IRC log for #asterisk on 20100512

00:02.01p3nguinAdding that localnet didn't help.  Here's sip debug and rtp debug: http://pastebin.com/DbRj4K8V
00:04.34p3nguinPeer audio RTP is at port 172.16.255.21:16428    <-- this seems like a problem.
00:05.01*** join/#asterisk kazaa_lite (~eddie@212.183.140.2)
00:06.27kazaa_litehi all
00:07.37lepineI remember reading about playing with goto and contexts recursively ... and that page mentionned being able to refer to the parent context (eg, goto) ...
00:07.59lepineCan someone point me in the right direction? (eg, wiki page name or topic)
00:09.14lepine<PROTECTED>
00:10.13*** join/#asterisk fleebailey33 (~fleebaile@unaffiliated/fleebailey33)
00:11.43p3nguin[tk]d-fender: Is that enough debug to see what's going on?
00:21.43*** join/#asterisk cidu (~PISSSSSSS@whthyt253-29.northwestel.net)
00:23.14ciduso, quick question here, having trouble with CALLERID(ANI) over PRI, cant seem to get asterisk to recieve, or transmit CALLERIS(ANI) properly, its always populated with the CALLERID(NUM) Value and not the charge number data
00:23.27ciduis this a known bug? or am i overlooking something simple
00:29.42*** join/#asterisk QbY (~kelvin@c-24-126-145-123.hsd1.ga.comcast.net)
00:30.15QbYwith a realtime voicemail configuration, how does one force * to re-load the voicemail table?
00:30.44ciduno clue, never used realtime, and the channel is really quiet
00:37.27*** join/#asterisk coppice (~chatzilla@m121-202-34-88.smartone-vodafone.com)
00:38.46*** join/#asterisk sourcode (~code@ppp-58-8-127-12.revip2.asianet.co.th)
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00:46.18LemensTSanyway to remove a user when you do sip show peers?
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01:20.48*** join/#asterisk prgmrchris (~chris@66.9.61.162)
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01:21.29prgmrchriscan someone help me figure out why my dialplan doesnt work? im really new to this and im sure its a simple mistake: http://pastebin.com/wWFshffM
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01:24.43*** part/#asterisk mpd (~chatzilla@bas1-malton22-1167904914.dsl.bell.ca)
01:25.16carrar~book
01:25.16infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
01:25.40prgmrchris?
01:26.41carrarexten matches on DNIS not DID
01:27.02carraror ANI
01:27.26prgmrchrisso how do i write a rule for a certain DID?
01:27.50prgmrchriswhat do i use instead of exten
01:27.56TJNII"from-trunk"  freepbx?
01:28.02carrarYou are matching on DNIS or ANI?
01:28.09prgmrchriscarrar: yeah thats what im trying to do
01:28.37prgmrchrisim looking at http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns for general info, if you know any other good links that might help it would be appreciated :)
01:29.08TJNIIprgmrchris: Did you install from source or are you using some packages Asterisk?
01:29.19prgmrchrisTJNII: does that matter?
01:29.59prgmrchristhe question is pretty generic i dont think the packaging matters in this case
01:30.07TJNIIYes, because if you used FreePBX or TrixBox your dialplan is wrapped in their configs and your call could be going off into GUI la-la land.
01:30.21carrarprgmrchris, install Asterisk from Source
01:30.25carrarthen come back
01:30.33prgmrchristhe distribution has freepbx but there is no incoming routes or anything in freepbx
01:30.40TJNII~freepbx
01:30.40infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
01:30.45carraror try #Freepbx
01:31.05prgmrchrisi have another box which is from source
01:31.09TJNIINot trying to be rude, but it is damn hard to know what that dialplan is doing, due to the complexity.
01:31.10prgmrchrisand it doesnt work there either
01:31.18prgmrchrisi dont think its an issue with that
01:31.24TJNIIpastebin from the source box.
01:31.30prgmrchrisok
01:31.30TJNIIWe can help you a lot better.
01:31.32carrarprgmrchris, install Asterisk from Source
01:31.36carrarthen we can help
01:31.37prgmrchriscarrar: i did, read above
01:32.34carraralso check out that book
01:32.39carrarthat goes over basic dialplan
01:33.35carrarWhy did you paste the context [test]?
01:34.51prgmrchrisi thought it needed to be included to work
01:34.56prgmrchrisim checking out the book now, thanks
01:35.20carrarplus you're using freepbx
01:35.24carrargood luck
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01:35.53prgmrchriscarrar: im not right now. no
01:36.02carrarThat paste is
01:36.26prgmrchrisi know it is, if you read what i wrote earlier you will see that i said the first box had freepbx and the second box doesnt
01:36.34prgmrchrisive been pretty clear about that
01:36.49carrardon't paste irrelevant stuff for us to fix
01:37.09encinomanAnyone have any advice on implementing call conferencing?
01:37.12prgmrchrisits not irrelevant, it was a honest question, to which TJNII advised to compile from source which i have
01:37.15TJNIIWell pastebin the output of the second box.  Then we can try and help you spot the error without the freePBX junk.
01:37.46prgmrchrisTJNII: i will, im going to read the book some first and see if i can figure it out myself first, more rewarding :) if i get stuck i will ask again
01:37.49TJNIII don't think you have a dialplan error, but I need some output to back that up.
01:37.51prgmrchristhanks for the help
01:38.28TJNIIOkay, make sure to verify your contexts in sip.conf.
01:39.33*** join/#asterisk halkun (~chatzilla@mke-66-97-112-72.milwpc.com)
01:39.44p3nguinI've encountered a NAT issue where the remote device's RTP packets have the PRIVATE IP address on them instead of the PUBLIC address.  If anyone cares to help, here's the sip debug and rtp debug: http://pastebin.com/DbRj4K8V
01:40.13p3nguinThe problem is related to this line:  Peer audio RTP is at port 172.16.255.21:16428
01:41.04p3nguinThe peer's config includes nat=yes, and the localnet setting is including all RFC 1918 ranges.
01:42.07p3nguinUsing a softphone rather than the SPA-3102 on that LAN works fine.  The RTP packets have the public address on them when using zoiper.
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01:49.55LemensTSMoOo
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02:10.18*** mode/#asterisk [+o Deeewayne] by ChanServ
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02:36.39[TK]D-Fenderp3nguin: pastebi your sip.conf masking only passwords ([general] and [203])
02:38.20*** join/#asterisk geneticx (~geneticx@adsl-10-113-95.mia.bellsouth.net)
02:40.58[TK]D-Fenderprgmrchris: Your dialplan looks fine.  It is however completely unused.
02:41.49coppiceyou mean "as new condition"? :-)
02:42.51[TK]D-Fendercoppice: "new code smell"
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02:48.33p3nguinsip.conf:  http://pastebin.com/F6biEM9y
02:49.43p3nguinErm, wait... something was cut off the bottom.
02:52.54[TK]D-Fenderp3nguin: Um, is this SPA just completely remote?  No VPN or anything, right?
02:53.59p3nguinThat's correct, simply a remote LAN using regular old UDP for SIP/RTP.
02:54.24[TK]D-Fenderp3nguin: If you don't have those subnets we talked about as actually local, trash them
02:54.31[TK]D-Fender(localnet)
02:54.37[TK]D-Fenderp3nguin: Just keep the real one.
02:55.10p3nguinA few of the settings for the peer got cut off in the first paste, not sure what happened.  This is the re-paste if you need it: http://pastebin.com/uxppq0mp
02:55.22[TK]D-Fenderp3nguin: Waitasec.... were you working on SIP AGL settings on a Cisco for this earlier with MmanxPower?
02:55.43p3nguinHe insisted I'm using it, but I'm not.
02:55.59[TK]D-Fenderp3nguin: what routers are on each end?
02:56.23p3nguinLinux box/iptables on the client side, Cisco 831 on the server side.
02:57.14coppice[TK]D-Fender: new code usually stinks
02:57.22p3nguinFor what it's worth, I started out with only my real localnet in the conf, then added the rest earlier.
02:59.08[TK]D-Fenderp3nguin: Which end doesn't get audio?
03:00.12p3nguinneither way, two-way no audio
03:00.41p3nguinplacing a call to that device or calling from that device, same result, no audio at all.
03:00.42*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
03:01.46p3nguinIt puzzles me that zoiper on the computer on that remote LAN puts the public IP address in the RTP packets, but this SPA device is putting the private on the packets.
03:02.02p3nguinand, of course, zoiper works.
03:02.08p3nguingood audio with it.
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03:05.28[TK]D-Fenderp3nguin: ensure that there is NO NAT awareness on the SPA
03:05.39[TK]D-Fenderp3nguin: Also, what port is it connected via?
03:05.53p3nguinI checked it, all boxes are either cleared or marked NO.
03:06.16p3nguinHmm, you mean either the WAN/Internet or the LAN port on the device?
03:06.53p3nguinIf so, I'm actually not sure about that.
03:07.15p3nguinWhich way is ideal?
03:09.28p3nguinIt is connected via WAN port.
03:09.34p3nguinNothing is on the LAN port.
03:10.23[TK]D-Fenderp3nguin: OK so far.... wondering if something else is off...
03:27.22p3nguinIt's a puzzler.
03:34.51*** part/#asterisk LemensTS (~LemensTS@adsl-70-238-147-211.dsl.stlsmo.sbcglobal.net)
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03:46.54*** join/#asterisk sawgood (~sawgood@173-13-158-25-sfba.hfc.comcastbusiness.net)
03:47.57sawgoodI get now ... I really do ... I understand and appreciate using Asterisk CLI only (even Asterisk 1.4.x) ... but the learning curve for voicemail and IVR and ring groups gets steep
03:48.09sawgoodhaving FreePBX to fall back on for these things is so tempting
03:48.20sawgoodI guess I'll take a break for a while
03:48.52[TK]D-Fendersawgood: Learning curve.... for voicemail?
03:49.06sawgoodwell, just needed a place to vent ... so to speak
03:49.21sawgoodhaving to build the entire Asterisk dialplan from code is time consuming and hard
03:49.23coppice[TK]D-Fender: maybe the voice mail is in a foreign language
03:49.37carrarlet it out
03:49.38[TK]D-Fendersawgood: Whats to learn?  a VM box is 1 line of config.  Sending a call to leve a voicemail is also 1 line of dialplan.  Picking up voicemail is also 1 single line of dialplan...
