IRC log for #asterisk on 20090605

00:00.36dlynesrj175: ok...if I don't reply it's because i'm just sitting down to supper....wife just finished cooking
00:00.51dlynesrj175: doesn't mean I won't reply...might have to wait 1/2 hour for a reply
00:00.58rj175dlynes: haha its 1AM here :P
00:01.23rj175dlynes: i get this
00:01.35rj175dlynes: [Jun 5 01:12:39] WARNING[4641] chan_iax2.c: Call rejected by 217.14.138.130: No such context/extension
00:01.57rj175dlynes: thats for a outgoing call
00:03.12dlynesrj175: You're in control of both ends of the iax channel?
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00:04.52rj175dlynes: ermmmm...?
00:12.40*** join/#asterisk CCIE20156 (n=bassem22@188.132.24.16)
00:12.49CCIE20156hello all
00:13.44CCIE20156could someone assist me please with the meetme application, it's saying unable to load pseudo device, while using dahdi, i have to digium hardware installed, and i'm trying to join a conference room via SIP.
00:14.01CCIE20156using 1.4.25
00:17.29CCIE20156hello ?
00:19.22CCIE20156could someone assist me please with the meetme application, it's saying unable to load pseudo device, while using dahdi, i have to digium hardware installed, and i'm trying to join a conference room via SIP. i'm using asterisk v1.4.25 on FC10
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00:19.45CCIE20156could someone assist me please with the meetme application, it's saying unable to load pseudo device, while using dahdi, i have to digium hardware installed, and i'm trying to join a conference room via SIP. i'm using asterisk v1.4.25 on FC10
00:21.20CCIE20156anyone home ?
00:21.33QwellDO NOT repeat the same question repeatedly.
00:21.38QwellDO NOT msg people without asking.
00:21.51CCIE20156sorry about that
00:21.58*** part/#asterisk seb- (n=seb@li30-51.members.linode.com)
00:22.03CCIE20156i thought maybe no one was looking
00:22.08QwellNo, you aren't.  You assume your time is more valuable than others.
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00:22.50CCIE20156i said i was not sure anyone was looking
00:22.58CCIE20156or my messages were not getting there
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00:23.47CCIE20156you have no right to judge me or anyone for that matter
00:25.44dlynesrj175: There's two boxes that are talking over the iax channel.  Do you have control over both boxes?
00:25.54rj175no just mine
00:26.00rj175i have gt outgoing calls to work thought:D
00:26.20dlynesrj175: Did your provider tell you to send your iax calls into a specific context?
00:26.40rj175i changed outgoing calls to go throught SIP instead
00:26.58xorldoes the POLYCOM SOUNDSTATION IP 3000 support SIP?
00:27.23dlynesrj175: I suspect you were trying to share dialplans between the two sides, and your upstream wasn't cooperating
00:27.43rj175i dnt get any audio back but hey :P
00:29.37dlynesrj175: if you're not getting audio back it's because your rtp isn't getting across your firewall
00:29.37rj175dlynes: now i just need to get incoming calls setup
00:29.46dlynesrj175: are you trying to port forward udp 5060?
00:29.58rj175my server is DMZ'd
00:30.24dlynesrj175: are you getting call progression?
00:30.56rj175as in? when i call my number i get "the number you have dial is not in service"
00:31.08dlynesOk, so you're getting feedback, then
00:31.26dlynesCan you pastebin a log at core verbosity of 10 or higher of an outgoing call?
00:31.30dlynesUsing sip?
00:31.53rj175culd u explain how i do that :$
00:32.01dlynes~pb
00:32.02infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
00:32.14dlynesJust copy and paste the output to that website
00:32.27rj175i mean how i get a verbisity of 10 or higher
00:32.28dlynesAfter you hit 'submit', it'll give you a link to your paste
00:32.37dlynesPaste the url to that pastebin paste in the channel
00:33.53rj175u want the logs of outgoing calls? cause outgoing works, just incoming isnt
00:34.22dlynesrj175: you just said your outgoing calls were giving you 'number you have dialed is not in service'
00:34.31rj175no thats my incoming
00:34.34dlynesah
00:34.41dlynesok...gimme log of incomign then
00:35.36rj175dlynes: hope this is what you want http://pastebin.com/m53f3a5d
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00:36.00dlynesAnd I'm guessing you're proficient at networking...just not voip, right?
00:36.04dlynesrj175: No...that's not it
00:36.23dlynesrj175: do you have errors and warnings turned on in /etc/asterisk/logger.conf?
00:37.03rj175dlynes: i have full => notice,warning,error,debug,verbose
00:37.24jayteeexit
00:37.39jayteeooops
00:38.02jayteesorry, wrong window :-)
00:38.45dlynesrj175: Let's see your sip.conf file and extensions.conf file then...make sure you scrub your passwords from the file
00:39.44rj175dlynes: it seems like nothins there...culd it be cause im using freepbx?
00:40.02dlynesrj175: I asked you before I started with you if that's what you were using
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00:40.19dlynesrj175: freepbx/trixbox
00:40.22rj175dlynes: im sorry :( seems like this whole thing is confusing me
00:40.29dlynesrj175: freepbx is quite old....it's now called trixbox
00:40.51rj175o
00:40.52rj175ok
00:41.03dlynesrj175: You're better off getting help with issues like that on #freepbx
00:41.16dlynesrj175: the reasoning is that most of us are not familiar with freepbx style configs
00:41.37dlynesrj175: You might also try #elastix if it exists, because it also uses freepbx under the covers
00:41.59rj175dlynes: wuld u sudjest that I just scrap freepbx or use the command line?
00:42.11drmessano[20:41] <dlynes> rj175: freepbx is quite old....it's now called trixbox  <-- NOT true
00:42.18dlynesrj175: I would suggest you use whatever you feel more comfortable with
00:42.33dlynesdrmessano: Oh...is trixbox the new name of amp, then?
00:42.42drmessanoTrixbox is a project using a fork of Freepbx (at one time it actually used freepbx)
00:42.51drmessanoIts a full ISO pbx distro
00:43.00drmessanoHas nothing to do with FreePBX
00:43.19dlynesdrmessano: ah...so it's an alternative to elastix, then
00:43.27drmessanoYes
00:43.43rj175dlynes: the freepbx channel seems to be dead haha
00:43.44drmessanoElastix also uses FreePBX, as does AsteriskNOW
00:44.02dlynesdrmessano: oh...thought AsteriskNOW was written from scratch
00:44.27drmessanodlynes: AsteriskNOW 1.5 uses FreePBX for its configuration GUI
00:44.42dlynesdrmessano: i've tried elastix's interface and the freepbx interface that comes with elastix as well..it seems to be quite restrictive
00:44.59drmessanoIts meant to serve a task
00:45.38rj175can i ask the fact i get this american voice sayin the number i have called in not in service, is that a astrisk thing ?
00:45.52dlynesrj175: it's probably a freepbx thing, yeah
00:46.03rj175least im gettin to my box then :P
00:46.04dlynesrj175: you can get british voices as well
00:46.49drmessanorj175: what distro are you using?
00:46.57rj175astrisknow
00:47.15drmessanoSeems like that would have saved a lot of time 2 hours ago
00:47.28dlynesrj175: there ya go...there's also #asterisknow and #asterisk-gui
00:47.46drmessano#asterisk-gui is for a specific project
00:47.57drmessano#asterisknow and #freepbx would be your best bets
00:48.03dlynesdrmessano: well, I don't know much about these guis...I'm having to deal with one right now, and not really big on it
00:57.55fileback in my day we had a console where you typed stuff and we were darn happy to have that
00:59.06RyanRRwhatever old man
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01:14.43killfillhey
01:15.44killfilli jsut loaded the wctdm module with debug=1. im seeing in var-log-messages "RING on 1/1, NO RING on 1/1, etc, etc"  withouth any 'polarity' string when hangs up occur.
01:16.49killfillWith this info, can i be 100% sure that the telco is not giving me "disconnect supervision"?
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01:41.43killfill:S
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02:04.11srf21cno ring tone on outgoing iax calls via lax.teliax.net server.
02:04.26srf21cbut the calls connect fine.
02:04.32srf21caudio is there once the other end picks up.
02:05.00srf21cDoes sound like a service provider issue?  that is my hunch.
02:05.11srf21cIf there's anything that I can try on my end, I'm all ears.
02:05.12tfrewsrf21c: paste your iax conf
02:05.25srf21cOutgoing calls via a sip provider work fine, I get the outbound ring tone.
