Polycom IDLE scrren <-
16:26.09russellbwe have astricon and then moving into our new building
16:26.21defswork[TK]D-Fender: do you cross reference with extension names ?
16:26.23GoRKif you have to make a lot of AMI connections you can also use 'astmanproxy' that reduces the load on asterisk
16:26.46CCFL_Man2i like b8zs instead of ami
16:26.50[TK]D-Fenderdefswork: only on the on-demand one.  the idle one has no room on screen.
16:27.00[TK]D-Fenderdefswork: I suppose I could do initials, but NAH....
16:27.07defswork:)
16:27.07[TK]D-Fenderdefswork: There's only 4 of them.
16:27.17defsworkI'll do something
16:27.20Strom_MCCFL_Man2: http://www.stromcarlson.com/misc/alternate_mark_inversion.png
16:27.43russellbo.O
16:28.09russellbStrom_M: http://www.russellbryant.net/DTMF_Task_Force.jpg
16:28.21Strom_Mbahaha
16:28.33CCFL_Man2Strom_M: stud
16:28.58*** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr)
16:29.35CCFL_Man2hah
16:29.50CCFL_Man2sweet
16:31.19CCFL_Man2i've let that bell box go
16:31.30CCFL_Man2i want a 685A
16:31.37*** join/#asterisk CVirus (n=GoD@82.201.174.251)
16:31.42[TK]D-FenderStud.... thats something that gets nailed to the wall and buried behind gyproc and only comes out when the whole house comes tumbling down, right? :)
16:31.54*** join/#asterisk ManxPower (n=manxpowe@71-8-61-95.dhcp.leds.al.charter.com)
16:32.09CCFL_Man2i'll paint the cover to match the green imperial WE202
16:32.14CCFL_Man2[TK]D-Fender: heh
16:35.51*** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com)
16:37.07RyushinI'm trouble shooting why the polycom phones aren't working on a remote network that connects to the asterisk server over a VPN.  I set up a SIP softphone on the server there, and it registered with asterisk just fine.  It's just that the polycom phones aren't.
16:38.16RyushinThe polycom log is here: http://www.pastebin.ca/706081
16:38.32*** join/#asterisk ikk (n=ikk@195.50.105.113)
16:38.43RyushinThe entries that I'm seeing are this:  Registration failed User: 7070, Error Code:480 Temporarily not available
16:39.10RyushinBut I'm not seeing any registration attempts when I'm watching asterisk using "asterisk -vvvvvvvvvr"
16:39.19ikkpeople where would i look to see why i cant make external connections - connections within the network work fine - but connections from internet do not :(
16:40.18Ryushinikk:  What protocol?
16:40.31ikkiax2
16:40.49RyushinTry using tcpdump.
16:40.53ikkwhen trying to use zopier i just get timeout when trying to connect
16:41.18RyushinDo you have the port available to connect to the internet?  It is UDP.
16:41.24*** join/#asterisk bmg505 (n=leon@196.209.180.191)
16:42.07ikkyes i can see the attempts via tcpdump but i just get timeouts in logs :(
16:42.25ikk17:42:09.105937 IP 192.168.1.66.4569 > XXX.XXX.XXX.XXX.4570: UDP, length 12
16:43.14ikkanyone on the internet 192.168.1.xx network can connect fine - but im external too ti and need to connect if possible
16:43.49davevg-btwtechRyushin, can the phones route correctly to the * box via IP?
16:43.52ManxPowerdid you forward the port 4569.  You need to do that when your asterisk is behind nat and you are using IAX2
16:44.52ikkyes i believe it is forwarded
16:46.01ikk(it must be otherwise i would be able to get as far as it showing in tcpdump on that server)
16:46.07ikkwould / would not
16:46.11Ryushindavevg-btwtech:  Well, the thing is, that that phones are pulling their configs fine via ftp.  I'm watching the vsftpd.log and that shows all the files are going there fine.
16:46.38RyushinSo they would have to route at least tcp traffic correctly.
16:47.01davevg-btwtechand the ftp server is on the other side of the vpn?
16:47.18RyushinThe only difference is that this is a new Cisco VPN when they got rid of the Fortigate Firewall VPN.
16:47.27ManxPowerand 1.66 is the IP of your server?
16:47.30RyushinYea, FTP is on the same server as asterisk.
16:47.57ikkManxPower, yes thats the ip of the asterisk server
16:48.01RyushinThe IP of the server is 172.17.127.15
16:48.40RyushinThey have a windows server on the same remote network, and I installed nmap to run a scan to make sure it could see UDP 5060.
16:48.42deeperrorare you fwd only tcp packets or udp as well?
16:49.08RyushinI wonder if zoiper will fall back to tcp on sip?
16:49.27RyushinIf zoiper did that, then I can see what the problem is.
16:49.27ManxPowerAsterisk does not support TCP for VoIP
16:49.54RyushinThen if zoiper worked via sip, I have to think that udp is working fine.
16:50.06ManxPowerRyushin: what about 10000-20000/UDP?
16:53.03*** join/#asterisk mtaht4 (n=m@239-106-62-200.enitel.net.ni)
16:53.16*** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net)
16:56.29|NexT|Hi, I have 1 TE420B, all the 4 spans are configured in 0 in timing sync (zttest show 100%), but in the messages log, show this: kernel: Zaptel: Master changed to TE4/0/1
16:57.42RyushinWill the Asterisk server have 10000-20000 available to scan?
16:57.56ManxPower|NexT|: what field is that timing option?  1st field, 2nd field, etc
16:58.00ManxPowerRyushin: no idea.
16:58.13ManxPowerbut you need to make sure those ports are not blocked.
16:58.17|NexT|this message is showed only if any of the spans reset
16:58.30RyushinOkay.  I'll log into the cisco router and take a look.
16:58.44|NexT|span=1,0,0,ccs,hdb3
16:58.46davevg-btwtech10000-20000 will only affect rtp, not the initial SIP registration I think
16:59.18|NexT|the second is the timing
16:59.52ManxPowernone of your spans come from the telco?
17:01.15|NexT|all spans come from PSTN
17:01.44ManxPower|NexT|: that is not going to work very well.
17:02.08ManxPoweryou want to get your sync source (timing) from one of the spans.
17:02.49ManxPowerBTW, zttest does NOT show sync timing
17:03.20ManxPowerit also does not test for sync timing
17:05.21|NexT|ok, if i put the span 4 for primary sync and the span 3 for secondary sync, when our span 1 (sync 0) is reset, the message "kernel: Zaptel: Master changed to TE4/0/1" appears again.
17:05.58tru_`z24Is there some zaptel test I can run to test my zaptel hardwarE?
17:06.33ManxPower|NexT|: is it causing problems?
17:06.50ManxPowertru_`z24: several tests depending on what you are trying to test.
17:07.02tru_`z24To make sure zaptel is operating correctly with the hardware
17:07.36tzafrir_home<ManxPower> BTW, zttest does NOT show sync timing
17:07.38*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
17:07.55tzafrir_homeWhat do you mean? It does show the timing from the Zaptel sync master
17:08.06|NexT|I don't know, but my first span resets (red alarm) every 5-10 minutes
17:08.49ManxPowertzafrir_home: as I understand it it tests the jitter in receiving test frames to/from the card.
17:09.06ManxPowerYou should be able to run zttest without a line even plugged into the card.
17:09.23tzafrir_homeManxPower, it simply cmpares ticks it gets from Zaptel to system clock
17:09.25ManxPowerand without a line to get sync from......
17:09.43ManxPowertzafrir_home: so it has nothing to do with the line sync source at all
17:09.47tzafrir_homeBecause the card can also provide ticks. Just like analog cards do
17:10.13tzafrir_homeok
17:10.31ManxPowerzaptel timing and T-1/E-1 timing are two different things.
17:12.37tzafrir_homeManxPower, hmmm... can be. Or can be the same. The zaptel clock is just the ticks of the syncing span
17:13.00ManxPowerIf you say so.