03:49.48sawgoodin general I meant not having a set of GUI  tools
03:50.05carrarhaha'
03:50.10carrarseriously?
03:50.12sawgoodmaybe voicemail was a stretch .... I really meant IVR ... and what not
03:50.32carrarGUI is restrictive
03:50.39sawgoodyou can say that again ...!
03:50.58[TK]D-Fendersawgood: Dial(SIP/100@SIP/110@SIP/120,20) <-- congrats.. a "ring-group" .... in again... 1 LINE OF DIALPLAN
03:51.00sawgoodis there an 'easy' way to have grep print out to the screen only lines which do NOT start with ;
03:51.34carrargrep  -v "^;" poop.txt
03:51.42sawgoodthank you
03:52.18carrarBut you should remove all the lines with ; anyways
03:52.29carrarunless you added them for a reason
03:53.35sawgoodwhat does the ^ sign do for the syntax?
03:53.46carrarplease visit regex
03:53.55sawgooddon't get me wrong it worked just as advertised ...
03:53.58sawgoodjust trying to learn
03:54.03carrarbeginign of line
03:54.46sawgoodI love that O'Reilly book, "Understanding Regular Expressions" .... I might should read it again soon
03:55.04carraror just google
03:55.45carrarthey are all in google books
03:58.14carrarI hsould visit this
03:58.14carrarhttp://en.akihabaranews.com/44454/misc/bandai-open-its-first-gundam-cafe-in-akihabara%E2%80%A6-and-we-tested-it
03:58.22carrarno idea what Gundam is though
03:59.20carrarBarista loks cute
04:00.29carrarsawgood, you remove your comment lines yet?
04:02.36sawgoodI did ... thank you
04:03.06carrarYou'll be graded on a clean and orderly extensions.conf
04:03.42sawgoodI can easily make the SIP trunk work if I set the box up on a FreePBX build, but Asterisk only (which you would think should be easier) is not working
04:03.50sawgoodI know I have something incorrect in sip.conf
04:03.55sawgoodI am getting closer and closer
04:06.53*** join/#asterisk adam_g (~adam@c-67-189-34-196.hsd1.or.comcast.net)
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04:15.19halkunOk, I have an account in asterisk, I and the server talks to my xlite. I have the 3102 on the network and can see it's config page when I go it it's IP address. I need some instruction on how to link the 3102 to asterisk so I can make calls on my land line.
04:15.41halkun(Almost there :))
04:16.44[TK]D-Fenderhalkun: Dial(SIP/peertothe3102fxoport/1234567890)
04:18.53adam_ghmm. strange issue where we've switched from nat = route to nat = no, all but one SIP peers is working as it should. the other still shows Addr->IP = its old source, as if it were still being natted. also with a weird port ( 1029 as opposed to 5060), any ideas?
04:19.26adam_gdo such address get cached for some period? rtcachefriends is not set to yes (though not explicitly set no)
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04:30.54halkunI found this as a quick setup
04:30.59halkunhttp://forum.voxilla.com/cisco-linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html
04:32.28halkunhowever, it's a little different than "default" as I don't need 10 digit dialing and asterisk is the internet gateway while the 3102 is behind that. It appears to be the reverse on this setup.
04:34.48halkunthe dial plan from that setup is Dial Plan: (*xx|[3469]11|0|00|[2-9]xxxxxxxxxS0|1xxx[2-9]xxxxxxS0)
04:35.04halkunis there a place I can go to decipher that?
04:36.18[TK]D-Fenderhalkun: thats from the FXS portion, right?
04:36.57[TK]D-Fenderhalkun: Do yourself a favor :  (*x.T|#x.T|x.T)
04:38.54halkunyou see, I don't know what that means/
04:39.32halkun(Embarrassingly, I'm getting FSX and FXO confused now)
04:39.55[TK]D-FenderhlLINE vs PSTN tabs in the GUI admn for it
04:40.04[TK]D-Fenderhalkun: LINE vs PSTN tabs in the GUI admn for it
04:40.23[TK]D-Fenderhalkun: LINE = FXS = Phone.  PSTN = FXO = your phone line
04:41.10*** part/#asterisk ruben23 (~ITadmin@125.212.40.2)
04:41.12halkunFXO = the thingy that makes the dial tone
04:41.27p3nguinno
04:42.00halkunFXS = walljack
04:42.20p3nguinyes
04:42.52halkunFXO = jack at the end of the phone cord
04:43.20[TK]D-Fender~fxsfxo
04:43.20infobot[~fxsfxo] 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.
04:43.38[TK]D-Fender~fxs
04:43.39infobot[fxs] foreign exchange station - type of port you need to connect an analog device (phone, fax machine) to a pbx.  This is the type of port found in your wall jack.
04:43.40[TK]D-Fender~fxo
04:43.40infobot[fxo] foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx. This is the type of port found on phone or fax machine devices. See also: http://www.digium.com/index.php?menu=fxsvfxo
04:44.18halkunyou man need to update that last URL
04:44.20halkunit 404s :)
04:45.27halkun~dialplan
04:45.27infobotmethinks dialplan is the thing configured in extensions.conf
04:45.44halkunhmm
04:46.51[TK]D-Fenderhalkun: just use what I gave you for the LINE page dialpan
04:47.27*** join/#asterisk sourcode (~code@ppp-58-11-75-89.revip2.asianet.co.th)
04:47.34halkuncan you explain it to me before I pop it in?
04:49.35[TK]D-Fenderhalkun: the dialplan on the device determines when it thinks you're finished dialing your number before passing it on (in whole) to *
04:50.02*** join/#asterisk Gopal (~Miranda@61.12.17.170)
04:50.24[TK]D-Fenderhalkun: You could set it so 1000-1999 are taken as "as soon as the 4th digit in the pattern matches, consider it complete and send immediately".
04:50.35[TK]D-Fenderhalkun: To speed up calling local phones, etc.
04:51.34[TK]D-Fenderhalkun: This requires you to program all the kinds of patterns you'll actually be coding in your ASTERISK dialplan anyway.  So better off just telling it "take what I dial and just wait till I stop for 3s before considering that I'm done".
04:51.39[TK]D-Fenderhalkun: the STFU dialplan.
04:52.50ChannelZthe Shitty Tit Fuck Up dialplan!
04:53.51JumpieChannelZ is that even possible?
04:53.59Jumpieto shitty tit fuck
04:54.05halkunso putting (*x.T|#x.T|x.T) in the 3102 will allow me to set up the dialplan in * and all I'm doing here is telling the 3012 to take what it is given
04:54.34ChannelZJumpie: well if you were doing it to Nancy Pelosi.. that'd be pretty shitty
04:54.41Jumpiehaha
04:54.43Jumpieugh
04:54.50Jumpiejust hearing that skanks name makes me wanna vomit
04:55.35ChannelZSee!
04:56.00[TK]D-Fenderhalkun: Yes
04:56.15halkunhttp://craphound.com/images/5jplr.jpg  <---- This needs to be the official asterisk t-shirt (funnier if you have read "Breakfast of Champions"
04:59.12halkunOk, reading ahead... This goes into the "Line 1" tab
04:59.38halkunbecause there is another dailplan that goes into the PSTN line
05:00.00[TK]D-Fenderhalkun: that one is very separate and different
05:00.17halkunOk
05:02.24ChannelZdamnit what's the app that plays signal tones?  I though it was PlayTones or something
05:03.57halkunOk, so now I'm configuring PSTN now....
05:04.24halkunproxy is the * server (192.168.0.1)
05:05.25halkununder dialplan 8 it says to put S0<:123@192.168.0.2>
05:05.36halkunoops, wrong I'm at .1
05:06.30halkunso in this example, the incoming calls are coming on extention 123
05:06.57*** join/#asterisk `Sauron (~sauron@dsl001-130-033.aus1.dsl.speakeasy.net)
05:10.07ChannelZhmm darn there is no indication sound for 'left your phone offhook'
05:11.58*** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein)
05:16.38halkunhmm
05:17.14halkunOk, I have a test config in my sip.conf that works with a usrename ans secret
05:17.56halkunoh, never mind... found it
05:19.49ChannelZHmm.  I need a really cheesy version of Girl From Ipanema, like Musak version with xylophones and crap
05:20.59*** join/#asterisk trelane (~trelane@funtoo/staff/trelane)
05:23.37carrarhttp://www.youtube.com/watch?v=_ZmQr78Otv4&feature=fvw
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05:24.44[TK]D-Fendercheckout time.  Later all
05:24.50carrarWAIT
05:24.53carrarDon't GO
05:25.18carrarI've giving away $1,000,000 buy only to you in 5 seconds if you are here
05:25.22carrarbut
05:25.31carrardoh, missed out
05:26.05halkunhmmm
05:26.27halkunI put the new setting in there and now xlite is giving me a 403 - forbidden error
05:27.21halkunI guess it's finding the account
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05:28.25ChannelZcarrar: that's pretty good, but instrumental.  I think I found a sufficiently shitty one on amazon
05:28.26Gopalhalkun: if it is 403 may be some authentication prob with username and password or auth name
05:28.53ChannelZIt's actually not quite the song I was thinking of though, I guess I don't know the name of the one I'm thinking
05:29.29halkunThe sip.conf I'm using is from here ---> http://forum.voxilla.com/cisco-linksys-sipura-voip-support-forum/starter-spa3102-asterisk-setup-18612.html
05:29.52halkunI'm logging in with with the username of "line1" and my passowrd
05:30.07halkunThat's the only lie I see with a username
05:30.17halkunline I mean
05:31.21halkununder the [xlite] section I don't see a username up there
05:31.49halkunonly under the [line1] section
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05:34.14Gopalusername will is the context name [xlite] is the uername
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05:37.46halkunERROR[9913]: chan_sip.c:8671 register_verify: Peer 'line1' is trying to register, but not configured as host=dynamic
05:38.26halkunyou know, you can get a much of really awesome errors when you actually look at what * is doing...