02:05.28srf21ctfrew: stand by
02:05.43tfrewsrf21c: i use sip with teliax because it is more stable
02:06.02tfrewthey use an iax2 driver, there sip connection works better
02:06.41srf21chttp://pastebin.ca/1448264
02:07.07srf21ctfrew: sip @teliax is more stable?  interesting....
02:07.26tfrewyep D:
02:07.36tfrewfrom my experiance
02:07.39tfrewbrb
02:07.41srf21cI thought iax was what they started with.
02:07.43srf21cok
02:08.41tfrewpastebin.ca seems to be broken
02:08.49tfrewthat, or my internet just broke
02:08.55srf21cwha?
02:09.04srf21cwhat's another good one
02:10.37tfrewsrf21c: paste it into pm
02:10.56tfrewor use pastebin.com
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02:37.45dlynesDo iax channels have a setting that works like the call-limit parameter does for sip channels?
02:38.32dlynes~pb
02:38.33infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
02:38.44dlynessrf21c: There's a whole bunch for you
02:39.07dlynessrf21c: you can also try ipv4.pastebin.ca and ipv6.pastebin.ca in case your issue is an ipv6 related issue
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03:16.11srf21cdlynes: grassy-ass...I'll bookmark 'em
03:16.43srf21crafb.net is no more btw.
03:17.09srf21c~pb
03:17.10infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
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04:25.01joakoI can get ${SIP_HEADER(Diversion) gives e.g.: Diversion: <sip:+13523782778@4.55.14.163:5060>;privacy=off;screen=no;reason=unconditional; counter=1
04:25.36joakoBut how can I get just "unconditional" from it?
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04:47.28AlmightyOatmeal[Jun  4 23:46:27] NOTICE[65330]: chan_sip.c:7759 sip_reg_timeout:    -- Registration for '...@sip.broadvoice.com@sip.broadvoice.com' timed out, trying again (Attempt #57) <-- that is getting really annoying :(
04:47.50AlmightyOatmealit was working just fine until my internet went out, restarted * and baam
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05:00.51spirihey, Im running debian 5.0 here on openvz and everything is working great except for the emailing of voicemail messages... Im think that this may not actually be the fault of asterisk but some mail settings in debian.. has anyone come accross this issue?  Ive been scouring google but have only come accross some comments
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05:02.24AlmightyOatmealspiri: * uses sendmail iirc, try sending a test message with sendmail from the cli and see if that works
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05:02.40AlmightyOatmealif it works, then its an * problem, if it does not work then it is your linux distro settings, usually a smtp issue
05:02.47spiriok cool
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05:21.35spiriAlmightyOatmeal: thanks.. I had to run dpkg-reconfigure on exim
05:21.43spiriit was setup only to work locally
05:21.46spiriderf
05:26.56AlmightyOatmealnp
05:46.46[TK]D-FenderAll * does is call a sendmail binary and whatever it does is not *'s problem
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06:47.05AlmightyOatmealcan i change the default playback volume for MOH without calling to a seperate application?
06:47.17AlmightyOatmeallike can i adjust the volume on the native playback
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07:33.59AlmightyOatmeali think i got it with madplay
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07:38.36scruzhello all
07:40.06scruzi've an extension properly defined in my extensions.conf, and asteriskwin32 is throwing up an error for it
07:41.05scruzhttp://www.mibbit.com/pb/5plHr2
07:42.20scruzhere's the error: *CLI> Jun  5 08:33:25 NOTICE[4860]: chan_iax2.c:7059 socket_read: Rejected conne ct attempt from 192.168.1.106, request '2000@firstcontext' does not exist
07:42.30scruzdoes anyone know why?
07:43.15AlmightyOatmealheh
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07:46.12scruzAlmightyOatmeal: i've checked - to make sure i
07:46.41scruzwasn't skipping anything. no idea why. if i place a call to 101, it works. calls to 200Z don't
07:47.00scruzyet both are defined in the context
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07:55.02AlmightyOatmealscruz: that error shows you dialing extension 2000 if i'm not mistaken
07:55.54AlmightyOatmealnot 200Z
07:57.02scruzyeah, isn't Z supposed to represent any digit from 1-9?
07:57.48AlmightyOatmealnot that i know of
07:57.53scruzhold up, one minute
07:59.06scruzmy bad. if you prepend the extension with _, you can use X to represent any digit from 0-9, Z for any digit from 1-9
07:59.44scruzit should have read: exten => _200X,1,Dial(IAX2/${EXTEN})
08:00.08AlmightyOatmealmuch better
08:00.16scruzi
08:00.25scruzi'd completely forgotten about the _
08:01.55AlmightyOatmeali actually don't know squat about the rules or much at all, i was just playing along but the error did look familiar o:P
08:02.14kaldemarZ is 1-9
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08:03.03scruzprepending the underscore allows asterisk to treat the extension a bit like a regex
08:03.15AlmightyOatmeals/rules/dialplans.. this isn't a firewall lol
08:03.20AlmightyOatmealscruz: no no, i get that part
08:03.53AlmightyOatmeali know whats going on, but i wasn't sure about actually resolving your issue, thankfully you were hehe
08:04.32scruzhaven't worked with asterisk in a while. trying to teach someone the basics
08:04.40scruzwell, later then. take care everyone
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08:06.34kaldemar"a bit like regex" is called pattern. and unfortunately it's nothing like regexp. :)
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08:21.28ectospasmyou know you've been working on this stuff too long when the first thing you type into any text input field is "asterisk"
08:21.45AlmightyOatmeal:P
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08:41.29rajmohanhi
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08:42.00rajmohananyone here can help me with asterisknow installation pls
08:42.51rajmohani burned the asterisknow to cd and boot from cd is not working
08:42.54rajmohanpls help
08:42.57rajmohanwhat is the problem
08:43.01rajmohan?
08:43.13rajmohanso far i have burnt it 6 six times
08:43.20rajmohannone of the cd is working
08:44.41kaldemarhave you been to #asterisknow ?
08:44.55rajmohanyou mean?
08:47.02kaldemarthe irc channel that supports asterisknow.
08:47.49rajmohaniam not getting help from there pls
08:47.55rajmohanno one has replied there
08:48.04kaldemaryou'll be better off asking in there, most people here don't use GUIs or distributions and can't help you with that.
08:48.24rajmohanoh ok pls
08:48.39AlmightyOatmealthe gui actually is harder in my opinion..
08:48.47rajmohani am a newbie pls
08:48.58rajmohani have installed asterisk in ubuntu
08:49.06rajmohanhow do i check if it is running or not pls
08:50.21kaldemar~book
08:50.22infobot[~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
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08:51.10kaldemarrajmohan: start with that book. if "asterisk -r" in shell gives you a CLI for asterisk, then it's running.
08:52.13CaveyCoUkMorning all.  Silly question - can Asterisk handle 16 analogue lines? Could someone point me towards a reseller or brand that would sell hardware capable of supporting this
08:52.14rajmohankaldemar: thanks pls
08:52.19rajmohani will check with that pls
08:53.36kaldemarCaveyCoUk: yes it can. either with analog cards or with a channel bank. http://www.digium.com/en/products/analog/
08:53.44CaveyCoUkThank you
08:54.08CaveyCoUkHad been googling, but most concerned Voip and we are a bit rural to have a reliable and decent upload speed :(
09:02.00joakoI need to run sed on a variable, is there a better way than AGI or app_backticks?
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09:05.21LtScarrey everyone
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09:05.52LtScarri hava a question about queues
09:06.29LtScarris it possible to have a queue without agents having to login in?
09:06.47kaldemarjoako: what are you actually doing to the variable? maybe you don't need sed at all?
09:08.51LtScarrsorry for the typos im on a phone keypad
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09:11.49joakokaldemar: I need the call diversion reason, e.g Diversion: <sip:+13523782778@4.55.14.163:5060>;privacy=off;screen=no;reason=unconditional; counter=1 I need "unconditional" as a variable
09:17.11phix:D
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09:23.35kaldemarjoako: tough to tackle with all the ;'s, but should be doable with CUT.
09:24.54rajmohanhi, can any one give asterisk installation guide on ubuntu
09:25.43joakokaldemar: Is cut not based on a fixed # of characters?
09:26.20kaldemarno, cut is based on a delimiter and fields
09:26.39joakoEither way I already wrote a 9 line bash AGI scipt...
09:29.18kaldemari thought of Set(var=${CUT(header,reason=,2)}) and Set(reason=${CUT(var,\;,1)}) or something like that.