17:13.16flujan[TK]D-Fender: I can achive the events showing using telnet?
17:13.16Nuggettelnet is eeeeeeevil!
17:13.37[TK]D-Fenderflujan: Thats all the TCP connection is effectively.
17:13.50ManxPoweraccording to Juggie on Asterisk-dev, zttest has NOTHING to to with T-1/E-1 timing.
17:13.54ManxPowerIt only tests IRQ stuff.
17:15.42twistedMANBOY
17:16.07*** join/#asterisk saftsack (n=saftsack@pD9E04F92.dip.t-dialin.net)
17:17.02*** join/#asterisk Sci_05 (i=Sci_05@ts.bestserversllc.net)
17:17.08Sci_05afternoon all
17:17.48*** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net)
17:17.53tru_`z24ManxPower: I'm just looking for a test to make sure the card can operate, without having to have asterisk configured to test it, that way i can install zaptel, make sure it passes all tests, then move on to installing asterisk without worrying if i have zaptel configured.
17:19.30*** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be)
17:19.39RyushinIs there a way to get asterisk to log failed registrations for sip?
17:20.52chemikkyes
17:21.56RyushinI couldn't remember what it was.  It was sip set debug.  Sorry for that.
17:23.23*** part/#asterisk ming_zym (n=ming_zym@124.254.56.170)
17:23.27tzafrir_hometru_`z24, there are a bunch of tests in the zaptel directory. I have really no clue what they are for
17:23.48tru_`z24roger.
17:23.58tzafrir_homeIf one of them is useful, please document it...
17:24.07tru_`z24k
17:24.17ManxPowertru_`z24: there isn't a lot you can do.  If zttest works, then the system is talking to the card.
17:24.26ManxPowernot much else you can test without a real connection
17:24.41|NexT|ManxPower, now I change span 1 sync 1, span 2 sync 2, etc..., the span 2-3 show in the intense debug: "Got RR response to our frame" and  "T203 counter expired, sending RR and scheduling T203 again" with two diferent frames, but the first span, show "Unsolicited RR with P/F bit, responding"
17:24.58tru_`z24Real connection ?  Do you mean real application connecting with zaptel, or real connection as in it plugged into a phoneline?
17:25.25tru_`z24This is an analog card, so I can have a phone line plugged in very easily
17:25.25ManxPowertru_`z24: I means sending calls over the line.
17:25.39tru_`z24k
17:25.59tru_`z24I was having problems with zapata.conf, so i wanted to test with zaptel only
17:26.10ManxPower|NexT|: what actual problem are you having?
17:26.12|NexT|the span 1 is one carrier and the 2 to 4 is other carrier, what's the problem?
17:26.16*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
17:26.27tru_`z24If i put the wrong thing in zapata, the zaptel module doesn't load in asterisk
17:26.39*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
17:26.39|NexT|the 1rt span reset every 5-10 minutes with a red alarm
17:26.41ManxPower|NexT|: failed calls, poor audio quality, lockups?
17:26.51ManxPower|NexT|: then call your telco and yell.
17:27.08ManxPowerred alarm = physical cable or line issue
17:27.26|NexT|I called to my telca and put new cabling, reconfiguring with pri_net anf pri_cpe but the problem persist
17:27.43ManxPower|NexT|: then they have to come out and test the line.
17:27.46tzafrir_hometru_`z24, so use genzaptelconf :-)
17:27.57|NexT|I change the t203 timers from -1 to 600
17:28.00ManxPowerfor more than the 1 min they usually test it for.
17:28.14ManxPower|NexT|: you are screwing with T-1 timers?
17:28.34*** join/#asterisk saftsack (n=saftsack@pD9E04F92.dip.t-dialin.net)
17:28.38|NexT|the red alarm is only for 3 seconds
17:28.45|NexT|I'm using E1
17:28.48ManxPowerAnyway, it does not matter.  A red alarm is a physical issue.
17:29.06|NexT|sorry
17:29.24|NexT|Sep 21 19:28:33 veuip2nou kernel: wct4xxp: Setting yellow alarm on span 1
17:29.35|NexT|Sep 21 19:28:38 veuip2nou kernel: wct4xxp: Clearing yellow alarm on span 1
17:29.44ManxPowerYELLOW alarms are different.
17:29.51|NexT|is a yelow alarm ;)
17:29.53ManxPowerThere is 20 mins of my life I will never get back.
17:30.01*** join/#asterisk ming_zym (n=ming_zym@124.254.56.170)
17:30.29ManxPowerdo you now have span 1 set as 1 as the sync/timing source?
17:30.40RyushinThis makes no sense!  Zopier using sip connects just fine to the asterisk server across the VPN.  The Polycom phones don't.
17:31.06*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
17:31.10RyushinThere is not a peep from the polycom phones over sip.  FTP works fine.  SIP works fine using Zoiper.
17:32.02|NexT|yes, now the config is this: span=1,1,0,ccs,hdb3,crc4 | span=2,2,0,ccs,hdb3 | span=3,3,0,ccs,hdb3
17:32.34|NexT|I test the first span with and without crc4 acording our carrier
17:33.32|NexT|I test with resetinterval to never, 600 and 3600
17:36.19*** join/#asterisk katsuodo (n=katsuodo@pool-72-68-117-42.nwrknj.east.verizon.net)
17:36.31|NexT|with my old TE411P, the problem is the same
17:36.59*** join/#asterisk saftsack (n=saftsack@pD9E04F92.dip.t-dialin.net)
17:52.30*** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
17:52.30*** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.4.11 (Aug. 21, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. -=- 1.2 is in security maintenance mode. No non-security related bug fixes will be applied. -=- Going to AstriCon? Join us in #astricon!
17:52.53[TK]D-FenderMACscr: Sort fingers the guilty party directly, now doesn't it?
17:53.35MACscrBut why would the jitter only happen on recordings and not regular talk talkie? =P
17:54.02MACscrWhoops, meant talkie talkie, but eh, whatever, you get my point. =P
17:54.47tru_`z24What is the pseduo zap interface?
17:54.48KwakwaU got a lot of hd activity going on on the machine?
17:55.03tru_`z24Ran zttest, and I have a bunch fo 100%'s and a couple 99%'s
17:55.30[TK]D-FenderMACscr: Whats on the other end?
17:55.44[TK]D-Fendertru_`z24: pseudo = ztdummy
17:55.59MACscr[TK]D-Fender : no idea, it happens with most calls i make from my grandstream
17:56.07tru_`z24k, so does that mean it's not running a test on the real hardware?
17:56.36[TK]D-Fendertru_`z24: what "real hardware"?
17:56.38KwakwaMACscr: Could also be if you're using a different codec, voice might be fine using u/alaw when talking to someone but if you playback a .gsm file it's a bit poor.
17:56.45tru_`z24a x100p clone
17:56.51ZeeekACE hardware sells it
17:56.53*** join/#asterisk lbow (n=lbow@41-195-77-250.access.uunet.co.za)
17:56.55[TK]D-Fendertru_`z24: not sure.
17:57.19[TK]D-FenderMACscr: and that doesn't answer my question at all.
17:57.33Kwakwa:)
17:57.53*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
17:58.14MACscr[TK]D-Fender : could you clarify your question a bit?
17:58.31[TK]D-FenderMACscr: I asked you whats on the other end of those calls....
17:59.36KwakwaMACscr: Make sure the audio you're playing back is in the same codec as codec used for the call.
18:01.36tru_`z24So is it possible to run zttest on /dev/zap/ctl instead of pseudo?
18:02.17MACscr[TK]D-Fender : what do you mean, whats on the other end? Its a recording, could be music, voices, whatever. I have no idea what equipment/software they use
18:04.28[TK]D-FenderMACscr: you don't know whats on the other end of "phone-to-phone" calls?  Can't come up with an answer like "digium digital/analog card", "ITSP using codec XXX", etc?