05:39.05halkunaha :)
05:39.46halkunhelps when you log in with the right username... thanks
05:40.40Gopalhalkun: you have to give host=dynamic in your context [xlite]
05:41.15halkunthat's there....
05:41.49halkunit's saying I need host=dynamic for "line1"
05:42.43halkunbut that's not right. I have "host=192.168.0.13" (The address of the 3102)
05:45.49Gopalhalkun: try giving host = dynamic
05:47.01Gopalhalkun: you have to give your sip provider IP as host
05:47.04p3nguinIf your device is trying to register, you can't use host=ipaddress, you have to use host=dynamic.
05:47.37halkunhttp://scsys.co.uk:8002/43469
05:47.59halkunOk, I'll make id dynamic ans see
05:48.29halkunthe URL above is the error and the sip.conf I coypasta-ed
05:50.53halkunwell, it's dynamic, and I'm not getting and more errors in CLI
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05:54.52halkun<PROTECTED>
05:55.18halkunlooks like I'm off to configure extensions.conf
05:57.02halkunhhhm
05:57.12halkunthe example here uses the 10-digit numbers again
05:57.36halkunI don't think that's going to work as I changed the dialplan in the 3102
05:58.04p3nguinThe dialplan in the 3102 just tells it where to send what calls.
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05:58.15p3nguinExtensions.conf is where you have to match dialed numbers for processing.
05:58.26p3nguinextensions.conf, rather.
05:58.40carrar~book
05:58.40infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
05:59.52Gopalhalkun: yes you need to have a dialplan to route the outgoing calls
06:01.09halkunin that case I'm going to bed for the night.
06:01.23halkunI'll deal with dialplans as another project.
06:01.33carrarafter you read the book
06:01.49p3nguinDon't confuse asterisk's dialplan with the dialplan on the device.
06:02.05halkunI'm just happy that xlite is logging in and (theoretically) the 3102 is connected
06:02.10carrardialplan or digit map?
06:02.33p3nguinPolycom calls it s digit map, but Cisco/Linksys call it dialplan.
06:03.05halkunthanks for the help guys :)
06:10.19ChannelZcarrar: call SIP/spark.idolum.com
06:10.47p3nguinWhen I dial out via future nine, the counter on the phone never starts, indicating that the call is still in progress instead of ever being answered (even though the far end has answered).  Is there anything I can do on my end to make the line go up?
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06:13.52p3nguinI guess there was... switching to ideasip instead of future nine solved it.  :/
06:14.17p3nguinDialed the same number through the other provider and it showed Connected and the counter started.
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06:16.16ChannelZsounds like their system isn't sending back progress
06:16.49p3nguinYeah, and that prevents me from being able to pass DTMF to the other end of my calls.
06:17.45p3nguinBut, since I wasn't paying them for any services, they probably don't really care that their system is broken.
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06:48.11b14ckIs there a way in dialplan to convert a string variable to an int?
06:48.22b14cklike INT(${my_string_var}) or something?
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06:52.57ChannelZvariables are sort of typeless.  They're all kind of strings.  Certain operators will treat them like numbers if they contain only numbers
06:53.05b14ckgotcha
06:53.06b14ckthanks
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07:33.47Gothicmaster86good morning
07:34.20xhelioxgood morning
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07:35.58Gothicmaster86has anyone here experience with AsteriskNow and ISDN?
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07:40.29zambaGothicmaster86: ask your question
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07:43.38Gothicmaster86zamba: i search a Version of AsteriskNow what works with ISDN-Cards. Without GUI.
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07:45.30kaldemarthe whole point of AsteriskNow is to be a complete distro that gives you a subset of features and is configured via a GUI. maybe AsteriskNow is not what you're looking for?
07:46.26kaldemarchoose your favorite linux distro (if you have one) and install asterisk on it. what kind of ISDN card do you mean?
07:48.57Gothicmaster86kaldemar: Maybe i've installed Asterisk 1.4.29.1. on a Debian Etch. The Problem is, i didn't found any Driver for an ISDN HFC-S PCI-Card (1 Port) what works without problems!
07:50.54Gothicmaster86kaldemar: Hardware-bridging is terrible (my Distro kills hisself at a Call) and Software-bridging or echocanceling is with the actual drivers impossible
07:52.47kaldemaryou tried wcb4xxp on the latest dahdi?
07:56.26kaldemarthere might be lack fo support for BRI in 1.4 chan_dahdi anyway. i'd try 1.6 and if it doesn't work for you, buy a card that is known to work. i've seen more issues than success with generic HFC-S based cards.
07:57.09Gothicmaster86kaldemar: i've tried, but it doesn't work. DAHDI doesn't acknowledges my cards. In the moment it works halfway with mISDN-1.1.7.2
07:58.31Gothicmaster86kaldemar: have you experience with the new vISDN? or mISDN v2?
07:59.51kaldemarneither. it's been years since i've used any kind of BRI. i used to use bristuff with HFC-S cards on asterisk 1.2.
08:01.03Gothicmaster86kaldemar: i heared bristuff didn't work with asterisk 1.4.* and newer...
08:01.59kaldemarit does, 0.4.0 versions are meant for 1.4: http://www.junghanns.net/downloads
08:03.33kaldemar0.4.0-RC3k is meant to build against asterisk 1.4.29.1. 0.4.0-RC4 is the newest, updated to asterisk 1.4.31
08:04.01Gothicmaster86kaldemar: and what you mean is better? Capi or Bristuff?
08:05.16kaldemari've never tried capi.
08:06.21Gothicmaster86hmm then i would check out bristuff. thanks ;)
08:08.20Gothicmaster86i must anyway make new my Distro. Then i could try the new driver. (i think i've killed my Kernel with a bad kernel-update)
08:09.17Gothicmaster86exit
08:11.04xhelioxhmms
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09:59.11AndroFlexhello Can anybody give me some fast advice? I'm trying to develope a CTI with Flex, but I don't want to use Java + Flex or .Net + Flex, I want to use flex alone? any ideas?
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10:06.21BrokenNozeHi all. I have a very serious problem with a production on asterisk 1.4.20.1, after about 2 weeks of successful operation my IAX channel reports "maxium retries succeeded" between 2 systems and then hangs. Asterisk will not restart, stop and I am unable to even use kill to kill the pid. The only solution is a full reboot. Upon the reboot the system comes back up without any problem and continues working again
10:06.58BrokenNozeIs there any known issue within the dialplans that could cause this? is there any work around? I would post on forums but i am desperate for a soultion!
10:07.39BrokenNozeI should say that it had been working for 3 years until very recently and the only change i can think of is one within the dialplan itself..
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11:40.11wonderworldhi. can i run shell commands from the asterisk CLI?
11:40.39kaldemarwonderworld: !<command>
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11:43.05wonderworldtnx
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11:51.52Gopalhi steve123
11:51.57Steve123hi
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12:25.56chripheris there any soft phone for linux that support auto provisioning
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12:28.30[TK]D-Fenderchripher: zoiper
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13:13.40hlueseahello channel
13:14.29Baylink-lastdayhears the channel say "Hi", and wonders why
13:15.37hlueseais anybody know that : i have 16 channels and 4 channels 2 different trunks and i want to outcalls using that  but they have a channel limitation. How can i limitation each channel and  whole outbound calls ?
13:16.46[TK]D-Fenderhluesea: GROUP()
13:17.22hlueseabut that is not dahdi channel, each one is the different brands sip gw mean sip trunks
13:17.31[TK]D-Fenderhluesea: GROUP() <- DIALPLAN FUNCTION
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13:19.23hlueseathanks [TK]D-Fender i am looking on
13:22.01BarthezZhmm, is it normal for a sip provider to deliver 1 account per possible active account? I'm used to get one (or one account per number) which can hold multiple calls
13:22.52[TK]D-FenderBarthezZ: One account per account?  Unheard of...
13:23.06BarthezZuhmm excuse me, one account per possible outgoing call
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13:23.22*** mode/#asterisk [+o Deeewayne] by ChanServ
13:23.30*** join/#asterisk sigius (~sigius@93-125-185-45.dsl.alice.nl)
13:23.36pabelangerBarthezZ: Anything is possible, their network, their rules.
13:24.38sigiusQ: What is a popular asterisk load testing tool ?
13:25.00BarthezZhmm, than I have to think of a way to check for which account is idle (no call on it) and use that one for an outgoing call
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13:27.19[TK]D-FenderBarthezZ: "core show application ChanIsAvail"
13:27.22[TK]D-Fendersigius: SIPP
13:28.08BarthezZyeah [TK]D-Fender, just my outgoing dialplan is going to be a bit messy in that case
13:28.30[TK]D-FenderBarthezZ: Pick another provider then
13:28.33BarthezZah well, sometimes you need to adapt to the customers wishes :( just wish they would have consulted me before selecting a sip provider
13:29.46Gido-EBarthezZ do they want technology or a sip account?
13:30.11BarthezZtechnology? what do you mean?
13:30.37Gido-EBarthezZ you can always say, buy this or do that.
13:31.05BarthezZyes... but before consulting me they already registered with a sip provider, got a block of numbers, etc... and signed a 2 years contract
13:31.32BarthezZthe guy who hired me does know something, but misses some important parts of knowledge :p
13:33.00[TK]D-FenderBarthezZ: Time for some messy dialplan.
13:33.09Gido-Eyep :-)
13:33.13BarthezZyup :(
13:33.34BarthezZah well, first get that crappy ISDN hardware running for the lifeline
13:33.53BarthezZopenvox is a pain in the ass
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13:42.53ruben23hi guys
13:44.26Baylink-lastdayMorning (he said, falling manfully on the grenade)
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13:48.23WolfeyesGood day everyone.
13:48.41WolfeyesAnyone here from South Africa?
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13:57.10WolfeyesAnyone willing to help a beginner in here?
13:58.23Baylink-lastdaySure.