09:31.18joakoEither way after I go through all this I figure out AT&T mobility only sends reason=unknown for all calls
09:31.37kaldemarheh, that happens. :)
09:32.17joakoactually reason=unconditional for unconditional and reason=unknown for conditional
09:33.58CaveyCoUkThanks again
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09:35.37joakoAny idea can asterisk handle it if I have two ${SIP_HEADER(Diversion)... only seems to do anything with the first header it recieved
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10:21.15iksikhey
10:21.49iksikis it normal, that my asterisk registers new account (on my operator server) with each restart? :|
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10:32.39cofeineSunshineorg.asteriskjava.manager.TimeoutException: Timeout waiting for protocol identifier
10:32.53cofeineSunshinei get this error, when i try to connect to asterisk
10:35.46AlmightyOatmealasterisk java?
10:38.30cofeineSunshineyes
10:38.37cofeineSunshine0.3.1 lastest table version
10:38.52cofeineSunshinefrom here: http://asterisk-java.org/development/xref/org/asteriskjava/manager/internal/ManagerConnectionImpl.html line 516
10:38.59AlmightyOatmealnifty
10:39.45cofeineSunshinehere is my class: http://p.defau.lt/?ZDlZiIp5y5AitWqEcF_ydw
10:40.03cofeineSunshinemanagerConnection.login();
10:40.12cofeineSunshinethere i got an exception
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11:11.01iksikhm, ok... asterisk is limiting length of caller id presentation?
11:11.34iksiki'm setting it up on: 6678 - ok, works fine, but longer number doesn't present, it's just shown: UNKNOWN on my cell phone
11:12.30AlmightyOatmealiksik: cell phones only carry over the number.
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11:12.53AlmightyOatmealPOTS carries name and number, unless coming _from_ a cell phone then it only gets the number as well
11:13.04AlmightyOatmealprivacy and etc, call your cell phone carrier for details
11:13.39AlmightyOatmealbut that reminds me that my caller ID isn't going through on POTS, wonder if my sip carrier somehow filters it
11:14.13kaldemariksik: you can't usually set an arbitrary caller id to PSTN calls. telco's usually have some sort of prevention for caller id spoofing.
11:15.15kaldemariksik: afaik, asterisk doesn't have any hard coded limits that are not originated from a protocol specification.
11:15.17iksikok, it works now ;P
11:15.19iksikmy bad ;)
11:15.49iksikit looks like spoofed numbers works great in Poland
11:16.04AlmightyOatmealkaldemar: i have callerid=blah set, but PSTN simply reads it as out of area caller while keeping my number
11:16.09AlmightyOatmealiksik: heh
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11:17.09kaldemariksik: lucky you. :)
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11:23.00AlmightyOatmealiksik: if you don't mind me asking, how are you setting your callerid?
11:24.14iksikfor now I have no idea... I mean, I don't know which option is responsible for this... that is why all possible options (username, authusername, callerid, account name) are just my CALLER ID Number ;P
11:24.25iksiki'm noob, don't kill me :X
11:25.03leifmadsen~kill iksik
11:25.04infobotACTION shoots a excited anti-photon gun at iksik
11:25.08iksikhehe
11:25.11AlmightyOatmealha
11:26.00AlmightyOatmeali assume setting callerid=blah in a specific user's context will work on outbound calls?
11:29.37lanceyAlmightyOatmeal yes
11:29.45lanceycallerid = "cidname" <cidnum>
11:29.45AlmightyOatmealstrange
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11:30.19AlmightyOatmeali just added exten => _1NXXNXXXXXX, 1, Set(CALLERID(name)=Hot Sex Hotline) to my outbount context
11:30.23AlmightyOatmealthat should work as well, right?
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11:31.00lanceylol
11:31.00AlmightyOatmealits an inside joke between my gf and me, long story :P
11:31.00lanceyi dunno if you don't have to quote it in "
11:31.18AlmightyOatmealah hmm
11:31.36Zeeekhello world
11:31.45lanceyhello Zeeek
11:32.11Zeeekanything interesting happen in the last two weeks?
11:32.56AlmightyOatmealZeeek: god made me his right hand man, otherwise you didn't miss much
11:33.42AlmightyOatmeali was really thinking of something funnier but it didn't happen :(
11:33.48Zeeekthen we're all set!
11:35.05leifmadsenAlmightyOatmeal: not bad for morning
11:35.20AlmightyOatmeal:D
11:35.20leifmadsenZeeek: I now own the nude picture of your first lady!
11:35.33AlmightyOatmealwell thats awkward
11:35.50leifmadsenAlmightyOatmeal: lol... well, not HIS first lady, but that of his country :)
11:37.25AlmightyOatmealooh lol
11:38.00ZeeekI gave a customer a whole directory of photos, then realized there were a few from our hot tub. Talk about awkward!
11:39.20leifmadseneep :)
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11:50.18rajmohanhi, when i do make install in asterisk folder
11:50.30rajmohaniam getting the following error
11:50.51rajmohansudo make install
11:50.51rajmohanmake[1]: Entering directory `/usr/src/asterisk-1.4.10/menuselect'
11:50.51rajmohanmake[2]: Entering directory `/usr/src/asterisk-1.4.10/menuselect'
11:50.51rajmohanmake[2]: `menuselect' is up to date.
11:50.51rajmohanmake[2]: Leaving directory `/usr/src/asterisk-1.4.10/menuselect'
11:50.52rajmohanmake[1]: Leaving directory `/usr/src/asterisk-1.4.10/menuselect'
11:50.54rajmohanmenuselect/menuselect --check-deps   menuselect.makeopts
11:50.56rajmohanGenerating embedded module rules ...
11:50.58rajmohanmake[1]: Nothing to be done for `all'.
11:51.02Pan3Daaaaahhhhh my eyes!!!!
11:51.02rajmohanmake[1]: Nothing to be done for `all'.
11:51.06rajmohan<PROTECTED>
11:51.08rajmohanchan_iax2.c: In function ‘jb_error_output’:
11:51.10rajmohanchan_iax2.c:769: warning: format not a string literal and no format arguments
11:51.12rajmohanchan_iax2.c: In function ‘jb_warning_output’:
11:51.14rajmohanchan_iax2.c:781: warning: format not a string literal and no format arguments
11:51.17Pan3Drajmohan: pastebin
11:51.20rajmohanchan_iax2.c: In function ‘jb_debug_output’:
11:51.22rajmohanchan_iax2.c:793: warning: format not a string literal and no format arguments
11:51.24rajmohanchan_iax2.c: In function ‘complete_dpreply’:
11:51.26rajmohanchan_iax2.c:5376: warning: ignoring return value of ‘write’, declared with attribute warn_unused_result
11:51.29rajmohanchan_iax2.c: In function ‘find_cache’:
11:51.33rajmohanchan_iax2.c:9757: warning: ignoring return value of ‘write’, declared with attribute warn_unused_result
11:51.36rajmohanIn function ‘open’,
11:51.38rajmohan<PROTECTED>
11:51.40rajmohan/usr/include/bits/fcntl2.h:51: error: call to ‘__open_missing_mode’ declared with attribute error: open with O_CREAT in second argument needs 3 arguments
11:51.43rajmohancan anyone help pls
11:51.51rajmohanyou mean?
11:51.55rajmohanwhat should i do pls
11:51.59rajmohaniam a newbie
11:52.11Pan3Drajmohan: Sir, it's impolite to post code or errors in the channel.
11:52.32Pan3Dinstead, use a service such as http://paste.lisp.org/
11:52.47rajmohanohh sorry...
11:52.57Zeeek~pb
11:52.58infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/
11:53.06Pan3DPeople will not be able to help you if the text is in the channel. Makes it very hard to see things.
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11:54.03rajmohanPan3D: i understand, sorry for that, iam going to use pastebin now
11:55.09Pan3Dno problem, now you know
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11:55.58rajmohanhi, i have got it pastebin
11:56.06rajmohancan anyone check with this error
11:56.09rajmohanand help me out
11:56.12rajmohanhttp://pastebin.com/m17b17c73
11:58.18rajmohanhi Pan3D: can you check that error and let me know what i should do so that asterisk install will be fine?
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12:00.44rajmohanhi, can anyone help me with this error during make install http://pastebin.com/m17b17c73
12:00.46rajmohan?
12:01.51smpsrajmohan, what version of asterisk do you have ?
12:02.38rajmohansmps: iam installing one now, the version is 1.4.10
12:03.06smpsrajmohan, why 1.4.10 ? its a bit old :)
12:03.58rajmohanohh.. i was using a .sh file which had that version
12:06.29rajmohansmps, if i try 1.4.26-rc1
12:06.34rajmohanwill that work?