18:05.11*** join/#asterisk Shido6 (n=shido6@74-130-56-203.dhcp.insightbb.com)
18:07.35*** join/#asterisk scastano (n=scastano@72.165.82.2)
18:08.17scastanocrazy question on IAX2 trunks
18:08.20scastanowho's got some time?!
18:08.21scastanohaha
18:08.25*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)
18:08.28Katty:>
18:08.53KwakwaAsk ur question scastano and if someone has time they'll answer :)
18:09.01Kattyi have a question!!!
18:09.09Kattyhow many ferrets does it take to screw in a lightbulb?
18:09.34Kwakwanone, they bite their owners fingers until they do it
18:09.42Kattylies.
18:09.47rob0None, because weasels have eaten our phone system?
18:09.49Kattyferrets do not bite unless overly wound up and excited.
18:09.56scastanohahahaha
18:09.57Kattyrob0: EXACTLY!
18:09.58Katty<PROTECTED>
18:10.08Kattythey've been carried away by monkies...
18:10.12scastanook... so... i've got IAX trunks setup between 2 boxes....
18:10.21Kattywe've been transfering this at&t rep to that wav file
18:10.24Kattyso fun.
18:10.27scastanocalls from my house to my office always go through......
18:10.35rob0Only a * user would get the humor.
18:10.37scastanocall from the office back out to my house seem to die after a while
18:10.41Kattyrob0: clearly.
18:10.44scastanothe extensions never ring
18:10.45Kattyrob0: now that the rain is gone
18:10.50scastanoit just goes straight to voicemail
18:10.55KwakwaU must only know know nice ferrets Katty, the ones I know will do all kinds of terrible things.
18:11.06ManxPowerscastano: you have trunk=yes?
18:11.11scastanobut the RTP stream is obviously going back and for cause I get my home voicemail recording
18:11.19scastanowhere?
18:11.22ManxPowerscastano: IAX2 does not use RTP>
18:11.24Kwakwaiax.conf
18:11.32scastanowell... I mean.. I see the packets go
18:11.47Kwakwaiax show debug ?
18:11.52scastanothe call comes through, the voice is transmitted, but it doesn't ever try to ring the extensions
18:11.55Kwakwaiax2 show debug rather
18:11.56ManxPowerscastano: well you can't have an iax2 trunk unless you have trunk=yes, unless you were using the term to mean "IAX2 connection", in which case STOP USING THE WRONG TERMS
18:12.01scastanodid that.... didn't see anything funny
18:12.08scastanono no
18:12.11scastanotrunks
18:12.17scastanoone side is asterisk 1.2
18:12.24ManxPoweran IAX2 trunk has a very specific meaning.
18:12.24scastanothe other is trixbox running the same
18:12.26KwakwaHave you set up the right context?
18:12.29scastanoyup
18:12.34ManxPowerscastano: You don't call connections "trunks" in Asterisk
18:12.34Kwakwascastano: are you sure?
18:12.35scastanocalls work... for a while
18:12.38ManxPowernever.  ever.
18:12.46scastanoyes....
18:13.04Kwakwascastano: Do they die or does audio only go 1 way?
18:13.04scastanoif I call 0201 right now at work.. extension 201 rings at home
18:13.11scastanoin about half hour when I do that....
18:13.13scastanono ring
18:13.21scastanobut I do get extension 201's voicemail
18:13.56scastanoand not sure about the audio... when it dies again
18:14.01scastanoI'll try to record a voicemail
18:14.02scastanohaha
18:14.24Kwakwascastano: You're kind of confusing me, "the call comes through, the voice is transmitted, but it doesn't ever try to ring the extensions"
18:14.33scastanoexactly....
18:14.35scastanokinda!
18:14.53ManxPowerscastano: that statement makes no sense.
18:14.56KwakwaIf it isn't ringing the extension, how do you know its not working?
18:15.02scastanomy trixbox says the call is "established"
18:15.07scastanobut.....
18:15.11scastanothe extension never rings
18:15.13ManxPowerscastano: trixbox is not supported here.
18:15.13scastanoI just get voicemail
18:15.16scastanoI know
18:15.20scastanothe trixbox works fine
18:15.21ManxPowerscastano: use the Asterisk CLI
18:15.28scastanoI'm having a problem with the asterisk box
18:15.30*** join/#asterisk katsuodo (n=katsuodo@pool-72-76-11-31.nwrknj.east.verizon.net)
18:15.52ManxPowerso put the CLI output of a failed call on pastebin.ca
18:16.06katsuodoHello Everyone
18:16.07KwakwaManxPower: Is trixbox just a GUI shoved over *?
18:16.17scastanobasically... yes
18:16.38scastanoits asterisk with freepbx preinstalled
18:16.39Kwakwaahh, get in the CLI then.. that'll help u solve the problem
18:16.41scastanobut its the same core
18:16.44scastanoI'm in the CLI
18:16.50ManxPowerKwakwa: no.  Trixbox is a large complex THING that turns Asterisk config files into a maze of spaghetti dialplan, macros, and AGIs.
18:16.50scastanothats where I'm seeing this
18:17.00ManxPoweryou can't even debug the thing on the CLI because it spews out so much crap.
18:17.14Kwakwa:/
18:17.32scastanowell the thing is.... even in a packet capture outside the box
18:17.33ManxPowerscastano: I'm still waiting for that pastebin
18:17.43scastanowhat do you want a pastbin of
18:17.43scastano?
18:17.50scastanofrom which side?
18:17.52ManxPowerTHE CLI OUTPUT OF A FAILED CALL.
18:17.58scastanothats the thing
18:18.03scastanothe call doesn't "fail"
18:18.05pjzanyone have recommendations for a good linux SIP softphone?
18:18.06tzafrir_homeKwakwa, no. Trixbox is a PBX built on top of Asterisk and other stuff.
18:18.11ManxPowerscastano: if you paste the tricbox CLI I'll feed you to the aligators.
18:18.13scastanothe cli says its going fine
18:18.16KwakwaCapturing packets etc... seems a bit over kill mate :)
18:18.22scastanohaha
18:18.24katsuodoI inherited a asterisk 1.2 box with tdm400p card and trying to dial from the console and I am receiving no such extension '4004' in context local first time introduce to asterisk
18:18.28ManxPowerscastano: *sigh*  Did it occur to you that I might see something you might miss?
18:18.32tzafrir_homeAsterisk is more of a toolkit for building PBX systems than a PBX
18:18.34scastanovery possible
18:18.44scastanohang on... I'll see what I can grab
18:19.01scastanogo figure... its working right now
18:19.13ManxPowerscastano: sounds like a NAT problem to me.
18:19.22scastanothats exatly what I thought
18:19.22katsuodoAny suggestions?  I am also reading the asterisk blue and white asterisk bible
18:19.37GoRKDoes anyone have Polycom's Technical Bulletin 25751 that details the SRTP configuration options for polycom phones?
18:19.41Kwakwascastano: I think your ambiguity helped heal it :)
18:19.46ManxPowerand you know what the standard fix for random failed calls in a nat enviroment, right?
18:19.47scastanobut I enabled DMZ setting on my linksys at home
18:19.57scastanoManxPower: no idea?
18:20.18ManxPowerscastano: qualify=yes will keep enough traffic on the link to keep the router from closing the NAT translation
18:20.24scastanogot that
18:20.37ManxPowerthen I have no more suggetions.
18:20.41scastanoeven dropped qualifyfreqok=5000 n there
18:20.59scastanojust to ensure the nat stays open
18:21.08*** join/#asterisk drwelby (n=mpfister@mail.enplan.com)
18:21.13scastanoand like I said... the stream gets from one place to another... I hear the voicemail
18:21.20scastanobut the extension doesn't ring
18:21.33scastanoits almost like its not "ACK"ing
18:22.55Kwakwaand I'm assuming u also have nat=yes in the config?