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13:58.48Baylink-lastdaysaid, falling manfully on another grenade. :-)
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14:01.31werty1sthello can somebody help with an asterisk sending a wrong/unknown ip address
14:01.50werty1sti dont know how asterisk detects its wan IP
14:02.46werty1ston debug i can read this: We're at 79.212.110.117 port 10036
14:03.10*** join/#asterisk UQlev (~yuriy@212.50.99.8)
14:03.18werty1stthis is definitely  not my wan address
14:03.44Gido-Ewerty1st dont use nat
14:04.35werty1sti have pfsense with siproxd
14:04.43werty1sti disable nat now
14:05.38*** join/#asterisk ManxPower (~manxpower@61.sub-75-254-195.myvzw.com)
14:06.58werty1stshouldi disable STUN too?
14:10.13[TK]D-Fender~sipnat
14:10.13infobot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
14:10.16[TK]D-Fenderwerty1st: ^^^^
14:12.06werty1sti have SoftPhone -> wanIP -> pfsense -> asterisk
14:12.24werty1stpfsense has siproxd and NAT configured
14:12.29*** join/#asterisk KavanS (~KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
14:12.55werty1stport 5060/tcp and 10000-10100/udp for rtp
14:13.28werty1stif i disable nat on asterisk i cant register my softphone any lnger
14:15.54ManxPowerYay wireless!  I get an external antenna for my EVDO card and I lose 3 DB of signal when I plug it in.  Maybe they send me an anti-antenna
14:16.15ManxPowerwerdan7, you realize SIP generally uses UDP, right?
14:16.41NivexManxPower: did it look like this? http://en.wikipedia.org/wiki/File:Cantenna.JPG
14:17.00ManxPowerNivex, no
14:17.26[TK]D-Fender[10:12]<werty1st>pfsense has siproxd and NAT configured <--- your firewall should GTFOP of *'s way
14:17.27ManxPowerCan you do cantennas on 1900/800 Mhz?
14:18.35coppiceManxPower: sure. don't you remember those early cellphone users with big parabolic dishes on their heads?
14:18.49ManxPowercoppice, must have been before my time.
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14:19.11Slugs_morning
14:19.18ManxPowerI guess I should check to see if I can return it to VZ
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14:21.36e3eli3hHi all. Anyone in here able and willing to help me troubleshoot a g729 codec installation issue?
14:22.08werty1sti have also setup the coresponding FW rules for incomming sip and rtp
14:22.58ManxPowerwerty1st, NO!  You do NOT need to do that when you are trying to make everything work.  FW should be the LAST thing you set up
14:23.19ManxPowere3eli3h, contact Digium for support for that commercial product
14:24.10werty1sti have no problem with internal clients
14:24.36ManxPowerJust let us know when you are ready to follow advice.
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14:25.00werty1stcould u explain me what to do first?
14:25.05ManxPowerservice iptables stop
14:25.22e3eli3h@ManxPower, tried that and I am getting no response, turning here as a last resort before I jump ship.
14:25.26ManxPoweror whatever you do to turn off your firewall on your asterisk box
14:25.51ManxPowere3eli3h, not much anyone here can do for you
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14:26.07e3eli3hI see.
14:27.10ManxPowere3eli3h, You copy the module to the asterisk modules directory, run the register application, enter your registration info and restart Asterisk,  This isn't rock science and is documented
14:28.44e3eli3hDone all that and the asthostid output is matching the host id in the generated .lic file. Hold on. You haven't even asked me what my problem is, have you?
14:29.13ManxPowere3eli3h, I doubt it will matter.  Digium is the only one that can support that binary non-open source codec.
14:30.00e3eli3hAgain, I see. Thanks for all your efforts.
14:31.58[sr]i'm sad
14:32.32*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
14:32.32*** mode/#asterisk [+o leifmadsen] by ChanServ
14:33.28ManxPowerYay leifmadsen!  The person that single handedly removed the requirement of TeX to read the documentation!  Yay!
14:34.48[TK]D-Fender[10:28]<e3eli3h>Done all that and the asthostid output is matching the host id in the generated .lic file. Hold on. You haven't even asked me what my problem is, have you? <--- You're supposed to TELL us what your problem is, not make us ask for it.
14:36.01*** join/#asterisk werty1st (~werty1st@p5B20FE46.dip.t-dialin.net)
14:36.26leifmadsenManxPower: yay! :)
14:36.44leifmadsenManxPower: that reminds me -- in the latest releases did it auto-generate the text version?
14:36.49leifmadsenI forgot to look for that
14:37.02ManxPowerleifmadsen, No idea. 8-)  I'm stuck on 1.4.x for a while more.
14:37.05leifmadsenit should have included asterisk.txt or something like that in the doc/ or doc/tex/ directory
14:37.15leifmadsenis there tex in 1.4?
14:37.19ManxPowerBut thousands of current and future generations thank you.
14:37.48leifmadsenI'm gonna go grab coffee and then I'll check when I get back unless you take a couple minutes to grab the latest RC and check for me :)
14:37.57e3eli3hAnyone in here able and willing to help me troubleshoot a g729 codec installation issue? <--- is a question. If you are both willing and able, I'll ask you the question and paste my logs in a private chat. I will not waste channel real estate.
14:38.10ManxPowerleifmadsen, I didn't pester you do to the text stuff for *me*.  I already know Asterisk pretty well.  I did it for all the poor n00bs out there.
14:38.18ManxPower~ask
14:38.18infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
14:38.40leifmadsenManxPower: I'm not sure where I said you did :)
14:39.05*** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek)
14:39.45ManxPowerleifmadsen, the thing is the txt versions were REMOVED in 1.6.mumble.
14:41.33*** join/#asterisk madduck (~madduck@debian/developer/madduck)
14:41.56madduck[TK]D-Fender: the provider (netvoip.ch) immedietaly responded and told me to set inband and relaxdtmf=yes, and now it works. just fyi
14:43.11[TK]D-Fendermadduck: Ah... CRAPPY inband DTMF.  If you have to do that... try to replace them ASAP
14:43.33ManxPowermadduck, Did they also tell you to use alaw (or ulaw)?
14:43.57madduck[TK]D-Fender: why? now it works.
14:44.06madduckManxPower: no. i told them i already tried alaw/ulaw
14:44.17ManxPowermadduck, because any carrier that requires inband is not a provider you want.
14:44.39ManxPowermadduck, um, inband DTMF doesn't work unless you are using ualw or alaw
14:44.49[TK]D-Fendermadduck: And any one that not only requires it, but also isn't stable enough that you require relaxdtmf is horrible
14:45.26ManxPower[TK]D-Fender, it's almost like a provider has a "kick me" sign on their back,.
14:45.28madduckManxPower: fair enough. inband means that the tones are transmitted, right?
14:45.42ManxPowermadduck, it means the DTMF is transmitted as audio.
14:45.59ManxPowerSo if your audio is always perfect, I guess inband will work.
14:46.01madduckright
14:46.16ManxPowerthe relaxdtmf thing is just bizarre.
14:47.57madducki asked them about it.
14:49.05ManxPowerin my experience relaxdtmf causes issues with double DTMF digits being seen as a single DTMF digits.  i.e. 54467 might be seen as 5467
14:49.26coppicereplace the DTMF decoder with a decent one
14:49.46ManxPowercoppice, that and removing relaxdtmf
14:49.46madducknetvoip.ch are the only ones that give me a zurich phone number for free. :(
14:50.03ManxPowermadduck, you.  get.  what.  you.  pay.  for.
14:50.15madduck;)
14:50.26coppicerelaxdtmf seems to be necessary to cope with some systems, but its pretty harmless with a good decoder
14:50.49ManxPowerhugs RFC2833
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15:01.31madducknow all i really need to figure out is how to turn a peer-to-peer phone call into a conference ;)
15:01.58madducki.e. how to call someone else and merge, or even let people join.
15:02.03werty1stthanks for trying no luck today :-(
15:02.06*** part/#asterisk werty1st (~werty1st@p5B20FE46.dip.t-dialin.net)
15:07.45leifmadsenManxPower: no removed -- converted I'm sure
15:08.32lepinejust deploy his first asterisk setup in production.
15:08.44WolfeyesAnyone from South africa in here?
15:13.04*** join/#asterisk MiserySoft (~LND@92.41.77.98.sub.mbb.three.co.uk)
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15:22.58Qwell<coppice> replace the DTMF decoder with a decent one
15:23.01Qwellcoppice: patches welcome :p
15:23.48coppiceQwell: you shouldn't have broken the one you have
15:25.09russellbyeah, thanks a lot Qwell
15:25.12russellb:-p
15:30.49leifmadsenQwell: sheesh!
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15:31.25wcselbyo/
15:32.04*** part/#asterisk MiserySoft (~LND@92.41.77.98.sub.mbb.three.co.uk)
15:39.41*** part/#asterisk ruben23 (~magisx@202.137.112.11)
15:43.01ManxPowerisn't there a DTMF detector in spanDSP?
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15:49.29coppice:-)
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15:52.57coppicedoes anyone know how fast Howlertech's new G.722 codec runs?
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15:54.30jonny330i am having problems dialing out on a dahdi line
15:55.34*** join/#asterisk analogkid (~xchat@dialin-145-254-158-213.pools.arcor-ip.net)
15:56.16analogkidhi, does anyone know if there are capi drivers for gentoo ppc?
15:56.22analogkidso linux
15:57.58jonny330i have a tdm800p and it takes 2-4 tries to call outbound, inbound works perfect
15:59.43Qwelljonny330: show us logs
16:02.13TSMis it correct that if someone hangs up from within the voicemail app it terminates the call script straight away instead of allowing the call script to continue to do other things?
16:02.39*** join/#asterisk MmixX (mixed@unaffiliated/mmixx)
16:04.53*** join/#asterisk Ad-Hoc (~nimbus@62.1.239.229.dsl.dyn.forthnet.gr)
16:05.34[TK]D-FenderTSM: "Call script"?  Pardon?
16:05.38wcselbyTSM - yes, when a call is hung up, it's hung up
16:06.23*** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
16:06.24TSMwell thats poo means that no one can script anything after VM, mabey there should be an option to continue with a status of VM for if a message was left or not
16:06.38wcselbyTSM - there's always the h extension.....