12:06.45wdoekes(you could add 0600 as a last argument to the open() call on line 1417, but you really should use something newer)
12:06.45smpsrajmohan, i think you will be better with 1.4.25
12:07.20rajmohansmps, let me do that now pls
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12:13.24zambaworking softphone for linux?
12:13.29zambaekiga is -shit-
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12:14.03[TK]D-Fenderzamba: linphone
12:14.10elredpjsua, xlite
12:14.11[TK]D-Fender~softphone
12:14.12infobot[~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga
12:14.20[TK]D-Fender^^^
12:14.27elredtwinkle is nice
12:18.49smpszamba, try wengophone too
12:19.04zambaok
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12:21.07brunnerdoes CNAM work in other countries the same way it does in the US?
12:21.19brunnerin North America, rather
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12:47.25Router222hi
12:47.48hfx-ed[TK]D-Fender: my audio issue the other day appears to have been hardware based. We switched out the MB and memory and the problem went away.  Thanks for your help the other day.
12:47.57Router222why when i set the directory for fax file =/root/fax2send the fax failes
12:48.14Router222and when set i set the directory for fax file =/tmp fax goes normaly ?
12:48.28Router222both chown as root
12:48.56[TK]D-FenderRouter222: And is * running as root?
12:49.10[TK]D-FenderRouter222: And root doesn't care about perms
12:50.47Router222[TK]D-Fender it running as asterisk
12:50.57Router222how to know on which user its running
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12:52.02Router222[TK]D-Fender kindly check it http://pastebin.com/d18deb0a5
12:52.04[TK]D-FenderRouter222: "it is running as asterisk" <- whats this supposed to be if you don't know what user * is running as?
12:52.21Router222i did a ps -aux | grep
12:52.27[TK]D-FenderRouter222: asterisk  3437  0.4  3.3  70072 17216 tty1     Sl   Jun02  19:49 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c <-----
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12:52.53[TK]D-FenderRouter222: Well if you tell it to write to a folder owned by ROOT, you expect this to work?  I wouldn't.  It should be owned by "asterisk"
12:53.12Router222[TK]D-Fender i choud chown asterisk:asterisk for the folder ?
12:53.33[TK]D-FenderRouter222: I think you should NOT be putting it in a folder under /root as well
12:54.14Router222i already did
12:54.34Router222root@pbx:~ $ ls -la fax2send/
12:54.34Router222total 120
12:54.34Router222drwxrwxrwx 2 root root   4096 Jun  5 12:17 .
12:54.34Router222drwxr-x--- 8 root root   4096 Jun  5 12:27 ..
12:54.34Router222-rwxrwxrwx 1 root root 104976 Jun  5 12:17 2.tif
12:54.41[TK]D-Fender...
12:54.55[TK]D-FenderRouter222: DO NOT SET AS OWNED BY ROOT!  What do you not understand!?
12:55.34Router222ok
12:55.42Router222i did that trying to solve the proble m
12:55.53[TK]D-FenderRouter222: and do NOT put ANY of this anywhere under "/root"
12:55.56Router222and didnt suceed
12:56.00[TK]D-Fender^^^^^^^
12:56.27Router222should i add a user asterisk
12:56.55*** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-185039576552c73f)
12:57.00ariel_Morning
12:58.08*** join/#asterisk Slashman (n=Slash@ariane.fimasys.com)
12:58.34[TK]D-Fender[08:52]<[TK]D-Fender>Router222: asterisk  3437 0.4 3.3  70072 17216 tty1  Sl Jun02 19:49 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c <----- Does this not tell you right to your face that you already have one, and that is who * is running as?
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13:03.01[TK]D-Fenderwonders how many times people can miss the "big print". Als how many licks it takes to get to the center of a Tootsie Roll Pop.
13:03.24kaldemar42?
13:04.04jayteethe official answer to the Tootsie Pop question is: "The world may never know"
13:04.58jayteeand the scientific answer is that it varies from person to person. some people have fatter tongues than others and longer licks, also there are variances in the acidity of their saliva.
13:04.59[TK]D-Fenderjaytee: I DEMAND TO KNO!
13:05.02[TK]D-FenderKNOW*
13:05.40jayteey'know, until this morning I had no idea that autoeroticism was one of the Kung Fu disciplines. I'm gonna have to look into that.
13:05.50coppicesome guitarists have longer licks, too
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13:08.05Zeeekcoppice tapping
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13:08.45Zeeektoday we're testing video on the conference: http://tinychat.com/voip if you have a camera. The conf starts at 12 Noon EDT as usual.
13:08.53[TK]D-Fenderjaytee: on that topic, I rocked out big last night. Did "whiskey In the Jar" (Metallica), "The Middle" (Jimmy Eat World), "Just Like A Paradise" (David Lee Roth), and "Dance the Night Away" (Van Halen) on guitar on top of my keyboard work.
13:08.54*** join/#asterisk Router222 (n=CK@212.98.141.199)
13:09.07ZeeekAsk for password on #voip-users-conference
13:09.16jaytee[TK]D-Fender, cool!
13:09.33coppiceshouldn't "whiskey In the Jar" be Thin Lizzy?
13:09.49Router222[TK]D-Fender what should i do to solve the problem  ?Chown -R can do that
13:10.07Router222i have group name asterisk not a user
13:10.18jaytee[TK]D-Fender, have you ever seen the YouTube of KT Tunstall doing Black Horse and a Cherry Tree where laying down one looped track after another adding to the song so she can do her own rhythm, percussion and chorus?
13:10.25[TK]D-Fenderjaytee: Was a real rush to get to play the DLR one as it came up amonst members of another group whose bassist didn't know it, so they subbed him out and I'm like "I know this!".  Totally awesome,,,,
13:10.56[TK]D-Fenderjaytee: Nope, link me!
13:11.06jayteehang on, I"ll have to find it.
13:12.04[TK]D-Fendercoppice: the song itself goes back centuries (old Irish folk), and yes Thin Lizzy did cover it, and Metallica covered their version (minor changes in the lyrics and several verses chopped of more traditional ones
13:12.44[TK]D-Fendercoppice: Frankly the psychedelic tremolo on Thin Lizzy's version kinda nauseates me ;)
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13:14.46jayteehttp://www.youtube.com/watch?v=2c4Olwqasvc&feature=related
13:15.10jayteethat's not the first video but she does it the same way each time live, not prerecorded tracks
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13:17.13coppice[TK]D-Fender: Phil Linnit turned that one into a bit of a durge, which was kind of untypical of his stuff
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13:22.04[TK]D-Fendercoppice: Another group did "the Boys Are Back In Town" which was cool...
13:22.46coppice[TK]D-Fender: well, that's the song most typical of Thin Lizzy
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13:24.00[TK]D-Fenderjaytee: Yeah, seems like she used a loop pedal there...
13:24.39jayteeyep
13:25.09[TK]D-Fendercoppice: Another of so many bands I don't know so well.  I tend to hear things and go "Yeah, I know that song", but could never name the artist
13:25.15jayteebut kept laying things down one loop over another adding to it each time till she had a good rhythm and backup vocals going
13:25.45coppice[TK]D-Fender: were they ever big in the US?
13:26.53[TK]D-Fendercoppice: Thin Lizzy?  Probably a little in their time, but more for single hits more than overall.
13:27.47[TK]D-Fenderjaytee: You know I've heard her on the radio several times, but I'd never seen her before.  I'd hit it in a heartbeat ;)
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13:43.08jaytee[TK]D-Fender, my buddy has music software that can emulate any kind of guitar amp and actually shows an amp on the monitor that looks like the real thing, comes with hundreds of presets too. Like an amp Hendrix used and the presets for All along the Watchtower. It's soo cool.
13:48.23DavidR2008how do you set up channel groups for dialing out with dahdi? I'm looking through the sample system.conf and can't find anything about groups
13:49.04DavidR2008with zaptel you just set a group before specifying the channel
13:50.54[TK]D-FenderDavidR2008: the same
13:50.56leifmadsenDavidR2008: groups are set in chan_dahdi.conf pretty sure
13:50.59[TK]D-Fender^^^^
13:51.04leifmadsengroup=1
13:51.11[TK]D-FenderDavidR2008: You seem to have missed which file replaces which
13:51.29DavidR2008ah, ok. thanks for the clue
13:51.33[TK]D-FenderDavidR2008: system.conf = zaptel.conf, what you need is zapata.conf/chan_dahdi.conf
13:51.37leifmadsengrouping is an asterisk concept, not a driver concept
13:52.15[TK]D-Fenderjaytee: Yup, i run a Boss GT-8 modelling pedal, and a Boss VF-1 half-rack in my rig which does this as well.