18:23.23scastanono
18:23.25scastano:(
18:24.41Kwakwa:/
18:25.26scastanobut without it... it works
18:25.31scastanoI'm gonna through it in there anyway
18:25.53KwakwaI'd like to think there was method to adding that option :p
18:26.13MACscr[TK]D-Fender : i am using a SIP to PTSN provider (callcentric) for making my calls. When you say other end, I am thinking about who receives the call. Which is why I said I have no idea what their equipment is.
18:26.16scastanoI didn't know it needed it with IAX2
18:26.19scastanoI thought that was the point
18:26.51[TK]D-FenderMACscr: ok, how is GXP -> PAP2?  PAP2 = fine all the time?
18:26.54ManxPoweryou can add nat=yes to the iax.conf, but it is not a valid option and will be ignored.
18:27.51*** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur)
18:27.53scastanothats what I thought
18:27.59scastanoIAX2 doesn't care about nats
18:28.01Kwakwahaha, good point :b
18:28.35KwakwaI gave up with IAX2 in favour of SIP because of multiple call issues, think I'm mixing the two in my head *hangs head in shame*
18:29.18MACscr[TK]D-Fender : my pap2 is just another extension of mine. I meant that its fine when i make calls with it. Calling from GXP to PAP2 is not going to result in any recordings, thus i could not test anything like that
18:30.02[TK]D-FenderMACscr: ok, your description is getting spotty.  is it ONLY GXP -> * prompts that is the problem?
18:32.11*** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br)
18:33.45MACscr[TK]D-Fender : ok, obviously my GXP is using my * to make any calls, thus why I am in this channel. When I call a business with my GXP, which would obviously be GXP -> * -> CallCentric -> Business (no idea what they are using, everyone would be different). Most businesses have some type of recording whe you call them or are on hold. I get a lot of "crackle" (poort quality) during these recordings. If i would call with a regular (ptsn) phone or even my pap2
18:33.45MACscrmy * box), i do not get this type of poor quality.
18:35.20ManxPowerThe SIPura line of devices frequently default to 0.30 (30ms) voice packet size.  Asterisk uses 0.20 (20ms).  Change the SIPura box to use 20ms packet size.
18:35.51ManxPowerperhaps the GXP (may it rot in hell) has the same issue.
18:37.17[TK]D-FenderMACscr: so basically the GXP is shitty on all calls then?
18:37.46MACscr[TK]D-Fender : only on recordings, thats why its odd
18:38.18MACscrBut your right, the GXP is a shitty phoen in general =P
18:38.22[TK]D-FenderAudio is audio at that point... the GXP blows.  Period
18:38.41*** join/#asterisk adorah (n=Michael@87.69.57.246.cable.012.net.il)
18:40.27*** join/#asterisk atomicd (n=AtomicDa@74-206-0-80.static-ip.m.telepacific.net)
18:43.32*** join/#asterisk GaryH (n=chatzill@wallace.garysoft.co.uk)
18:51.22*** join/#asterisk GreggB (n=GreggB@66.206.86.107)
18:53.19atomicdMy boss says I get to go to Astricon! :-)
18:55.15the_laleluatomicd: your boss should talk to my boss. ;)
18:56.18atomicdI'm going on the cheap though...  My Dad live in Scottsdale so I don't need a hotel.  Plus I'm driving from Anaheim, CA.
18:56.23*** join/#asterisk funxion (n=nunya@63.214.236.169)
18:59.23the_laleluatomicd: well, i'm from hamburg, germany. *g*
18:59.58atomicdthe_lalelu:  That would be a long drive... :-)
19:00.19*** join/#asterisk Yourname` (n=IM@unaffiliated/yourname/x-837320)
19:00.20hmmhesaysejabberd is driving me insane
19:00.31the_lalelui guess your right. :D
19:00.46moghmmhesays, whats wrong with ejabberd
19:00.52*** join/#asterisk lbow (n=lbow@41-195-77-250.access.uunet.co.za)
19:02.16hmmhesayswell I can't get my registration to work
19:02.48mogdo you have it allowed in your acl?
19:03.14hmmhesaysI have everything open, but it seems to be trying to use ssl to auth my client, and I can't find in my config file where it says to do that
19:04.12hmmhesays{s2s_use_starttls, false}.
19:05.39hmmhesaysalso where you configure the listen ports, I if I comment out the tls lines and uncomment the "use these if no tls support"  I get no log files at all
19:11.22*** join/#asterisk PorkSale (n=barney@c-98-202-51-107.hsd1.ut.comcast.net)
19:13.07PorkSaleI just learned about asterisk and I'm interested in setting up a system for up to 8 lines.  I'm trying to figure out what an FXS and FXO module is for.
19:13.59hmmhesaysyou plug a phone in to fxs modules and you plug a phone line into fxo modules
19:14.50jsmith(And then configure the signalling just opposite of that)
19:15.48PorkSalehttp://www.digium.com/en/wheretobuy/digiumdirect/productview.php?product_code=1TDM844BF-01
19:16.07PorkSaleso that would allow me to have 4 lines connected for incoming or outgoing calls?
19:17.14PorkSalemaybe I don't understand asterisk as I thought.  Do you need lines going into the fxo modules if it is all voip?
19:17.27jsmithNope... the FXO ports are just for connecting to analog lines
19:17.36jsmithIf you're doing all VoIP, you don't need any cards
19:17.46PorkSaleok, so I would just get the 8 port with 2 quad fxs modules
19:18.04jsmithIf you do have a card though, then you can mix/match/bridge calls across VoIP and analog
19:19.44PorkSaleok, but if I just have the fxs module I can still mix/match/bridge calls all on the VoIP right?  Like I can forward a call to another location though the VoIP
19:20.24PorkSalejust not from VoIP to analog lines I've got coming in
19:20.30[TK]D-FenderAt 8 channels I would suggest looking into a partial PRI
19:21.58pots_lineAnyone ever use isymphony?
19:22.09pots_lineop panel
19:23.12jsmithPoincare: Exactly
19:23.23jsmith[TK]D-Fender: His eight ports would be FXS ports, not FXO ports
19:23.36jsmith[TK]D-Fender: Oh, and a temporary asteriskdocs.org is back up
19:23.48jsmith(while I re-design the site, again)
19:23.52[TK]D-Fenderjsmith: Enough that all the links work?
19:23.56jsmithYes
19:24.04[TK]D-Fenderok, I'll de-list mine then
19:24.45jsmithNo, that's fine
19:24.55[TK]D-Fendermirror*
19:25.03*** join/#asterisk tomcats (n=fgonzale@189.157.152.170)
19:25.44*** join/#asterisk zeromobile (n=zero@64.78.21.135)
19:26.11tomcatsWhat could cause asterisk to only enable sound coming from the agent to the caller, meaning that the caller CAN hear the agent but not the other way?
19:26.43WonkaNAT?
19:27.01[TK]D-Fenderentirely
19:27.08tomcatsohh and this only happens sometimes.. like 1 out of 10 calls the agent recieves...
19:27.15zerohaloDID provider issue?
19:27.45tomcatswe checke dthe local loop T1 acces and they say that everything is fine...
19:28.03*** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)
19:28.25tomcatsa full log doesn't show any errors too...
19:28.34*** join/#asterisk jtoy (n=jtoy@mail.backchannelmedia.com)
19:28.46jsmithtomcats: Did you check the output of "rtp debug"?
19:28.51jtoycan sending emails of voicemails be turned on/off through the manager?
19:29.01*** part/#asterisk dasuberdavid (i=david@nat/digium/x-e6a5d9c3adf33b2d)
19:29.11tomcatswhat's rtp debug?
19:29.13jsmithtomcats: That'll tell you one way or the other if RTP is getting through (assuming this is SIP, of course)
19:29.39jsmithjtoy: Well, yes, but it's convoluted.  You'd have to use the UpdateConfig action to rewrite voicemail.conf
19:29.52jsmithjtoy: So it's *possible*, just not so easy
19:30.09jtoyis that way safe? is there a better solution?