16:07.13wcselbyplus, there's an option in voicemail.conf that will execute a user defined command once a voicemail has been left
16:07.40wcselbyTSM - look at externnotify option in the sample voicemail.conf file
16:08.36TSMwcselby: yup but it would be nice if that was within the dialplan
16:08.55wcselbyTSM - ...... try the 'h' extension in the dialplan then
16:09.08wcselbyTSM - since we have no idea what you're trying to do (you're pretty vague), it's hard to help
16:09.45TSMahh, i have a script that notifies users of missed calls, prob is that it wont if they go into Voicemail and then just hang up
16:10.41wcselbyTSM - call the script before it goes to voicemail()....
16:10.41Qwellif it hits voicemail, it's already missed.  so why not notify before?
16:10.41p3nguinEh, the missed calls on the phone and the message waiting indicator isn't enough?
16:11.06TSM@Qwell: because i dont want a message that a call was missed if they do leave a message as then its two messages, one that its missed and another that they have left a message, but i do see your point
16:11.34wcselbyTSM - then we're back to the 'h' extension
16:12.34p3nguinIf there's voicemail, that's a pretty good indication that there's voicemail.  :/
16:12.44p3nguinAnd if the phone shows a missed call, that's a pretty good indication that the call was missed.
16:12.49wcselbyp3nguin - he's saying if they hangup before leav ing a voicemail
16:12.57*** join/#asterisk gelo (~gelo@mx01.quobis.com)
16:12.58ciduso, quick question here, having trouble with CALLERID(ANI) over PRI, cant seem to get asterisk to recieve, or transmit CALLERIS(ANI) properly, its always populated with the CALLERID(NUM) Value and not the charge number data, is this a known bug or what?
16:13.00p3nguin"missed calls"
16:13.09ciduerr, CallerID(ANI) even
16:16.15jhirleyo/ Hello Peeps
16:24.11*** part/#asterisk gelo (~gelo@mx01.quobis.com)
16:25.04p3nguinI've encountered a NAT issue where the remote device's RTP packets have the PRIVATE IP address on  them instead of the PUBLIC address.  If anyone cares to help, here's the sip debug and rtp debug:  http://pastebin.com/DbRj4K8V
16:25.51p3nguinUsing a softphone rather than the SPA-3102 on that LAN works fine.  The RTP packets have the public  address on them when using zoiper.
16:26.00*** join/#asterisk MhaddogM1 (~MhaddogM1@adsl-64-223-140.mia.bellsouth.net)
16:27.14p3nguinHere's a comparison of RTP from zoiper and from the SPA-3102: http://pastebin.com/1w3T3PKt
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16:42.20*** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt)
16:42.23[sr]howdy people
16:42.28[sr]i need to ask a dumb question :P
16:42.49[sr]i have a 2 port ISDN card: 00:10.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-4S] (rev 01)
16:43.15[sr]my question is, with this card, i can only connect it to my telephone network NTBA's
16:43.22[sr]and then route all in asterisk
16:43.35[sr]am i correct?
16:43.55[sr]if yes, i defined a incoming route for any DID/CID
16:44.14WIMPyNo you can also connect Telephones to it.
16:44.16[sr]to a SIP extension, but nothing happened and the number of the line i called, doesn't ring
16:44.27[sr]hum... Wimme
16:44.30[sr]WIMPy:
16:44.41*** part/#asterisk Baylink-lastday (~jra@cerberus.vicimarketing.com)
16:44.46[sr]i tried to connect a isdn extension from the central
16:44.57WIMPyWhat driver are you using?
16:45.11[sr]but the port didn't came green, as it did when i connect it to the PSTN NTBA
16:45.45WIMPyYou need a crossover cable to connect telephones.
16:46.04[sr]ahhhhh
16:46.23[sr]for the NTBA connection i was using a direct RJ45
16:46.30[sr]the driver is: wct4xxp
16:46.41WIMPyThat NOT the same as an ethernet crossover, however.
16:46.46[sr]is there a schema of the crossover cable?
16:47.33WIMPyHmm, I didn't succceed in NT mode with the dahdi driver.
16:47.55[sr]hum, which one then?
16:47.57WIMPy2<>4 and 5<>6
16:48.18p3nguin[sr]: First things first.  There is no "SIP extension."  There are extensions (found in extensions.conf) and there are SIP devices (found in sip.conf).
16:48.54[sr]WIMPy: is there a visual schema on the net? may be beter for me..
16:48.58p3nguin[sr]: So start from the top.  Do you have a peer entry in sip.conf for your device?  If yes, do you have an extension to Dial() to that device in extensions.conf?
16:49.17[sr]p3nguin: defined it on the interface
16:49.20[sr]freepbx
16:49.24Qwell[sr]: http://www.google.com/images?hl=en&q=isdn%20crossover%20cable
16:49.38WIMPy[sr]: With older kernels you can use misdn/can_misdn or with more recent kernels misdn2/lcr/chan_lcr.
16:49.39[sr]Qwell: merci, gonna check
16:50.11*** join/#asterisk ebroad (~EB@72.11.213.195)
16:50.17p3nguinFreePBX, eh?
16:50.51[sr]:$
16:50.52[sr]yes
16:51.05[sr]but wait, i'm going to start with the cable
16:51.19[sr]and make it work with the isdn extension
16:51.28p3nguinI wish my issue was with a cable.  It would make things so much easier.
16:51.45p3nguinEspecially since no one knows much about overcoming NAT problems.
16:51.57[sr]for the isdn extension i need a crossover, and for the NTBA connection i can use the direct cable, correct?
16:52.09WIMPycorrect
16:52.14[sr]perfect
16:52.58[sr]before i go, i have another machine with a 4x FXO card, and to make it work i have to change on dahdi to : from-zaptel
16:53.24[sr]i have to do the same herE?
16:54.00*** join/#asterisk dabaR (~dbernar1@24.77.24.161)
16:54.28[sr]it has the default: from-pstn
16:55.01dabaRI am trying to send voicemails as attachments and it looks like after the separator I get a blank line before the headers in the source of the email. This of course results in the attachment being interpreted as text. Any hints on where I could fix that?
16:55.33ManxPower~freepbx
16:55.33infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx
16:55.55*** join/#asterisk iluminator101 (~iluminato@unaffiliated/iluminator101)
17:07.24wcselbyasterisk 1.4.26 - is there any way to get the device state? i need to know if a phone is off the hook or available to be rung...
17:08.00hardwirestretches
17:08.25ManxPowerwcselby, SIP or ZAP?
17:08.30wcselbySIP
17:08.47Gido-Echanaivalable ofzo?
17:08.47Qwella SIP phone won't tell Asterisk that it's offhook
17:08.49ManxPowerwcselby, the phone does not contact the pbx when it goes off hook so you'll never get that info on sip
17:09.04[TK]D-Fenderwcselby: "core show application ChanIsAvail"
17:09.06ManxPowerchanisavail can check if the phone is inuse .
17:09.17ManxPowerthere are DEVSTATE backports for 1.4.x, IIRC
17:09.18*** join/#asterisk decaffeine (~decaffein@unison.bgp.cmk.ru)
17:09.19wcselby[TK]D-Fender - thanks, I'll look at that
17:09.29ManxPowerwcselby, looks like you might be using FreePBX
17:09.35decaffeineHi :)
17:09.43Gido-Efreepbx is much better dan asterisk.
17:09.55ManxPower*** Gido-E added to Ignore List
17:09.55wcselbyManxPower - no, I'm not
17:09.56QwellGido-E: cake is much better than buildings
17:10.10[sr]hey Qwell, with the crossover cable, connected to the central extension to the isdn card, it should become green algo right?
17:10.11Gido-E:)
17:10.34p3nguin<Gido-E> freepbx is much better dan asterisk.   <-- Do you have any idea what you're even talking about?
17:10.41Gido-Enobody gets the joke, are you are all on to much caffeine
17:10.42ManxPowerp3nguin, he is trolling
17:11.27p3nguinMight as well say something like, "CentOS is so much better than Linux."
17:11.33decaffeineCan somebody answer some question about linking astersik accounts with existing accounts on other sip provider?
17:11.33wcselbyManxPower - curious, what about what I said made you think I was using FreePBX?
17:11.33ManxPowerwcselby, as much as I HATE to admit it, you might try the asterisk rpms from FreePBX/TrixBox.  They have DEVSTATE and some other things already patched.
17:11.38[sr]Qwell: it is blinking red like it dopesn't have nothing connected
17:11.50ManxPowerdecaffeine, the answer is "Yes, everyone does that"
17:12.06decaffeineCool :)
17:12.12ManxPowerwcselby, 1.4.26 is the EXACT Asterisk version in many of our FreePBX installs.
17:12.28Gido-Edecaffeine that is one of the basic things you want normally.
17:12.28wcselbyManxPower - freepbx / trixbox is not an option for this client
17:12.34[TK]D-Fender~devstate
17:12.34infobot[~devstate] Devstate is an Asterisk 1.4 module for custom BLF device state, see the following link -=-  http://svncommunity.digium.com/community/russell/asterisk-1.4/func_devstate-1.4 , or http://www.asterisk.org/node/48325
17:12.34wcselbyManxPower - ahhhh, gotcha
17:12.36[TK]D-Fender^^^^^^
17:12.45decaffeineI'm new to astersik and PBX
17:12.55decaffeineso i have no skilz :)
17:12.56[TK]D-FenderFor non-device specific stuff
17:12.58ManxPowerwcselby, FreePBX is not an option for ANYONE.  People just ignore that fact and use it anyway
17:13.05wcselbyChanIsAvail should be what I'm looking for
17:13.06ebroadwasn't devstate ported to 1.6?
17:13.09ManxPowerdecaffeine, then you should be reading the asterisk book
17:13.13[TK]D-Fenderebroad: Yes
17:13.15ManxPower~book
17:13.15infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
17:13.36[sr]WIMPy: ideas? with the crossover cable the port doesn't became green as when i connect it to the NTBA
17:13.41decaffeineI've started it already
17:13.41ebroadthought so
17:14.08decaffeineThe main idea i want to use asterisk+adhearsion to test telephony on other sip prov
17:14.16decaffeinei.e IVR
17:14.30jonny330so i am having a lot of problems dialing out on a tdm800p the call does not go throguh on the first try they have to make 2-4 attemps calling the number
17:14.36*** join/#asterisk tokozedg (~toko@94.240.227.209)
17:14.50jonny330here is my dial string
17:14.50WIMPy[sr]: Maybe it's not configured properly? i.e. not in NT Mode?