13:52.22*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)
13:52.47[TK]D-Fenderjaytee: I tend to use either Marshall 1959 I/II or Peavey 5150's as my base
13:55.01killfillhey guys,  i loaded wctdm with debug=1, and in /var/log/messages, i see from when a call gets in until it hangs up:  "RING on 1/1, NO RING on 1/1, etc, etc"  withouth any 'polarity' string when hangs up occur.
13:55.22killfillthis means i can be 100% sure that my telco doesnt provide me "disconnect supervision", right?
13:55.25jaytee[TK]D-Fender, yeah, but his is ALL software except for his guitars and keyboards. He has real amps but he doesn't need to use them unless he's playing live. when he's jamming he uses his computer and lays things down a track at time and he can mute any of the track or record several lead tracks and switch between them during playback to choose which one he likes best. it's incredible what he can do with his stuff.
13:56.19jayteehe's probably spent at least 10 grand if not more on the basic software and all the add-ons
13:56.40[TK]D-Fenderjaytee: eek.  I'm "frugal" with my setup...
13:56.57jayteehe's got money
13:56.57coppicejaytee: the latency of the PC is a pain for a performer, but other that it an excellent DSP :-)
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13:57.42jayteecoppice, he has an interface box that handles most of the DSP processing and it's made for the software
13:57.43[TK]D-Fendercoppice: yup, post-production it is...
13:58.15coppice[TK]D-Fender: not too easy on stage :-)
13:58.22jayteenope
13:58.27[TK]D-Fendernada
13:58.33jayteebut for mixing stuff it's great
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14:00.16KattyGOOD MORNING YOU BEAUTIFUL PEOPLE
14:00.23Zeeek<PROTECTED>
14:00.27Zeeekhere's your chance
14:00.32Kattyhugs Zeeek
14:00.48[TK]D-Fenderjaytee: Next item I'm looking at : http://www.long-mcquade.com/?page=products&ProductsID=2563
14:00.57ZeeekKatty http://vuc.me/video you can paek at us
14:01.16Kattyorly
14:01.30jayteemorning Katty *hugs*
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14:01.38Kattyhugs on jaytee
14:02.28KattyZeeek: it asks me for a passwordededed.
14:02.30jaytee[TK]D-Fender, that's cool!
14:02.46Zeeekyes the pw is the default SIP port
14:03.24Kattyohiseethenk
14:03.33[TK]D-Fenderjaytee: Analog modelling synth. not a ROMpler....
14:03.46*** join/#asterisk davevg-btwtech (n=davevg__@67.76.177.147)
14:03.49[TK]D-Fenderjaytee: Looking to get a bit of the dynamic vintage feel
14:03.58[TK]D-Fenderjaytee: And doubles as a good MIDI controller
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14:10.03c4rganyone using asterisk+iaxmodem+hylafax for mass faxing?
14:11.03coppicehow "mass" are we talking about?
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14:14.19[TK]D-Fendercoppice: About 2kg ;)
14:14.32killfillim not sure about this.. ive loaded wctdm with debug=1, and i dont see "polarity switch" or snything like that in /var/log/messages, just the rings.
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14:14.56killfilldoes this mean my telco is not sending the polarity switch's?  or is it that my zptel does not recognize them?
14:15.09[TK]D-Fenderkillfill: Did you confirm with your telco that they support it?
14:15.34coppice[TK]D-Fender: how many calls do you get for 2kg?
14:15.38killfill[TK]D-Fender: does dont know what im talking about, and wont helpwith anything.
14:15.55killfilldoes/they. sorry
14:16.53killfillso i trying to figured this out myself...
14:17.14[TK]D-Fendercoppice: Depends waht it is I have 2kg of, who the buyer is, current market value and the result of haggling ;)
14:18.48killfill[TK]D-Fender: can i be sure they does not use polatiry switch with this info?
14:19.22[TK]D-Fenderkillfill: I'm not the best to advise....
14:19.36killfillok..
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14:22.21festr_Hi. I need some ideas how to achive this: Dialplan -> Answer,Play,Dial. this will save CDR always with ANSWER flag. But I want to reset billsec and duration asap Dial starts. Is it possible?
14:23.28[TK]D-Fenderfestr_: resetCDR()
14:24.11rajmohanhi can anyone tell how to i configure ekiga for asterisk
14:25.43[TK]D-Fenderrajmohan: 3 fields for user, pass, and server IP.
14:25.56kaldemarDON'T use this, but i'm too tempted to say "aptitude remove ekiga".
14:26.18[TK]D-Fenderrajmohan: Plenty of guides for this you can google in about 10 seconds flat if you can't hunt these 3 simple fields yourself
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14:28.22iksik<iksik> which option is the best solution for providing fax2mail / mail2fax services?
14:28.22iksik* Disconnected (No route to host).
14:28.29iksikthere was any answer?
14:28.59leifmadseniksik: nope
14:29.10iksikok :P
14:29.15*** join/#asterisk Attilas (n=lord_att@modemcable048.42-83-70.mc.videotron.ca)
14:29.26leifmadseniksik: try hylafax for fax2mail
14:29.33*** part/#asterisk Attilas (n=lord_att@modemcable048.42-83-70.mc.videotron.ca)
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14:34.02ZeeekTalking to Sipgate today about their new invasion of the US DID market by the way: http://vuc.me in about 90 minutes
14:36.14xhelioxGood morning. Using a TE122 and when the call is bridged with a Polycom SIP phone, the remote end hears a random popping/clicking noise. It doesn't happen if we call in over a SIP trunk or if we simply play a sound file.. I've tried disabling echo cancellation just for fun, that didn't seem to solve anything. Using Ulaw on the phones.. Any advice?
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14:38.03*** part/#asterisk fred-tmft (n=fred-tea@c-69-244-180-112.hsd1.mi.comcast.net)
14:38.04[TK]D-Fenderxheliox: Its the card.  Check for shared IRQ, system load, try different EC, etc
14:38.04*** join/#asterisk acxty (n=acxty@201.220.136.117)
14:38.05acxtyHi guys
14:39.05iksikoh, ekhm, is it possible to send SMS ? :|
14:39.27xhelioxSystem load is fine.. EC is off (at the moment).. nothing on the same IRQ (there was, but not now).. doesn't that usually also effect call stability too?
14:40.16[TK]D-Fenderxheliox: Often
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14:41.14acxtywell guys I am trying to make a record() to stop, is there a way you can press 9 and record() stop for example?
14:41.43xheliox[TK]D-Fender: Also dahdi_test is showing very good, 99.998% or greater..
14:42.18[TK]D-Fenderacxty: "core show application record"
14:42.39acxtythanks TK
14:52.04*** join/#asterisk watchy (n=watchy@76.196.98.139)
14:52.10watchyis realtime mysql or odbc better?
14:53.07*** join/#asterisk Whitor (n=Whitor@64.128.237.124)
14:57.09lanceywatchy: odbc is just an abstraction layer
14:57.14leifmadsenwatchy: odbc is more stable
14:57.27leifmadsen(in terms of asterisk implementation) and more supported
14:57.27lanceyuse native support for your db if available (mysql, postgre)
14:57.33leifmadsenlancey: incorrect
14:57.34*** join/#asterisk Attilas (n=lord_att@modemcable048.42-83-70.mc.videotron.ca)
14:57.35leifmadsenuse ODBC.
14:57.57lanceyleifmadsen, could be. i've never had any problems using mysql directly
14:58.18lanceybut that's nothing more than a personal opinion ;)
14:58.29*** part/#asterisk Attilas (n=lord_att@modemcable048.42-83-70.mc.videotron.ca)
14:58.37jayteelemme see here, one person says ODBC and another says native.....think I'm gonna go with the guy who wrote the book :-)
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14:59.00watchywhich wrote the book?
14:59.05leifmadsenlancey: ya, ODBC is much better supported than the -addons mysql stuff. You use what works for you, but I've seen crashes with things like the MYSQL() application
14:59.14leifmadsenpoints at lancey
14:59.15shido6I used both and u neeed to make sure u use a stable odbc version
14:59.23leifmadsenshido6: amen
14:59.36lanceycould be.. i use it for cdrs and realtime peers only
14:59.37leifmadsentest, test, test on whatever it is you're going to use
14:59.43leifmadsenthis is just IRC -- all opinions are wrong
14:59.55shido6then tear it down and build it up on another server and test again
15:00.11leifmadsenyep
15:00.19watchyim just using it now for rael time voicemail confs
15:00.19jayteewhile I haven't had any issues with mysql fortunately, i'd like to migrate to using ODBC to MS SQL Server.