19:31.07*** part/#asterisk zeromobile (n=zero@64.78.21.135)
19:31.24tomcatsjsmith: ohh interesting.. is there a way to filter rtp debug output?
19:31.53jsmithtomcats: Not really...
19:32.02jsmithjtoy: It's safe
19:32.16jtoyis realtime voicemail a better option?
19:33.11*** join/#asterisk defsmac (n=andy@defsdoor.gotadsl.co.uk)
19:33.22PorkSaleare there any modules or options that'll send out a transcript of your voicemail to your email?
19:33.39defsmacPorkSale, trixbox does that out of the box
19:33.57defsmacso you should be able to find out how
19:34.20defsmacpart of freepbx I guess
19:34.33*** join/#asterisk r00tlz (n=Cero@190.41.12.173)
19:34.56PorkSaleis trixbox built on top of asterisk or they're totally seperate?
19:35.08defsmacits a distribution
19:35.09scastanodefsamc: I don't think it does it out of the box, there's a few modules to figure
19:35.31scastanoand trixbox is CentOS with asterisk and freepbx already running on it
19:35.34scastanowith a few other goodies
19:35.44scastanolike hylafax and asterisk recording interface
19:35.56defsmacok - by transcript do you mean voice recognition to text ? if so - then no  :)
19:36.14pepsehas asterisk gui had any improvements?
19:36.21pepselike, is it fully functional yet
19:36.38*** join/#asterisk ivrc (n=chatzill@74.228.54.150)
19:36.40tomcatsjsmith: rtp debug show packets in and out... any other idea?
19:37.25pepseit made some -craaazy- configs when i tried it.. but it's problem was some bugs that made a lot of stuff not work
19:38.56jsmithtomcats: If it shows packets coming and going, then everything is fine on Asterisk's side... is there a firewall between the Asterisk box and the phone that might be blocking the audio?
19:40.43*** join/#asterisk blinkbot2k (n=me@c-75-69-77-42.hsd1.vt.comcast.net)
19:42.26tomcatsjsmith: I am suspecting is not a connection problem since the problem happens randomly.. I am having trouble to replicate the issue, althought the problem keeps happening...
19:45.13PorkSaleAny idea what a partial PRI costs?  I'm looking at hosting our service through vitelity.  Our phones aren't really that busy most of the time so I'm thinking it'll be less than $200 per month total to do it this way.
19:45.55jsmithPorkSale: It all depends on your location and your telco
19:47.15scastanoVitelitys prices are hard to beat
19:47.26scastanoI'm in DC in a really well lit area
19:47.40scastanoa partial PRI for me, 12 channels is around 280
19:49.27scastanoand also... I used Vitelity here for rollovers when my 2 PRI's fill up
19:49.43*** join/#asterisk thunter (n=Tee@rawb.fttp.xmission.com)
19:49.57*** join/#asterisk Defraz (n=t0tal@208.98.184.140)
19:50.36thunterhow do I send a # key to a caller instead of # acting like the transfer key?
19:52.09*** join/#asterisk r00tlz (n=Cero@190.41.12.173)
19:54.49ManxPowerthunter: don't put t/T/w/W on the Dial lone
19:54.49defsmacscastano, a month ?
19:54.49ManxPowerlne
19:55.23ManxPowerMy last quote for a 11 channel PRI was $800/month
19:55.32defsmacwtf ?
19:55.41defsmacin uk it a tenner a channel
19:55.50[TK]D-FenderManxPower: Kinda sucky
19:55.51defsmacso that would be 80 a month
19:55.59scastanoand yes... per month
19:56.05defsmacminimum 8 channels
19:56.18ManxPowerdefsmac: I've never heard of PRI that cheap.
19:56.45ManxPowerGenerally PRIs are priced based on the loop + channels.
19:57.14JerJerManxPower:  unless the loop is a few feet long   :D
19:57.32JerJerand your workin with a friendly xLEC
19:57.43defsmacmaybe I'm misunderstanding what you are meaning by PRI ?
19:57.50Wonkawtf? we paid about 400EUR/month for a PRI with 30 B-channels
19:57.58ManxPowerdefsmac: T-1 or E-1 PRI
19:58.22scastanoWonka: where the hell are you?!
19:58.22ManxPowerWonka: Prices vary by quite a bit.
19:58.24JerJerback a year or so ago we could get real T-1 PRI loops for $125 a month  - not any more though  :(
19:58.26scastanoI wanna live there! :-P
19:58.28Wonkascastano: northern germany
19:58.28defsmacManxPower, well an E1 in the uk is about a 10 a channel - minimum of 8
19:58.43ManxPowerdefsmac: no loop charge?
19:58.44scastanoE1 is actually 30 channels
19:58.49scastanoT1 is 24
19:58.58scastanoeach way you loose one channel for the D channel
19:59.09defsmacManxPower, not that I am aware of
19:59.10ManxPowerJerJer: there no facilities CLECs in my area
19:59.20[TK]D-FenderE1=31, T1=24, E1 PRI=30, T1 PRI=23 <---
19:59.30Wonkascastano: E1 is 32 timeslots. one ist used for sync, one for D channel, leaves 30 for B channels
19:59.34scastanodoh
19:59.35scastanoyou're right
19:59.39scastanoI was off by 1
19:59.40scastanohahaha
19:59.56defsmacD channel is channel 16 on BT
20:00.08defsmacI learned that the hard way on my first install
20:00.15JerJerManxPower:  find a couple mil and start one
20:00.18JerJer....or not
20:00.31ManxPowerJerJer: Only a couple of mil?
20:00.47scastanoManxPower: I've get $8 man... .I'm in
20:00.52scastanobut I wanna be a partner!
20:00.53scastano:-P
20:00.58JerJerif you had a market a couple mil would be a start
20:01.16[TK]D-Fenderscastano: You'll be a very small "part" of PARTner ;)
20:01.58scastanohahahahaha
20:02.05ManxPowerJerJer: I shutdown my micro isp, telco, cableco about a month ago.
20:02.34ManxPower"irreconcilable differences"
20:03.06J4k3wtf, cable? :)
20:03.33ManxPowerJ4k3: *nod*
20:03.46JerJerManxPower: that sucks
20:03.55J4k3but then I thought better of it
20:04.04J4k3ManxPower: buy an old cable modem head end and just back-feed your node? :)
20:04.11J4k3err back-fed?
20:04.19ManxPowerJ4k3: I would need a NOC first.
20:04.23JerJeri've got about 9 customer radios online here in my neck of the woods
20:04.35ManxPowerthose "irreconcilable differences" means I don't have a NOC anymore
20:04.41J4k3I've got around ~70 subs
20:05.13JerJer3 of which are family (but they still pay :)
20:05.15ManxPowerI'll revisit the idea in 18 - 24 months
20:05.39J4k3ManxPower: my "noc" is a wide hallway in my home, with a water heater closet (2Mx2M) holding most of the gear
20:05.55J4k3and a 45m (150') tower in my side yard.
20:06.51J4k3well, I call it a 'wide hallway'... its got open doorways on both ends feeding other rooms...  its maybe 5Mx6M
20:06.57scastanohahaha
20:10.31JerJermy 'NOC'  is an equipment box mounted to a 100' tower out back, on the big hill
20:10.38JerJerpowered by solar  :)
20:13.44hmmhesaysI got the msn transport working but this stupid client doesn't support nicknames
20:14.37*** join/#asterisk lbow (n=lbow@41-195-77-250.access.uunet.co.za)
20:15.55*** part/#asterisk thunter (n=Tee@rawb.fttp.xmission.com)
20:19.06FXOLanyone alive that might have some ideas on a problem with a service provider?