17:15.03[sr]WIMPy: hum, how can i know that?
17:15.09[sr]know/configure
17:15.09WIMPyThat was what didn't work for me when using dahdi.
17:15.19jonny330Dial(DAHDI/r0/1${EXTEN},60,r)
17:15.42[sr]WIMPy: switchingtype ?
17:16.14WIMPyswitchtype must match, yes.
17:16.30[sr]hum ok i have euroisdn
17:16.36[sr]that's for NTBA for sure i guess
17:17.02[sr]and the other option should be?
17:17.52*** join/#asterisk AndyGraybeal (~AndyGray@128-177-27-78.ip.openhosting.com)
17:18.02WIMPytries to remember... I think it's 'signalling'. There is bri_cpe and bri_net or something.
17:18.15AndyGraybealwhat phone has the best (loudest, quality) speakerphone available?
17:18.34WIMPyBut the hardware also needs to know. That's what failed for me.
17:18.43WIMPyAnyway, I'm out for lunch.
17:18.44AndyGraybeali want to be able to hear the phone in a busy kitchen
17:19.06AndyGraybealon speakerphone
17:20.07*** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br)
17:20.19decaffeineThanks for the link
17:21.28jonny330anyone able to help me out with dahdi?
17:23.21*** part/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk)
17:23.23decaffeinegot no clue what it is....
17:23.24decaffeine%)
17:23.32ChannelZAsk an actual question
17:23.35decaffeineIt's in FreePBX
17:23.59p3nguinFreePBX still isn't supported here.  Still.
17:24.38decaffeineThanks for the book i will be later with questions :)
17:24.56p3nguinFreePBX won't be supported here when you come back, either.
17:25.37jonny330so i am having a lot of problems dialing out on a tdm800p the call does not go throguh on the first try they have to make 2-4 attemps calling the number
17:25.41jonny330Dial(DAHDI/r0/1${EXTEN},60,r)
17:26.06*** join/#asterisk lupino3 (~andrea_@static-217-133-45-108.clienti.tiscali.it)
17:26.17lupino3hello everybody
17:26.47ChannelZjonny330: do you get 'the number couldn't be completed as dialed' or somthing on a failed call?
17:27.02TSMfoudn the info i wanted to check if someone has actualy left a VM, ${VMSTATUS} returns FAILED if they dont leave a message and SUCCESS if they do
17:27.39jonny330ChannelZ: no nothing happens it takes a long time for anything to happen so users are hangning up
17:28.03lupino3can anybody help me with my Astribank? I randomly get USB errors "device not accepting address" and "device descriptor read/n" with errors -71 and -10
17:28.12lupino3in those cases the astribank is not available via USB
17:28.21lupino3but sometimes it Just Works (tm)
17:28.30lupino3I can't reproduce the issue consistently
17:28.38lupino3i tried two astribanks, two cables and two MB's
17:28.40ChannelZjonny330: well one possible problem is your telco isn't providing a dialtone quick enough when the hardware picks up the line.  Try putting a 'w' in before the 1 of your dial...
17:28.49ChannelZDial(DAHDI/r0/w1${EXTEN},60,r)
17:28.54lupino3does anybody have a suggestion?
17:29.16dabaRI am trying to send voicemails as attachments and it looks like after the separator I get a blank line before the headers in the source of the email. This of course results in the attachment being interpreted as text. Any hints on where I could fix that?
17:29.18tzafrir_laptoplupino3, "randomly" - also same combination of astribank and system?
17:29.25jonny330w1?
17:29.25lupino3yes
17:29.36ChannelZw means 'wait a second'
17:29.40*** part/#asterisk iluminator101 (~iluminato@unaffiliated/iluminator101)
17:29.42*** join/#asterisk d00gster (~dt@94.98.9.227)
17:29.44jonny330ahh cool
17:29.45lupino3tzafrir_laptop, the same hard disk on two different MB's
17:29.49ChannelZthe 1 was yours, it was already there in the string you procided
17:29.51ChannelZprovided
17:30.06*** join/#asterisk minaguib (~mina@modemcable115.49-57-74.mc.videotron.ca)
17:31.00lupino3tzafrir_laptop, when it is working, it survives the reboot of the machine, the reboot of the astribank and the usb unplug/replug cycle
17:31.12lupino3tzafrir_laptop, but when I get the USB errors... no way!
17:31.29jonny330yeah thats seems to be making calls more reliable
17:31.42jonny330but they are just taking a little longer to make it out
17:31.46lupino3tzafrir_laptop, I tried different USB ports (with same results)
17:32.00lupino3tzafrir_laptop, and on each port I tried an USB key (it works)
17:32.15lupino3tzafrir_laptop, suggestions? while you think I just go banging my head on the nearest wall :(
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17:32.22jonny330ChannelZ: is there anytying else you can recommend?
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17:33.01jonny330i just made a couple of test calls and 2 of them were good now these old ones are back to normal
17:33.11minaguibHey folks. I've been reading a bit about VOIP security, but all of it relates to SIP.  Do SIP headers map 1-1 with IAX2 headers ?
17:33.22ChannelZwell if the reason your calls are failing are because of that, then you can either complain to the telco and see if they will do anything, or leave the 'w' in and wait an extra couple of seconds
17:34.10ChannelZYou might be experiencing a different problem all together, not sure.  Not enough information.
17:34.31ChannelZminaguib: no IAX is totally different
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17:35.26halkunok, so i've decided to start from scratch and go through the asterisk book
17:35.30minaguibChannelZ: I understand that, but does that mean that an asterisk server that forwards a SIP call to another asterisk server using IAX will strip info ?
17:35.42[TK]D-Fenderminaguib: SIP is to IAX2 as MS Word .doc is to OOo .odf.  Same kind of goal, completely different contents
17:35.52halkunbut it assumes I have something called a "Zatpel" which I don't have
17:35.58halkunso most of the examples are useless
17:36.07p3nguinminaguib: Define "forward."
17:36.10[TK]D-Fenderminaguib: There is no "strip".  There is no relationship for the information contained in one, to the other.
17:36.33[TK]D-Fenderminaguib: And there is no "forward".  With * these are 23 absolutely different calls
17:36.37halkunit's all "let's set up asterisk! First load the zaptel module..
17:36.42jonny330Channelz: well i am doing test calls by using dial phone#@internal from the cli and the calls are not always making it to me
17:36.54halkunso I'm pretty much stuck on step 2
17:36.57Qwellhalkun: s/zaptel/dahdi/
17:36.59[TK]D-Fenderminaguib: The only thing they may have in common is taht the CallerID from call A is used as the outbound for call B
17:37.04p3nguinhalkun: replace... what qwell said.
17:37.12halkunI don't have one of those eather
17:37.12AndyGraybealhas anyone used the MCD100-M USB Speakerphone from plantronics? http://www.plantronics.com/north_america/en_US/products/computer/unified-communications-headsets/mcd100-m
17:37.24minaguibOk for example, SIP has well-defined headers such as from/to, but many others in various RFCs (for example caller id presentation, ANI...).
17:37.45halkunI don't have any telephony equipment other then the 3102
17:38.11p3nguinYou'll still want dahdi at least for its dummy module.
17:38.23[TK]D-Fenderminaguib: IAX2 doesn't have "headers".  IAX2 is not a routed protocol like SIP is.  There are no proxies.  There are no special headers because IAX is a CLOSED protocol that isn't "extended" the way SIP gets extended
17:39.27minaguib[TK]D-Fender: So that means a SIP call will lose all its SIP headers except for the ones that map cleanly 1->1 with the IAX protocol ?
17:39.50Qwellminaguib: no, it means that they are 2 *separate* channels.
17:39.51[TK]D-Fenderminaguib: No.. a SiP call coming in to * is a SIP call coming in to *.  It doesn't LSOE anything
17:40.02QwellAsterisk is not a proxy.
17:40.15[TK]D-Fenderminaguib: the fact that * acts on that incoming call and decides to call OUT with IAX2 has absolutely no impact on the originating call
17:40.27[TK]D-FenderminAs qwell said... Asterisk is NOT a proxy
17:40.44minaguibHmm. Perhap I'm confusing things then.
17:41.01[TK]D-Fenderminaguib: There is not route.  There are no headers.  There is no proxy.  THERE IS NO SPOON
17:41.34p3nguinminaguib: Think of it like taking a ride on a bus to the train station, where you get off the bus and get on a train.  Your bus ticket is not good to ride on the train.
17:41.52Qwellp3nguin: but the other way usually does work
17:41.55[TK]D-Fenderminaguib: A SIP call you pump out IAX2 is no different that acoustic coupling a CB radio to a cell phone.  Yes audio passes, but each end has jack shit to do with the other
17:41.59Qwellfor example, the bus is free if you just got off the train
17:42.06minaguibI'm explicitly thinking of a case where a SIP call is dropped into a context that does DIAL(IAX2/foo)
17:42.19Qwell...only in your analogy, of course.  that isn't the case in Asterisk :P
17:42.38[TK]D-Fenderminaguib: If you call out using a PCI FXO interface... where do SIP Headers go?  Bell does not know about "SIP headers".  Your IAX2 call is no different in that respect
17:42.46[TK]D-Fenderminaguib: * is a B2BUA.
17:43.22[TK]D-Fender[13:42]<minaguib>I'm explicitly thinking of a case where a SIP call is dropped into a context that does DIAL(IAX2/foo) <-  IAX2 is completely different tech.  Again, only things in common are CID & Audio
17:44.19minaguibCool I get it
17:45.51minaguibAnother question, security-wise, does IAX2 make provisions for the whole trusted/untrusted peers thing with regards to forwarding/stripping callerid based on presentation ?
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17:47.28[TK]D-Fenderminaguib: There is a rpesentation flag.  Thats about it.
17:49.31minaguib[TK]D-Fender: Are there provisions for asteriskA to deem asteriskB untrusted and if that flag is set, to strip out the CID ?
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17:50.33[TK]D-Fenderminaguib: B Already knows if it trusts A or not and do whatever the hell it wants.