15:00.23leifmadsenand then get scared, and just avoid the whole thing
15:00.49shido6then write a script to fux0r it on purpose maybe 10 - 20  or 30 calls per second and see if it craps out a blue baby duck with the linux and windows logos on each bum cheek
15:01.01jayteehahaha
15:01.11[TK]D-Fenderleifmadsen: Everyone is entitled to my own opinion ;)
15:01.16*** join/#asterisk smps (n=smps@193.170.53.51)
15:01.38jayteeor as my boss always says, "When I want your opinion I'll give it to you!"
15:01.45[TK]D-Fenderleifmadsen: And I may not always be right, but I am never wrong!
15:02.02leifmadsenjaytee: that's what my boss says too
15:02.07leifmadsenis his own boss
15:02.49jayteebeing your own boss has it's pros and cons, the worst con being you can't point to someone else and say, "He's an assmunch!"
15:03.05DavidR2008since leifmadsen didn't duck and cover, either he really is his own boss, or his boss isn't in this channel
15:03.19ZeeekKatty: is soooo charming
15:03.36Zeeekor is it just that we almost never have female voices on the channel?
15:04.07watchyi like girls
15:04.09Zeeekanyway, I digress. See you in an hour in full color and wideband audio #voip-users-conference + http://vuc.me
15:04.11leifmadsenDavidR2008: I really am my own boss :)
15:04.36DavidR2008you go man, I was just picking at you :-)
15:04.44jayteeit's harder being your own boss than working for someone else
15:04.48DavidR2008but I do have one question ... are you married ;-)
15:04.49Zeeektotally
15:05.00ZeeekI'm married to my boss/employee
15:05.02jayteebut at least you don't get stuck implementing someone else's brain dead decisions
15:05.23Zeeekno, we get the customer's braind dead decisions and HAVE TO PRETEND to agree
15:05.34Zeeekyou can't do the ugly baby syndrome
15:05.37leifmadsenjaytee: totally
15:05.40Zeeekyou lose the customer
15:05.41DavidR2008yea, but it worse when half way through implementation you realize it was YOUR brain dead decision ;-)
15:05.44leifmadsenjaytee: I do a lot more work that I don't get paid for :)
15:05.59Zeeekyou're gonna have to serve somebody
15:06.13KavanSZeeek, are you saying it might as well be Satan?
15:06.39Zeeekno Satan was my neighbor but he finally moved away. No kidding
15:06.41jayteeleifmadsen, when was the last time you could just take a week off for vacation on short notice? :-)
15:06.57Zeeekvacation doesn't exist if you're self-employed
15:07.06leifmadsenjaytee: NULL
15:07.17leifmadseng/f is making me take a week vacation in July though
15:07.19[TK]D-FenderKavanS: Great way to get rid of Jehovah's Witnesses : Tell them to holda minute at the door says "Just s sec guys, I have to time the sacrifice perfectly or the Dark Lord will be most displeased...."
15:07.21leifmadsenwe'll see how that goes....
15:07.35leifmadsenlast year I took a month vacation though
15:07.41leifmadsendrove across north america
15:07.44KavanS[TK]D-Fender, lol....I might have to do this....it has been awhile since Jehovah's Witnesses have stopped by
15:07.46leifmadsenfirst vacation ever :)
15:07.55KavanSleifmadsen, nice, where did ya go?
15:08.06jayteeleifmadsen, and that's one of the other downsides to working for yourself. too many people think it's glamous but it's actually much harder.
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15:08.25flujanhello guys, I am having a problem with CDR. It's is not recording on the cdr table the FAILED and BUSY values. Here goes a example: http://pastie.org/501640
15:08.39flujanI enabled the unaswered = yes at cdr.conf without success.
15:08.54leifmadsenKavanS: Toronto --> Standard, Alberta --> Vancouver, BC --> San Francisco, CA --> Las Vegas --> AstriCon 2008 (Phoenix) --> Toronto
15:08.57jayteeI've driven across the US from one coast to the other twice. It really makes me appreciate what a beautiful country I live in.
15:09.29KavanSleifmadsen, ahh nice, you hit up some good spots :)  I like visiting Vegas, but I couldn't live there
15:09.39leifmadsenKavanS: I don't even like visiting it
15:09.43leifmadsenwasn't very impressed
15:09.46KavanSlol y eah
15:10.02jayteeVegas, meh!
15:10.05KavanSI can see how brothels and gambling made the town...
15:10.11KavanSconsidering it's a desert! :)
15:10.13jayteejust a hole in the desert you can lose money in
15:10.55ectospasmyeah, but that's supposed to be FUN!
15:11.32jayteeunless you like being broke, having crabs or VD, have a fetish for lots of neon lights and getting scorched by the sun I just don't see what attracts people to Vegas
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15:12.03shido6its only fun when you've been clobbered and had your life threatened in a backroom for hacking the shluts
15:12.04ectospasmyou're going to the wrong hookers if you get crabs or an STD
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15:12.15jayteeI liked San Francisco and Seattle and Salt Lake City is really pretty but they roll up the sidewalks at 8pm
15:12.40shido6I dont wake up till 11 that would suck
15:13.16ectospasmtoday's an exception, I don't usually wake up until two pm
15:13.31hackt0rhow can I see which type my ISDN line is using: p2p or p2mp?
15:13.42Kattysleepy
15:13.53jaytee"Revive with Vivarin!"
15:13.55ectospasmhackt0r: your telco should be able to tell you if it's ptp or ptmp
15:14.20Katty^_-
15:14.39ectospasmor, since there's only two choices, you can try each and see which one works
15:16.35hackt0rectospasm: thnx
15:16.46ectospasmhackt0r: np
15:18.48ZeeekKatty have a Kit-Kat!
15:19.43*** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-206.phlapa.fios.verizon.net)
15:22.36watchykatty is super awesome
15:23.11watchywhen i grow up i hope to have a girlfriend just like her!
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15:25.01eppigyKatty: me2
15:25.07eppigytoo much fun last nigth
15:25.09eppigynight
15:25.11eppigyD:
15:25.35watchyi was up t ill 6am fighting with a chick then came to work at 8
15:25.37watchyawesome night
15:26.31ectospasmwatchy: sounds like it
15:26.36[TK]D-Fenderwatchy: When did the fight start, and did ESPN cover it?
15:26.43lanceylol
15:26.50watchytk: no but i think youtube did
15:28.18*** join/#asterisk bijit (n=benji@190.241.15.48)
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15:30.26[TK]D-Fenderwatchy: So how long for the argument(s) themselves?
15:31.09watchyhours
15:31.47*** join/#asterisk moy (n=moy@74.12.123.90)
15:33.47ectospasmand no makeup sex?  That's lousy...
15:34.33*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
15:34.42*** join/#asterisk nny_1 (n=Scott@64.203.237.47)
15:34.46watchynah, she thinks i'm crazy
15:35.00ectospasmis she right?
15:35.08watchyof course she is. i'm on irc aint i
15:35.26nny_1eh. Any reason why removing all traces of a registration from sip.conf and I am still getting chan_sip.c:15889 handle_request_register: failed spam ? Very annoying considering sip show peers on both sides has no sign of it now.
15:35.30ectospasmthat doesn't ipso facto mean you're crazy
15:35.58watchyplus shes mad cause id rather work than be with her or banging her
15:36.01ectospasmnny_1: is the phone still trying to register with the old registration credentials?
15:36.16nny_1not a phone, asterisk - asterisk link i set up for testing
15:36.33nny_1and i took out all traces of the registration from both sip.conf files, yet spam spam spam
15:36.37*** join/#asterisk marv[work] (n=timr@router.asteriasgi.com)
15:36.37*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
15:36.43nny_1(and reloaded)
15:36.53ectospasmwatchy: might as well ditch the relationship now, unless you really want it to work out.  You'll need to budget time with her better to make her happy (if you care about that)
15:36.58nny_1well sip reload, not restart, as the system is being used
15:37.54ectospasmnny_1: you may want to wait until a maintenance window and restart asterisk
15:38.11ectospasmIt seems weird that a reload isn't doing it.  You did make sure to remove the register => lines, right?