20:19.34FXOLWe were using an IAX trunk to connect to them... and calls always connected, but with alot of breakup in the call
20:19.50FXOLThey had us switch to SIP, and call quality was pefect
20:20.05FXOLhowever... system keeps acting like connection drops betweeen us
20:20.09FXOLoutbound calls wont work
20:20.16FXOLand inbound calls just ring and don't hit *
20:20.26FXOLit's Random tho when it works
20:22.10[TK]D-FenderFXOL: sounsd like a NAT problem.
20:22.25FXOLwhy would it work 50/50 tho?
20:23.35tru_`z24when i do zap show channels at the asterisk cli, it only shows the pseudo interface....
20:23.59ManxPowertru_`z24: then you don't have any channels configured
20:24.15ManxPowerFXOL: because nat routers close
20:24.21ManxPower"inactive" connections
20:24.25FXOLintersting
20:24.32FXOLso how might this get resolved?  And why only w/ SIP?
20:24.33tru_`z24I have the channel configured in zaptel, so let me see what is going on in the zapata.conf then
20:25.05ManxPowertru_`z24: /etc/zaptel.conf is the CARD config file  /etc/asterisk/zapata.conf is the ASTERISK config file.
20:25.13tru_`z24right, the card config file is right
20:25.23ManxPowertru_`z24: the asterisk config is not
20:25.30[TK]D-FenderFXOL: ...
20:25.31[TK]D-Fender~sipnat
20:25.32jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3  otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
20:25.33[TK]D-Fender^^^^^^^^^^^^^^^^
20:25.38ManxPowerFXOL: IAX is more chatty than SIP and could easily keep the NAT reanslation active.
20:25.42tru_`z24right, is there a similar configuration program like genzaptelconf ?
20:25.54ManxPowertru_`z24: I've never used that program.
20:25.59ManxPowerconfig them manually
20:29.22*** join/#asterisk Schumie (i=SteveWri@cpc2-rdng2-0-0-cust382.winn.cable.ntl.com)
20:29.32FXOL(ManxPower): I assume this would still be a problem if I put on a DMZ?
20:29.51ManxPowerFXOL: that would totally depend on your router.
20:30.04FXOLNetgear WR614 :P
20:30.12FXOLWRG614 I mean
20:30.13FXOL:P
20:30.17ManxPowerI'm not a magical router fairy.
20:30.21FXOLI know ;P
20:31.02tru_`z24Got it now :-)
20:32.19tzafrir_hometru_`z24, zapconf :-)
20:34.56*** join/#asterisk Cyon (n=cyon@216.179.31.170)
20:35.15tru_`z24well, genzaptelconf generated a zapata.conf.channels file
20:35.23tru_`z24i just appended it to my current zapata.conf file and it works
20:35.42*** join/#asterisk ivanfm (n=ivanfm@c906b486.virtua.com.br)
20:36.41tzafrir_homezapata-channels.conf , but yes
20:38.21*** join/#asterisk guillote_GNU (n=bancaria@host35.201-253-17.telecom.net.ar)
20:42.45lirakisnight all
20:42.47*** part/#asterisk lirakis (n=lirakis@65.200.191.253)
20:43.59*** join/#asterisk copantl (n=copantl@63.161.232.126)
20:44.24copantlany body know howto install phpagi on asterisk?
20:45.07ManxPowerit doesn't come with any docs?
20:46.59*** join/#asterisk inv_arp[work] (n=junya@c-76-26-24-197.hsd1.fl.comcast.net)
20:47.24copantli saw it but i dont see a place to install it
20:47.32copantlis not a module?
20:47.47ManxPowerIT is not an asterisk module.
20:47.54ManxPowerI dunno if it is a PHP "module" or not.
20:48.30*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
20:48.39copantlbut i dont see any make install or something like
20:50.14*** join/#asterisk jimmysolis (n=jimmy@190.41.82.1)
20:51.15jimmysolisHello i cant compile the zaptel 1.4.5 =(
20:51.24jimmysoliswho know that: make: *** No rule to make target `install-inlcude', needed by `install-programs'.  Stop.
20:52.00*** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
20:52.55[TK]D-Fenderjimmysolis, there is a typo in the makefile.
20:53.17[TK]D-Fenderjimmysolis, install-inlcude should be install-include'
20:53.33[TK]D-Fenderjimmysolis, load it up in a text editor and fix it and "make install" should work
20:53.53jimmysolisi use debian and i dont found this package "install-include"
20:54.18[TK]D-Fenderjimmysolis, there is nothing to find.
20:54.24jimmysolisok
20:54.35[TK]D-Fenderjimmysolis, there is an ERROR in the make file.  a simple tex-edit corrects this
20:54.42[TK]D-Fendertext-edit
20:54.46jimmysolissorry but i dont understand english very well
20:54.49jimmysolisGracias =)
20:55.00[TK]D-Fenderjimmysolis, hopefully you just "got it"
20:55.26*** join/#asterisk dlynes_laptop (n=dlynes@s142-179-114-141.bc.hsia.telus.net)
20:56.21dlynes_laptopIs there a way to make it so that when asterisk picks a line from a zaptel channelgroup to call out on, that it starts picking from line 8, instead of line 1, on an 8 line card?
20:56.46tzafrir_homejimmysolis, right. Get 1.4.5.1
20:56.46dlynes_laptopThat way the user isn't picking up a ringing line when they try to dial out?
20:56.47jsmithdlynes_laptop: Use a capital G
20:56.55dlynes_laptopjsmith: ah, ok
20:56.56jsmithdlynes_laptop: Zap/G1 instead of Zap/g1
20:56.58dlynes_laptopjsmith: thanks
20:57.03jsmithdlynes_laptop: No problem
20:57.05tzafrir_homeIt fixed Makefile typos and such
20:57.51dlynes_laptopjsmith: didn't even know the feature existed...thought 'G' and 'g' were the same thing :)
20:58.14tzafrir_homedlynes_laptop, or you can use r or R for round-robin
20:58.22jimmysolisthanks guys now work :)
20:59.03jsmithWork?
20:59.19dlynes_laptoptzafrir_home: what would round robin be good for?
20:59.35jsmithjimmysolis: Trabajo?  No sabemos trabejar!
21:00.12tzafrir_homeMake things less predictable? Make sure all channels are used?
21:01.21jimmysolisaka si sabemos trabajar nos negrean jajaja
21:01.54*** part/#asterisk zerohalo (n=zeroHalo@pool-72-70-79-233.bstnma.east.verizon.net)
21:03.28Kattyoh
21:03.30Kattyi'm still in here
21:03.35Kattycrazy!!
21:03.39Kattyalso, hi
21:04.39hmmhesaysheh
21:04.40hmmhesaysthat is crazy
21:05.09[TK]D-FenderKatty, Mew.
21:06.12dlynes_laptoptzafrir_home: but if you have calls coming in from line 1 up, and calls going out from line 8 down, does that not use the lines up?
21:06.30*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
21:06.40hmmhesaysI really wish the jabber transports had voice
21:06.57dlynes_laptoptzafrir: I mean I can see round robin for call agents, but that's for extensions, not lines
21:08.18jsmithdlynes_laptop: It's so that if you have a bad channel, your calls don't keep getting stuck on that one bad channel
21:08.18Katty[TK]D-Fender: mew
21:08.24Katty[TK]D-Fender: started a shiny new website.
21:08.56Katty[TK]D-Fender: with ampache, and gallery2, and phpbb
21:09.07[TK]D-Fenderdlynes_laptop, That almost never matters.  any telco hunt group will work around whichever channels are in use and pick a free one.
21:09.14*** join/#asterisk amarzouk2 (n=chatzill@217.54.201.152)
21:09.22Katty[TK]D-Fender: it r purrty.
21:09.24[TK]D-FenderKatty, oooohhh
21:09.30[TK]D-FenderKatty, Link meh!
21:09.37Kattyk
21:10.30Wonkapurrty? can has pusseh?
21:10.41Katty^_-
21:11.04Wonkar cute!
21:11.08[TK]D-FenderWonka, Sure thing.... "Willy" :p
21:11.13Kattyoh.