17:50.46[TK]D-Fenderminaguib: All processing = dialplan
17:52.28minaguib[TK]D-Fender: But what about the other way around ? If providerA has to forward a call to providerB but it does not trust them to respect the flag, it must, itself, filter out the CID before forwarding
17:52.42[TK]D-Fenderminaguib: Clearly
17:53.06[TK]D-Fenderminaguib: Don't tell me a secret and then think about whether you should tell me.
17:53.08minaguib[TK]D-Fender: Does * offer dialplan tools for providerA to inspect that flag and strip out the CID ?
17:53.20[TK]D-Fenderminaguib: Yes.
17:54.28anonymouz666sharing a sipfriends table between two boxes, if I call from A to B in first box, but B is registered in the second box, will that work?
17:54.57minaguib[TK]D-Fender: Any tips on which commands will accomplish that ? I'd like to read up more offline instead of asking here.
17:55.26[TK]D-Fenderminaguib: "core show application SetCallerPres" , "core show function CALLERID"
17:55.44minaguib[TK]D-Fender: Thank you very much.  I appreciate the time and info
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17:56.59[TK]D-Fenderminaguib: You're welcome.  Make it a topic for an MLUG get-together or something....
17:57.42minaguibHehehe. No time for LUGs/sleep after kids :)
17:57.46p3nguinIs there a comprehensive description of the presentation types?
17:58.12p3nguinThe basic list provided in the docs assume you know what each thing means.
18:02.11jonny330ChannelZ: i think i got some more info that can help us troubleshooth this problem
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18:18.47ChannelZsorry had to wander away
18:21.52iscariohi, just a question abt sip : i heard sip had problem with NAT. Does it mean that if my sip clients are using nat i would need to set up a sip proxy ? or does sip encounter prb only when the server is behind nat ?
18:26.30[TK]D-Fender~sipnat
18:26.30infobot[~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
18:26.32[TK]D-Fender^^^^^^^^^^^^^^^
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18:33.18jdoehrm. Crap. Should have done the math.
18:33.21jdoeulaw streams add up quick.
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18:46.35niekvlessertwould it be possible to add options (call recording for example) to a call coming from a queue to a queue agent?
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18:52.54jonny330is there anyway to get the inbound caller id from a pots line on a dahdi card?
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18:53.28jonny330i would asume we would have to have caller id on the line right?
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18:55.38jonny330any ideas on how to get inbound caller id from pots lines?
18:55.55paulcjonny330: it should support it out the box, provided you're getting it from the telco
18:56.07paulcsome config file tweaking for format maybe (US vs UK vs Europe etc)
18:56.13jonny330yeah i don't think i am getting it from telco
18:56.24paulcplug a regular phone or caller ID box in to verify?
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18:57.37anonymouz666Using app_while (not using AEL2) is it possible once you hit a condition to BREAK the loop?
18:58.25jonny330k thanks
18:59.46anonymouz666ExitWhile
18:59.47anonymouz666lol
18:59.56anonymouz666pretty smart
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19:22.03yonahwI am having trouble setting up my box to receive calls from my sip provider, I have posted the configuration details at http://pastie.org/957566
19:22.59yonahwi am receiving an error message of "Call from 'userid' to extension 'userid' rejected because extension not found." although I have setup a user in sip.conf which is mapped to an existing extension in extensions.conf
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19:23.22[TK]D-Fenderyonahw: extension 'userid' rejected because extension not found <--- as it states, you have no extension that matches that pattering in yoru dialplan in the context the call is landing in
19:23.45yonahwbut shouldn't it match 's'?
19:23.52ManxPoweryonahw, no!
19:24.02yonahwoh ok
19:24.14ManxPower"s" means "no extension".  It does not mean "any extension"
19:24.25yonahwok this makes more sense
19:24.28[TK]D-Fender~stdextens
19:24.28infobot[~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf.  "s" is also used to make IVRs & macros.
19:24.43yonahwthanks guys
19:26.00wcselbychanisavail always wants to return Status Unknown  ... :(
19:26.25wcselbyhmmmmm......
19:26.26[TK]D-Fendercheckout time, later all
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19:30.31wcselbyTrying to determine the device state (in use, not in use, etc) for members of a queue before they're called.  Not having much luck.  Everything works as expected if the agent answers a call from the queue, they are not sent any more calls from that queue until they hang up.  However, if they answer a call placed to them directly (not through the queue), or if they make a call outbound, then the queue will continue to send them calls and they will get cal
19:32.20edwin_quijadahow can I asociate a recording call with agent and extension took the call?
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19:36.48ACK-NAKhas the CLI originate command moved in 1.6.2?  I know a lot of other ones have
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19:37.25ACK-NAKwhat's its new name?  Anyone?
19:37.32ACK-NAKBueller?
19:38.20wcselbyACK-NAK - don't think it's changed....
19:38.40ACK-NAKHmmm.
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19:39.21ACK-NAKI should be able to type at the CLI, originate [tech... etc] and originate a new call, right?
19:39.23wcselbyif you're trying to originate from an AMI command, I think they added a new originate permission in manager.conf
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19:40.14wcselbyACK-NAK - I think so.
19:40.30wcselbyACK-NAK - if you type orig and then hit tab at the CLI prompt, does it auto-complete?
19:40.48ACK-NAKIt does not.  I'm trying to do something simple such as asterisk -rx originate ...
19:41.01ACK-NAKwcselby: I just get odbc
19:41.40KnucKles_Hi All... Any of you could help me with DAHDI using Tormenta 3 E1 card(Tor 3 - from zapata telephony) ?
19:42.39ACK-NAKFOUND IT:  Now called "Channel Originate"
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20:00.02KnucKles_Does anybody have the driver for tormenta 3 E1 card to run with DAHDI ?
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20:07.55jasonjjohnsonjrDoes anyone have any recommendations for a simple CDR tool? I looked at freeside and a2billing but they seem like overkill just to give users access to CDR data.
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20:09.43KnucKles_jasonjjohnsonj: Look at http://www.areski.net/asterisk-stat-v2/about.php
20:09.51KnucKles_It is a good tool
20:10.48jasonjjohnsonjrKnucKles_: It shows that the last release was 2005. Are there any issues with it working with 1.6.2?
20:12.13KnucKles_Yes... I had tested into 1.6.1
20:12.19KnucKles_works ok!
20:12.41KnucKles_opss.... No issues!!!
20:12.55jasonjjohnsonjrI will take a look. Thanks,
20:13.00KnucKles_no problem!
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20:18.22jsidhuanyone know if we can use the Vonage Business plus account as a sip trunk?
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20:34.53devmodI was trying to add a local ext as a queue member, but I see invalid when doing queue show:  Local/1000@agents (dynamic) (Invalid) has taken no calls yet
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21:29.20wcselbycan hardware echo cancelation modules be bought and installed on digium boards that we already own?
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21:29.36Qwellwcselby: sure.  find a reseller that will sell it separately
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21:37.11d_preston215What does switchtype=national mean?
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21:41.23wcselbyis _**XXXX, supposed to also match *8., ?
21:42.25wcselbynevermind, misread my cli
21:45.32kazaa_litehi all
21:45.46kazaa_litehow can i configure asterisk to start at some very high priority?
21:45.55ManxPowerd_preston215, It means the switch is provisioned for National ISDN 2 service
21:46.06ManxPowerIt is the most common PRI setup in the USA and Canada
21:46.12d_preston215ok
21:47.07ManxPowerkazaa_lite, "man asterisk"  pay special attention to the "pseudo realtime" switch.  Don't expect it to do anything other than cause problems.
21:48.13d_preston215PRI T1 cards for the most part use pri_cpe as signaling?
21:48.36ManxPowerd_preston215, when connecting to a telco, yes
21:49.02d_preston215Would it be the same if I was using a PRI appliance such as a redFONE?
21:56.47ManxPowerit would depend on how the redFONE was configured
21:57.13ManxPoweryou might need a crossover t-1 cable if you are connecting Astersik to a non-teleco port
21:57.31d_preston215Made my own T1 crossover cables.
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21:58.21ManxPoweryou are sure they are not ethernet crossover?
21:58.35d_preston215I'm sure.
21:59.08d_preston215Basically switch out pins 4&5 with pins 1&2.
21:59.36d_preston2151 & 2 with 4 & 5 I mean.
22:00.23d_preston215http://blog.elastixdepot.com/2009/11/19/configuring-a-redfone-fonebridge2-with-elastix-part-1.aspx
22:00.40d_preston215Unless the instructions for the cable on here are wrong.
22:01.43d_preston215But in any case, actual lines from the teleco would be signaled normally with pri_cpe.
22:01.48d_preston215If I'm correct.
22:02.27d_preston215Just that any loop back port would have to be signaled with pri_net.
22:04.08edwin_quijadahow can I figure out in a queue what agent took the call and extension for recording the call??
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22:16.50rare1980_hi all i am trying to load modprobe ztdummy
22:16.56rare1980_but i am getting msg that
22:17.03rare1980_FATAL: Module ztdummy not found.
22:17.10QwellIs it installed?
22:17.19rare1980_please can any one guide me?
22:17.41rare1980_Qwell: yes i have install zaptel
22:17.49rare1980_on ubuntu kernal 2.6
22:18.02QwellHow did you install it?
22:18.32rare1980_i have download this ---- svn checkout http://svn.digium.com/svn/zaptel/branches/1.4 zaptel
22:18.44rare1980_./configure
22:18.44rare1980_make
22:18.45rare1980_sudo make install
22:18.45rare1980_sudo modprobe ztdummy
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22:18.54Qwellshow me the output of those commands.
22:18.56Qwell~pastebin
22:18.56infobot[~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
22:19.00rare1980_sure
22:19.09rare1980_plz 1 sec
22:19.35rare1980_u mean output of all installation?
22:19.38rare1980_results
22:19.47Qwellthe entire output from all of those commands
22:20.10rare1980_rite
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22:20.57rare1980_i am using putty.. i can't see all installation result on it.. is there any log file from where i can get the results?
22:21.58rare1980_?
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22:23.33rare1980_qwell? please tell me is there any installtion log file
22:23.38Qwellno
22:25.06rare1980_ok then let me get those results which i can see :)
22:25.56rare1980_http://pastebin.com/zW7Ma3Jx
22:26.08rare1980_here are the results on pastebin
22:28.20Qwellyou didn't install it for the correct kernel version
22:29.42rare1980_how can i install correct kernal version?