15:38.28nny_1ectospasm: yeah thats what i figured. Heh yay! gonna have to move the log files to, as right now there is a bunch of the fail spam
15:38.34watchyectospasm: i think relationships are a waste of time really. but i'm willing to deal with the bs if she is. i could careless
15:38.48nny_1ectospasm: oh yeah, i even grepped the entire conf folder for the auth name etc
15:39.34nny_1[Jun  5 11:39:17] NOTICE[4134]: chan_sip.c:15889 handle_request_register: Registration from '"3030" <sip:3030@XXXXXXXXXXX>' failed for 'XXXXXXXXXXXX' - No matching peer found
15:39.36ectospasmwatchy: doesn't sound like you'd make healthy relationship material.  But what do I know, I haven't had a real girlfriend in over three years )-;
15:39.45nny_1every second
15:39.49Zeeekfor more on VUC friday #voip-users-conference  http://vuc.me
15:40.12ectospasmyeah, something is trying really hard to register
15:41.40nny_1ectospasm: yeah however if I go into the system listed at the IP and (i even restarted it) shut it down, it keeps trying, or at least the spam is there
15:42.17ectospasmyou checked both users.conf and sip.conf?
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15:42.29ectospasm(if users.conf is applicable)
15:42.30nny_1ectospasm: aye don't use users.conf
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15:43.04nny_1not hurting anything other than the log file, just curious as to why it would get stuck in such a state. sip show peers on both systems shows no entry for 3030 either
15:43.31nny_1oh ha
15:43.35nny_1i figured it out
15:43.53nny_1had someone set up an xlite client for testing before they realized the box was NAT
15:44.03nny_1must still be running in the sys tray on the dev station
15:44.04nny_1-_-
15:44.24nny_1note to self, HATE systray applications that don't really shut off
15:44.26ectospasmheheh
15:44.50ectospasmI find Xlite to be repugnant myself.
15:44.53nny_1but at least that solves the great mystery
15:44.56nny_1as do i sadly
15:45.29nny_1haven't found a good open source client that isn't 1/2 gimped without pay either
15:48.47ariel_I have been happy with sjphone
15:49.21Kattywatchy: that would not be advisable, unless you like a moody zookeeper
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15:51.25Kattyeppigy: who is hungry :<
15:51.34Kattyeppigy: what's fur lunch :>
15:51.59rue_mohrhmm is there a bot here with the help info on the functions? i want to know if there is a paramiter for dial to say which type of distinctive ring should be used to a sip phone
15:52.04ariel_Pizza
15:52.51Kattyinfobot: seen twisted?
15:52.52infobottwisted <n=twisted@router.asteriasgi.com> was last seen on IRC in channel #asterisk-dev, 80d 23h 25m 5s ago, saying: 'jsmith: but they do, that's part of the issue ;P'.
15:53.00*** part/#asterisk nny_1 (n=Scott@64.203.237.47)
15:53.01Kattyholy crap, 80 days
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16:01.38Kattyuploads new video of BB to youtube.
16:01.48*** join/#asterisk nullable_type (n=nullable@76.77.70.226)
16:01.57rue_mohrputs on body armour in prep of asking tk something
16:02.01*** join/#asterisk propellerhead (n=yogurt2u@host74.190-31-202.telecom.net.ar)
16:02.06Qwellpresses "record"
16:02.15Kattygives rue_mohr football helmet
16:02.30Kattyshares popcorn with Qwell
16:02.36nullable_typeTo detect a user entering 1. Is WaitExten() or Read() better choice.  For some reasons WaitExten() stopped working
16:02.42rue_mohrdoublechecks google...
16:02.53Kattynullable_type: during an auto attendant?
16:03.07nullable_typeyes
16:03.13Kattynullable_type: WaitExten(seconds)
16:03.16ariel_I use Read
16:03.55nullable_typeI was using WaitExten with empty parameters, was working fine, until today
16:04.12rue_mohrso, if I'm gonna set up paging, I need to configure the phones for autoanswer...
16:04.12Kattywaitexten doesn't know how long to wait for data, unless you tell it.
16:04.23Kattyrue_mohr: yes.
16:04.33Kattyrue_mohr: possibly ring answer
16:05.26nullable_typealrite i will try passing the parameter
16:05.33rue_mohrdoes it operate on a line context or a cid context?
16:05.59Kattyrue_mohr: would you like to see a working example?
16:05.59Kattyrue_mohr: for polycoms
16:06.06QwellKatty: guess what phones he doesn't use
16:06.06rue_mohrI have 4 polycom 601 and 2 aatra-somethings
16:06.14Kattyoh i see.
16:06.24rue_mohr33i
16:06.57rue_mohrso yes, please
16:07.07*** join/#asterisk theron (n=theron@216.51.246.211)
16:07.36Kattyhttp://pastebin.ca/1448841
16:07.44nullable_typeKatty >> I tried WaitExten with different seconds (10) and (15). It still was unable to get it
16:08.13Qwellnullable_type: waitexten is for...extensions.  are you wanting to read a variable, or what?
16:08.13Kattynullable_type: then you probably should look at your auto attendant more closely
16:08.21*** join/#asterisk v0lksman (n=shayne@ottawa-hs-64-26-169-151.s-ip.magma.ca)
16:08.35rue_mohrso the page function takes care of hte one way audio then I assume
16:08.53Kattyrue_mohr: show application page
16:08.57Kattyrue_mohr: there are a few options
16:09.13djMaxcan someone point me towards implementing a "whisper" before a call is delivered?  Is that just with Dial() or something special?
16:09.21*** join/#asterisk theron (n=theron@216.51.246.211)
16:09.25nullable_typeQwell >> Yes, i read somewhere before to use WaitExten just for my purpose of detecting the user entered "1"
16:09.44nullable_typeQwell >> And it worked fine, until today the provider changed something
16:09.45KattydjMax: that would be the 'A' option
16:10.28Kattynullable_type: pastebin
16:10.29djMaxperfect.  And it rings the other side while it's playing that announcement?
16:10.51KattydjMax: we use it here to annouce which company the call is coming in for.
16:10.57KattydjMax: so the receiptionist knows how to answer th ephone
16:11.03KattydjMax: the calling party does not hear it, just ringing
16:11.17djMaxwow, this feature is going to take all of 5 seconds to build. :)
16:11.27KattydjMax: less. here's a working example
16:11.32watchykaty: when the recept picks up it says something?
16:11.38KattydjMax: exten => s,n,Dial(SIP/125,45,A(companies/SECURITAS))
16:11.44djMax(if anybody remembers me from a month back - amazingly XO is still not able to figure out how to give us 2BCT!)
16:11.58Kattywatchy: she hits answer, it says 'securitas' then she answers the phone
16:12.09Kattywatchy: or talks, i guess. not really answer button again
16:12.15djMaxmy outbound is external, I imagine that doesn't change things
16:12.15rue_mohrKatty, OOooo! I use call display data!
16:12.16watchyah sweet
16:12.28Kattyrue_mohr: we use annoying smbclient popups too
16:12.31watchyi display the company in the CID katty
16:12.37rue_mohrasks the receptionist if this would be a better idea
16:12.38Kattywatchy: no...
16:12.41Kattywatchy: not that company
16:12.47Kattywatchy: we host about 4 companies in this office
16:12.51watchysame here
16:13.01Kattywatchy: she still needs the callerid (=
16:13.06watchywe have a company called GoWireless i just put GW in front of the CID
16:13.08Kattywatchy: of the original caller. so i don't overwrite it
16:13.17watchyi just apend to the front
16:13.17Kattyah
16:13.18djMaxpops are nice too, how do people normally do those - with a persistent manager connection or some messaging thing?
16:13.19watchy2 leters
16:13.21Kattythat would work
16:13.25watchyyup
16:13.30KattydjMax: well we use smbclient
16:13.41KattydjMax: which uses the windows messenger service to create a popup
16:13.43djMaxyou don't get to customize the display of that through right?
16:13.53djMaxlike if we wanted to pull up a CRM page or something
16:13.59Kattynope, just a plain ole windows popup
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16:14.08watchymy friends writing a delphi app to do that
16:14.20rene-hello does anybody know about this error (rc_avpair_new: unknown attribute 1490026597)
16:14.23Kattyexten => s,n,System(/bin/echo -e "'-------------------- \\r Incoming Call From: ${CALLERID(all)} \\r -------------------- v300 \\r Thank you for calling R.W. Duke Capital Management. How may I direct your call'"|/usr/bin/smbclient -M SO9319)
16:14.27djMaxGuess it could send UDP instead, or just hang on to a manager conn
16:14.30watchyit connects using the manager and will display callerid on the pc
16:14.36Kattythat's all it is
16:14.42watchyi was thinking about having it use googletalk to do it
16:14.42Kattyquick, dirty, but effective.