21:11.25Kattyi pulled that from a little picture of a kitty that said 'i r not squeeze toy'
21:11.51Kattyhttp://icanhascheezburger.files.wordpress.com/2007/04/i-r-not-squeezy-toy.jpg
21:11.56Wonkaicanhascheezburger.com has many of these
21:11.56Kattyand so i've been using that ever since :P
21:11.58Wonka:)
21:12.32Katty(=
21:12.39amarzouk2Hi, I am having a problem with get_data , it does not accurately get the input digits! what can I do to insure more accuracy?
21:12.51*** join/#asterisk metfan2007 (n=metfan20@189.135.156.38)
21:13.23Kattyamarzouk2: threaten it with a stick.
21:13.32Kattyamarzouk2: and tell it no pudding!
21:13.58amarzouk2:) tried that already no use :(
21:14.22Katty:<
21:15.36Wonkaand no jelly donut either.
21:16.33atomicdAs I'm registering for Astricon, there's a field labeled:  "Enter Discount Code"  Anyone know of a discount code?  I'm always up for saving my company a couple of bucks...
21:17.17*** join/#asterisk h0 (n=Fakhir@unaffiliated/fakhir)
21:17.26Qwellatomicd: check digium.com, I think there's one on the front page...or was?
21:17.56BadPacketDigium-Astricon-2007 or VoicePulse-Astricon-2007
21:18.33Qwellhttp://www.digium.com/en/mediacenter/events/viewevent/55
21:18.35Qwellyeah
21:18.55atomicdYou guys rock...thanks!
21:19.24*** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr)
21:22.16tomcatswhat could cause rtp packets to be able to get from the phone to the pbx but not the other way arround on a LAN setup?
21:22.43atomicdJust saved me $110 (actually, my company) (20% of $550)
21:22.59dlynes_laptop[TK]D-Fender: ah....well, the whole reasonn i'm doing this is so that when someone picks up a phone, and then dials a number, they're not connected to a call that happens to come in at the same time they finish typing in their phone number
21:23.13dlynes_laptop[TK]D-Fender: and ultimately dialing hte phone number in that person's ear
21:23.32[TK]D-Fenderdlynes_laptop, then yeah your best odds are ascending incoming, decending outgoing
21:23.48dlynes_laptop[TK]D-Fender: exactly...that's the way the nortel phones usually do it, too
21:25.32metfan2007hi All, is there any way to add more G729 licenses to a Asterisk that already has a few licenses registered???
21:26.02Qwellmetfan2007: yes, just drop in another license file, and it'll pick it up
21:26.16*** part/#asterisk jtoy (n=jtoy@mail.backchannelmedia.com)
21:26.22Qwellatomicd: so, that means you're buying us beer?  on the company, of course :p
21:26.43dlynes_laptopmetfan2007: same way
21:26.53atomicdQwell:  Drinks are on Reliant Manufacturing!
21:27.00metfan2007Qwell: drop?? there? :S do you mean to run againt the registration utility?
21:27.17Qwellmetfan2007: well, you get the new license (by running register), and it puts it where it needs to be
21:27.26Qwellatomicd: I am, of course, only joking :)
21:27.40perdwhere does the voicemailmain app store the passwords if a user changes it?
21:27.44metfan2007thanks!
21:28.18putnopvutperd: in the voicemail.conf file
21:28.27perdreally.. hrm
21:28.34putnopvutor users.conf if that's where you have your mailboxes defined.
21:28.41perdthe passwords in there dont work, and i have no users.conf
21:31.09perdhaha oh, i'm retarded is why..
21:31.43[TK]D-Fenderperd, the first step is admitting you have a problem....
21:31.52perdi sure do have a lot of them.
21:32.13theHub/bye Have a nice weekend, everyone.
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21:34.52_Sam--data23:  i had a very pleasant experience with ebay pandora purchase -- worked mint, shipped quick.  no problems.
21:34.54_Sam--er
21:35.00_Sam--wrong win
21:35.48[TK]D-Fender_Sam--, All stable?
21:36.11_Sam--so far so good, thank you for checking.  System uptime: 1 day, 1 hour, 56 minutes, 18 seconds
21:38.26[TK]D-Fender_Sam--, great...
21:39.03_Sam--thank you again for your help.
21:39.47*** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus)
21:39.51[TK]D-Fender_Sam--, np
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21:56.53riddleboxhrmm is there a way to get Asterisk to detect a hangup from a TDM fxo? right now if someone hangs up, it rings the full 20 sec I set then goes to voicemail?
21:57.24RyushinAre there alternative voices for Asterisk beside the girl that can be bought/downloaded?
21:58.34*** join/#asterisk zeromobile (n=zero@64.78.21.135)
21:59.20fakhirRyushin, there are two on the digium site -> http://www.digium.com/en/products/voice/
22:00.39jsmithriddlebox: Yes, if your line has remote disconnect supervision on it
22:01.03jsmithriddlebox: If it does, the telco will temporarily open the loop within about 6 or 7 seconds of the remote party hanging up
22:01.03riddleboxjsmith, right now I doubt it, as I have a line provided by the cable company
22:01.24jsmithRyushin: Yes... check the wiki
22:01.46Kattyi guess he's getting fast food
22:04.24hmmhesaysfast food, fast heart attack?
22:05.19Kattyclearly
22:07.08perdmmm vmail.cgi is nice
22:07.18perdjust thought i'd throw that out there.
22:13.39*** join/#asterisk barrys (n=barrys@ool-4577407d.dyn.optonline.net)
22:14.49*** part/#asterisk barrys (n=barrys@ool-4577407d.dyn.optonline.net)
22:15.27hmmhesaysi'll throw you out there
22:15.35*** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br)
22:15.43perdyou're too kind!
22:17.03*** join/#asterisk Zipper_32 (n=None@d205-250-2-107.bchsia.telus.net)
22:20.33tomcatsfor some reason app_queue is bridging the call halfways with the agent.. sound only comes from the agent but not from the caller... any ideas?
22:26.51*** join/#asterisk disgrntld81 (n=asdf@CPE-75-81-155-105.wi.res.rr.com)
22:27.59disgrntld81total newb needs direction...  i want asterisk automatically add 1 + area code if i only dial 7 digits, what should i google to find out?
22:28.56gremzoidextensions.conf variables...
22:29.03*** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)
22:29.04disgrntld81sweet, thanks
22:29.04gremzoid1${EXTEN} ?
22:29.14perdGotoIf
22:29.29*** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net)
22:29.32gremzoidyea that to
22:29.52perdif it's 7 chars, do this, otherwise, do that
22:30.02perdthere's probably a better way to do it but i'm no guru
22:30.12tzafrir_homeRyushin, http://www.voip-info.org/wiki/view/Asterisk+sound+files+international
22:30.20disgrntld81great, thanks!
22:31.07perdhttp://www.voip-info.org/wiki-Asterisk+cmd+GotoIf disgrntl
22:32.24disgrntld81oh cool, good link
22:32.41RyushinThanks for the links.  Thats going to get me started.  I appreciate it.
22:36.38riddleboxjsmith, my telco isnt providing remote disconnect
22:36.53GreggBRyushin: there's a company giving out sound files.
22:38.10*** join/#asterisk Strom_M (n=strom@208.127.172.112)
22:38.25GreggBRyushin: Ahh, found them. http://www.voicevector.com/  These folks are great to work with, and you can get an entire voice pack for free from them (checkout the link on their homepage)
22:38.55GreggBRyushin: We've been happily using the voice pack since early this year...
22:39.31perdhas anyone here integrated asterisk with a ccm 3.3 server, i have a quick dumb question if you dont mind
22:39.56sevardEverytime somebody mentions a "new service" they've been using for "quite a while" and been "pretty happy with it" i always assume they're the CEO pushing their drabble on IRC.