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22:31.21rare1980_Qwell: please can u guide me a bit?
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22:39.09rare1980_atleast guide me which kernal ver is comptble with zaptel
22:39.10rare1980_?
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22:39.37Deeewaynerare1980_, did you run make config like the output told you to do?
22:40.19Deeewayneoh nm
22:40.24XTeohi
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22:41.07XTeoit's necesary install Mysql with Asterisk, (without Freepbx or another web gui)  ??
22:42.14decaffeineXTeo, no
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22:43.45decaffeinegood night all
22:45.04XTeook
22:45.08XTeothans
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22:47.59devmodHow can I check before dialing if a sip dev is online or not?
22:48.25hardwirehmm.. getting duplicate DTMF using AGI get_data.. seeing correct results with dtmf debugging
22:48.37hardwirejuts the result from get_data is wonky if I dial too fast.
22:49.00hardwireso.. rfc2833 works fine.. can verify that in the pcap as well as the DTMF debugging via logger.conf
22:49.08hardwireI see the begins and ends.. they don't overlap
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22:52.07nnystupid question, is Asterisk sip notifyringing=no suppose to supress ringing notifications for hints?
22:53.16nnyand if so, is there any reason why it wouldn't work?
22:53.36ManxPowerhardwire, make sure you don't have rfc2833compensate=yes
22:53.47ManxPoweras well as make sure relaxdtmf is not set
22:54.16rare1980_i am unable to install kernal 2.6.26--- apt-get install linux-source-2.6.26
22:54.32rare1980_on ubuntu it says
22:54.36rare1980_no package found
22:57.18nnyright now, when I dial in with Asterisk sip notifyringing = no I get http://pastebin.org/227652
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22:57.42hardwireManxPower: indeed.. since DTMF debugging is correct that appears to be a non issue (it's also not set)
22:57.58nnyand the blfs on the phones flash... I have another system I can't test right now, but I thought it disabled ringing hints.. is it wrong or a bug with the newer version of asterisk i am running?
22:58.00hardwireboth are not set
22:59.02hardwireManxPower: I'm about to test something.. the playback file doesn't last very long.. I'm going to add a ton of silence to the end
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23:10.51nnyhmm only found a little info on google, and the same query (still get ringing notification) anything I can do to diagnose further?
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23:13.05nny[TK]D-Fender: can I pick your brain for a sec?
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23:13.49[TK]D-Fendernny: sure
23:15.38nny[TK]D-Fender: is Asterisk sip notifyringing = no suppose to keep asterisk from doing "Extension Changed 102[hints] new state Ringing for Notify User 101"
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23:17.33[TK]D-Fendernny: It's supposed to suppress SIP 180 Ringing messages between the invite & answer/abort on that channel
23:18.01sawgoodexten => _X.,1,Dial(SIP/itsp-01/${EXTEN})
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23:18.19sawgoodhow can I 'change' this statement to 'must' match a 9 in front
23:18.37[TK]D-Fendersawgood: exten => _9X.,1,Dial(SIP/itsp-01/${EXTEN:1})
23:18.42[TK]D-Fendersawgood: Shove a 9 in front
23:18.44sawgoodthank you
23:18.49sawgoodperfect answer
23:18.52sawgoodthank you so much
23:19.04nny[TK]D-Fender: ahh. I misuderstood. Is there a way to suppress ringing states for phones subscribed to hints?
23:19.06[TK]D-Fendersawgood: And as you hopefully noted I STRIPPED it off of the actual number being passed to that peer
23:19.33[TK]D-Fendernny: Hmm... odd request.... Nothing I can think of short of vi
23:19.38sawgoodty
23:20.18[TK]D-Fendersawgood: "Dial 9" prefixes are soooo 1980.....
23:21.04nny[TK]D-Fender: have a sidecar that blows up when all the phones ring at the same time, and another phone that has 2 blfs for other users. In both cases the blf flashing when the call comes in is more of a distraction than information for the users
23:21.39nnyblows up = flashes/lights up all buttons at the same time ;)
23:21.52nnyall though the other way would be damn interesting
23:22.02[TK]D-Fendernny: Sounds like An Aastra 5i series :)
23:22.09nny[TK]D-Fender: Cisco spa500
23:22.21[TK]D-Fendernny: Shit-acular!
23:22.30[TK]D-Fendernny: Craptastic even...
23:22.33ManxPowerReal Men use Polycom (tm)
23:22.50nny[TK]D-Fender: so I assume polycoms have a way to suppress that?
23:23.06[TK]D-Fendernny: No... Polycom's simply don't have a hissy fit over it :)
23:23.07nny[TK]D-Fender: or do they not flash blfs when a call comes in...?
23:23.08ManxPowernny, I doubt it, but I don't think they crash
23:23.13nnynot hey dont crash ha
23:23.15nnythey*
23:23.33Naikrovekpolycom > *
23:23.39nnysorry miscommunicated, the fact that the blf blinks in a network where all phones/most phones ring on an incoming call is bad
23:23.47nnythis would be the same for all phones I assume
23:24.05nnysuprised that you can't disable it though
23:24.09nnyat least not in *
23:24.18[TK]D-Fendernny: 2 solutions : vi chan_sip.c  OR run it through a proxy of some kind that lets you filter.
23:24.30WIMPynny: On the SAP962 you can define the visuals.
23:24.41nnyWIMPy: i'll check the 508 thanks
23:25.09WIMPyIf it's supported, just change the pattern to 'off'.
23:25.25nnyWIMPy: gotcha, was gonna look over the interface again if asterisk could not suppress it
23:25.28WIMPyI think that was the only thing I liked about the SPA.
23:25.59nnyI dunno, overall the phones work rather well
23:26.18nnydo the polycoms have backlights now?
23:26.23WIMPyI like it so much, it's not even connected any more.
23:26.56[TK]D-Fendernny: Most of the newer modules, yes
23:27.01WIMPyI've never seen a Polycom. Would like to, but I'm not going to buy one, just out of interest.
23:27.12[TK]D-Fendernny: 335,450,550,560,650,660 all do
23:27.38nnythat's better at least
23:28.07nnyhere's a question that popped up. Anyone implemented BLF state changes based on duration of a call on hold? if so what phone model?
23:28.35nnySome client had a Avaya or something that did that, they wanted the feature but I haven't had the chance to look around
23:28.55Naikrovekhow would you do that?  /me ponders.  not being sarcastic.  voip doesn't have the notion of "lines" like analog and digital systems sometiems do
23:29.39Naikrovekmaybe you could have something that notified the phone via message or something when a parked call had been waiting too long
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23:29.44Naikrovekor perhaps just call the receptionist back
23:30.00nnyNaikrovek: apart from parking lot queue times, nothing I know of yet
23:30.07nnyand all that does is return the call to the user
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23:30.13Naikrovekyeah
23:30.21WIMPyhold or park?
23:30.36nnyhmm well, the avaya was hold, but park would be more useful
23:30.54WIMPyMakes sense that way.
23:31.12nnyit was more of an answer to a simple system, numerous lines on hold, which one has been there the longest kind of thing (as far as I know)
23:31.24Naikrovekyeah that's valid
23:31.48nnyWIMPy: yeah the cisco SPAs show promise, although their engineers craptastically add bugs to every firmware update...
23:32.02Naikrovekpolycom phones can kinda handle that.  they can show you how many calls you have on hold and how long each has been on hold.  doesn't do sorting based on hold time or anything though
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23:32.20nnyNaikrovek: yeah the SPAs will show you how long a line has been on hold for too
23:32.21WIMPynny: I don't see any promise there.
23:32.38nnyWIMPy: well, it's not game breaking bugs, so far it
23:33.09WIMPyI think they're totally crappy from the users point of view.
23:33.38nnyit's been sidetone missing, (patch other things fixed) firmware provisioning via tftp doens't catch changes to the xml files (fixed). The latest hasn't show any issues. To be honest the Cisco SPAs are 6 months old or so
23:33.39Naikrovekthe SPAs?
23:34.09WIMPyNice hardware design (from the outside) but usability went past them.
23:34.12nnyI dunno, I have 40 or 50 SPAs in the field at various sites, and only a half dozen minor requests or features that they want
23:34.22nnyno real complaints per se
23:34.35nnyand even the bugs were only known and handled by me, they didn
23:34.40nny't even realize sidetone was missing
23:35.07nnyI am a bigger nazi about the phones than anyone else heh
23:35.38nnythen again this is the Cisco SPAs predominately, the linksys ones lacked some useful features that the ciscos do normally
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23:36.13WIMPyJust ask someone used to a modern telephone system and they will usually be quite unhappy with any voip phone.
23:36.35nnylike customizable line keys (change to BLF, speed dials) and programmable soft keys (which is still not perfect) they end up doing the button combination on a new line
23:36.44nnymy number one "complaint" is shared line appearance related
23:36.55nnyblfs that show parked call states pretty much rides that out
23:37.04WIMPyYes, complete bullocks.
23:37.13nnymost people are used to saying pick up line one etc
23:37.21WIMPy(the SLA stuff)
23:38.01WIMPyPeople stuck in the 70s that is.
23:38.24nnyyeah
23:38.44nnytbh I tell people it's a hold over from key systems, there is no reason to use it on a modern VoIP system
23:38.53nnyand then show them park ;)
23:39.03nnyi even numbered the parking lots 1-20 heh
23:39.11nnyso they can still say "pick up one"
23:39.16WIMPyUnfortunaletly people in non-american countries usually aren't. ;)
23:40.12ChannelZHmmm.  Does the ksoftirqd process just sit there consuming ~15-20% CPU for anyone else?
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23:44.02nnyhmm on the whole change led states thing. With the Line Key LED Pattern changes, plus a sip header modification, you could in theory do something with park + led patterns. probably have to do some "manual" parking stuff though
23:45.00nnynot sure how much control i have over the park application, meh, back to my first issue
23:45.12nny[TK]D-Fender: WIMPy thanks for the heads up
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23:47.13devmodcan reliably I use ChanIsAvail on SIP endpoints registered to asterisks to know if they are online or not?
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