22:40.12perdhaha
22:40.41gremzoideww pommy voices
22:44.53*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)
22:46.31GreggBsevard: you talking about me? heh, no, I'm a lowly SysAdmin running an Asterisk box for my employer (who is a medical records management company).... now that I think of it, the voice pak did have one issue; The American Female Asterisk voice pack's "vm-options" file reads an incorrect list of options (apparently for *1.2). If anyone wants to hear more about this, just lemme know - otherwise, just email Robert Christian (robert@voicevector.com), and he sen
22:49.01sevardhe sen
22:49.09sevardt me a packet of cookie chrisps for christmas
22:50.00*** join/#asterisk thieums (n=Mathieu@rny93-4-82-231-54-139.fbx.proxad.net)
22:50.06GreggBI dont think you can add those to the wiki
22:55.49hmmhesaysnew avenged sevenfold song is pretty good
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23:01.18*** join/#asterisk alephcom (n=darren@h66-112-187-16.mcsnet.ca)
23:01.27alephcomgreetings
23:01.54alephcomAnybody aware of an ftp server that will accept mixed case usernames?
23:02.04alephcomI'm trying to use the default usernames in the polycoms.
23:03.18[TK]D-Fenderalephcom, I'd bet vsftpd does as well as proftpd
23:03.26hmmhesaysI bet most do
23:03.27Qwellany linux ftpd, no doubt
23:03.39alephcomThat's what I thought to be it doesn't appear that way.
23:03.41hmmhesaysand probably most windows ftpd's
23:03.44alephcomMaybe I have something screwed up.
23:03.58Qwelland if it doesn't accept them with mixed case, it'll just lowercase everything anyways
23:06.23dlynes_laptopIf I have say ten extensions, and each extension has eight accounts, is there an efficient way to pick a free account on every phone, when a call comes in?
23:06.52[TK]D-Fenderdlynes_laptop, how can an "extension" have an "account"?
23:06.55dlynes_laptopI'm finding IsChanAvail() and other means are either not reliable, or they're too slow
23:07.10dlynes_laptop[TK]D-Fender: each 'line' appearance on the extension is a separate account
23:07.39[TK]D-Fenderdlynes_laptop, are they intended to be completely different identities?
23:08.04[TK]D-Fenderdlynes_laptop, And stop calling a PHONE an EXTENSION.
23:09.13alephcomThanks, I'll snoop around and try a few more things.
23:09.22dlynes_laptop[TK]D-Fender: no...just different 'lines' on the same phone
23:10.05[TK]D-Fenderdlynes_laptop, please clarify your dangerously vague terminology.....
23:10.41dlynes_laptop[TK]D-Fender: I've got a call coming in, I need to send it to all ten phones, whether those phones are on a call, or not
23:11.09perdterror alert!
23:11.18dlynes_laptop[TK]D-Fender: each phone is capable of handling up to nine simultaneous calls, but I only have 8 peers and users defined for each
23:11.34[TK]D-Fenderdlynes_laptop, what models?
23:11.37*** join/#asterisk zeromobile (n=zero@64.78.21.135)
23:11.39dlynes_laptop[TK]D-Fender: 9133i's
23:11.42*** join/#asterisk metfan2007 (n=metfan20@189.135.156.38)
23:11.44dlynes_laptop[TK]D-Fender: Aastra's
23:12.05[TK]D-Fenderdlynes_laptop, And why all running multiple regs on the same phone if they are not infact "unique"?
23:14.00dlynes_laptop[TK]D-Fender: how else can I handle eight simultaneous calls on one phone?
23:14.24[TK]D-Fenderdlynes_laptop, 1 reg spanning all keys.  they should not be run as sperate reg's
23:14.33*** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net)
23:14.50dlynes_laptop[TK]D-Fender: I remember trying that a year ago or so on Aastra's, and it never worked
23:14.55dlynes_laptop[TK]D-Fender: I've never tried it since
23:15.14dlynes_laptop[TK]D-Fender: is that working on aastra's now?
23:15.25[TK]D-Fenderdlynes_laptop, shove the same auth in each "line" (what a bastardized use of the word they do...) and it should count them as *1* reg and span calls across those keys
23:15.40dlynes_laptop[TK]D-Fender: just the 'auth'?
23:15.41[TK]D-Fenderdlynes_laptop, its how I had it on my God-aweful 57i CT
23:15.56[TK]D-Fenderdlynes_laptop, set 1-3 up identically and see it span...
23:16.00dlynes_laptop[TK]D-Fender: and the 'auth' matches the '[auth]', or the username=auth?
23:16.10[TK]D-Fenderdlynes_laptop, No, nothing to do with *.
23:16.15[TK]D-Fenderdlynes_laptop, on the PHONE'S config
23:16.18dlynes_laptop[TK]D-Fender: ok...lemme try
23:16.19metfan2007hi all, how can I tell Asterisk to always try to start a trunk if the link is down? After some time Asterisk stops to try to look for the other Asterisk box...
23:16.20dlynes_laptop[TK]D-Fender: thanks
23:16.36[TK]D-Fenderdlynes_laptop, go into each "lines" definition and set them IDENTICALLY and test to see that it spans.
23:16.47[TK]D-Fenderdlynes_laptop, place a call out and have another call come in.
23:17.00[TK]D-Fenderdlynes_laptop, manual "channel hunting" is raging BS
23:17.31[TK]D-Fenderdlynes_laptop, unless its your INTENTION for each "line" to have seperate identities
23:18.12[TK]D-Fendermetfan2007, is the remote side on a fixed IP / host?
23:18.24metfan2007I see a few messages in the CLI saying that chan_iax2.c:7238 socket_read: Peer 'pedregal' is now TOO LAGGED (2028 ms)!
23:19.44metfan2007Fender: both Asterisk boxes has dynamic IPs with Dyndns domains
23:20.27[TK]D-Fendermetfan2007, then set "host=[thehostnameforthebox]", and "qualify=no" and forget about registering between the two.  There's no need.
23:20.33metfan2007Fender: after some "LAGGED", "UNREACHABLE" and "is now REACHABLE" messages Asterisk stops to reach again the other box
23:21.16[TK]D-Fendermetfan2007, this way when you dial out it'll just dial and not care if it think they're up or not.  It'll try and if it succeeds, then it succeeds, if not, just try again
23:21.16metfan2007Fender: currently I have to do a manual iax reload to Asterisk start to reach the other host
23:21.34*** join/#asterisk agx (n=badpengu@81-174-46-120.dynamic.ngi.it)
23:21.36metfan2007Fender: ok ok
23:22.01[TK]D-Fenderthose lag warnings only come if you tell * to MONITOR (and hence CARE) the remote host.  You should not BOTHER caring in this case
23:22.21*** part/#asterisk agx (n=badpengu@81-174-46-120.dynamic.ngi.it)
23:24.00metfan2007Fender: so do I need to change "qualify=no" on both sides? they are acting as a trunk
23:24.10metfan2007Fender: with "trunk=yes"
23:25.42[TK]D-Fendermetfan2007, shouldn't matter
23:26.02[TK]D-Fendermetfan2007, once a channel is up it should stil aggregate the calls
23:26.21metfan2007Fender: ok, I'll try, so what's "qualify=yes" used for?
23:26.48[TK]D-Fendermetfan2007, as a way for * to chose to GIVE UP trying to call a dead host
23:27.10[TK]D-Fendermetfan2007, and helps as a NAT keep-alive, etc
23:27.28metfan2007Fender: and "maxregexpire" and "minregexpire"??
23:27.28[TK]D-Fendermetfan2007, none of which applies to your situation.
23:27.56[TK]D-Fendermetfan2007, you don't even NEED to register.  you have a host name.  as long as thats valid there is no point in registering at all
23:28.10metfan2007Fender: ok ok
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23:31.30[TK]D-Fendermetfan2007, the only thing "register"ing does is inform the server to which IP it should send calls.  This doesn't change how the calls themselves are authed, and as you always know where to go it serves no purpose
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23:37.19metfan2007exit
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