16:14.50*** join/#asterisk porche (n=kursad@88.233.60.211)
16:14.51Kattywe did jabber for awhile
16:14.55djMaxthere's a summary of asterisk. :)
16:14.56Kattyrather than sending it to smbclient
16:15.05watchywhy did you switch to smb
16:15.10Kattyit was too slow
16:15.19watchyoh, yea i didn't think about that
16:15.20porchehi all, i have a problem with GET DATA/AGI, escape digit #, causes the call to hang up immediately
16:15.38porchewhere must I check?
16:15.50Kattywatchy: isympony supports jabber as it's foundation for chat
16:15.58Kattywatchy: it was really nice, if it had only been a bit quicker.
16:16.01watchyi think isymony sucks doesnt it
16:16.06Kattyi lov eit
16:16.22Kattywe were using FOP
16:16.24Kattyit was awful.
16:16.33watchyhaha fop is so ghetto
16:16.38Kattyso is asterisk tho
16:16.39Kattyghetto
16:16.43Katty;p
16:16.49watchyyour ghetto :(
16:16.58Kattyyour face is ghetto
16:16.59ectospasmit's very interesting if you google the word "ghetto"
16:17.02watchyyour booty is ghetto
16:17.10Kattyyour mom's booty is ghetto
16:17.15watchyyea :(
16:17.22Kattyhehe
16:17.56porcheany clue for me?
16:18.29nullable_typeKatty >> WaitExten issue: http://pastebin.com/d412cc0b6 I numbered so you can refer to the line number. Thanks
16:18.48nullable_typelol didn't realize pastebin numbers too
16:19.10Kattyalright guys, i'll cya later. it's lunch time!
16:19.13Kattyeppigy: let's gooooo :>
16:19.16watchyanyone know what portage mysql odbc is under?
16:19.23nullable_typeKatty >> Can you look into that pls
16:19.31Kattynullable_type: nope, i'm gonna have lunch.
16:19.36nullable_typelol ok
16:19.40Kattynullable_type: good luck with your issue (= i hope you get it resolved.
16:19.42watchyits myodbc what an idiot i am
16:19.47ariel_port 3306
16:19.53nullable_typeQwell >> Can you help
16:20.02watchygentoo portage, not port
16:20.41watchy13 packages just for myodbc wtf
16:20.56*** join/#asterisk lanning (n=lanning@173.8.187.197)
16:21.04ariel_watchy: sorry, I am a apt-get person....
16:21.13watchythats debian right?
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16:21.20ariel_yes
16:21.25watchyi dont use deb or ubuntu
16:21.31watchynever have never will
16:21.35*** join/#asterisk Nilzao (n=nils@189-68-191-170.dsl.telesp.net.br)
16:21.39watchyi like gentoo because its so base on the install
16:21.44ariel_I also do yum for some systems... CentOS in those cases.
16:21.48watchyless overhead
16:22.06ariel_If it's a secure basic system I go the FBSD route
16:22.15rue_mohrok dial() dosn't seem to have a distinctive ring setting, I want to have it so that the sip phones use a different ring tone when called from another sip phone, and the aastras dont seem to support that. anyone know what I do?
16:22.41watchyariel_: me too. i love fbsd
16:22.46watchybut for * its not really a choice
16:23.02watchymy entire isp is based off freebsd.
16:24.02ariel_rue_mohr: you should look into the sip info header section
16:24.15ariel_as it's how you can send to most sip phones different rings
16:24.41rue_mohrah
16:25.20rue_mohrexten => 1,1,SIPAddHeader(Alert-Info: Auto Answer) <- along that line?
16:25.35Nilzaohi all, i'm trying to hangup with an extension when runing Dial()
16:25.36ariel_watchy: at work I use debian, due to management, at home I use CentOS due to it's easy and supported by almost all, For F/W or secure system it's FBSD.  There are many choices and it's just what I have gotten used too. They are all good really.
16:25.58Nilzaohungup with some number, like *0 or *99
16:26.56rue_mohrhow do I get a list of things I can add to headers?
16:27.05*** join/#asterisk nunito (n=particul@15.Red-83-41-179.dynamicIP.rima-tde.net)
16:27.26ariel_rue_mohr: google and voip-info
16:27.34ariel_it's lunch time and I have to run. BBL
16:27.35Nilzaorue_mohr: http://www.voip-info.org/wiki/index.php?page=Asterisk+-+documentation+of+application+commands#CallerpresentationIDNameetc
16:27.59nunitoHi I need help for install voices in spanish in my asterisknow 1.5
16:28.06nunitoCan yuo help me?
16:29.21nunitoI have researched the past two days on google but I have not found anything to help me
16:31.02Nilzaoguys, i got it
16:31.11Nilzaoin the features.conf the *0 disconect
16:31.32Nilzaothe channel inspired me lol
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16:33.55*** join/#asterisk ManxPower (n=manxpowe@98.sub-70-214-99.myvzw.com)
16:35.10nunitohello i need tanslate the asterisknow 1.5 to spanish voices Anybody can help me
16:35.23Nilzaoyou know where the .wav or ogg voices are?
16:35.46Nilzaonunito: the .gsm files
16:35.53Nilzaonunito: /var/lib/asterisk/sounds
16:36.01Nilzaonunito: just replace what you whant
16:36.08Nilzaonunito: *whant
16:36.14Nilzaonunito: *want
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16:36.33*** join/#asterisk war9407 (i=war@liquidswords.org)
16:36.33Nilzaonunito: got it?
16:37.14nunitook And ok there are some voices in the repository to download Spanish
16:37.28nunitosorry for my english
16:37.40nunitois very badly
16:37.58Nilzaonunito: so you want the .gsm voices in spanish you don have it
16:38.46Nilzaonunito: http://www.voip-info.org/wiki/view/Asterisk+sound+files+international
16:39.32nunitoyes
16:40.22*** part/#asterisk porche (n=kursad@88.233.60.211)
16:41.21nunitook from the centos conslola as download the voices, and as the copy in the folder
16:42.17QwellHe's already installed Spanish prompts.  He wants to setup his extensions to be in Spanish.  (He's using freepbx..)
16:45.08*** part/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek)
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16:51.19nunito@Qwell i don't load any voice in espanish in de folder of sounds
16:53.57nunito
16:53.57nunitocan I install the voices through yum install asterisk-core-sounds-es-ulaw
16:54.21Nilzaothing... somone call to my asterisk from telco, the dahdi answer, and play the hold music, nice
16:54.40Nilzaothen, the caller just hung up...
16:54.58Nilzaohere my telco send a busy tone to say "the caller hung up"
16:55.20Nilzaoand keep making the busy tone "500,500" too, too, too , too
16:55.42Nilzaoto fix that, i put a timeout
16:56.04Nilzaothe busydetection not working for incoming calls
16:56.13Nilzaoso have to wait the timeout
16:56.18Nilzaoany tip?
16:57.00Nilzaoi tryed the hanguppolarityswitch but nothing
16:57.12Nilzaohanguponpolarityswitch=yes
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17:02.43*** part/#asterisk willianmazzardo (n=willianm@201-14-254-127.smace701.dsl.brasiltelecom.net.br)
17:05.20[TK]D-FenderNilzao: You should ask your telco to provide :
17:05.22[TK]D-Fender~cds
17:05.23infobot[~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up.  This is typically done either by a momentary battery cut, or by a polarity reversal on the line.
17:06.08ManxPowerasterisk detects the voltage cut method automatically
17:06.12[TK]D-Fender[12:25]<rue_mohr>exten => 1,1,SIPAddHeader(Alert-Info: Auto Answer) <- along that line? <- yes if that is the code you configured in your provisioning
17:06.20Nilzao[TK]D-Fender: whell... humans can detect when the other human diconected the other line
17:06.39Nilzao[TK]D-Fender: when the caller hang up, we just listen a busy signal
17:06.52ManxPowerNilzao: Asterisk is not a human and not an AI.  Feel free to submit patches to add the feature you desire
17:06.52[TK]D-Fender[12:26]<rue_mohr>how do I get a list of things I can add to headers? <- You can invent whatever you want.  What kinds a particular device will care about depends on the deivce.  AA, etc are all model specific
17:07.21Nilzao[TK]D-Fender: but asterisk can detect busy signal when calling to a number
17:07.30[TK]D-FenderNilzao: "callprogress=yes" <-Option, however its nearly synonymous with "disconnectmycallswhenleastconvenient=yes"
17:07.54Nilzao[TK]D-Fender: tryed that too
17:08.10[TK]D-Fenderjust finished an AYCE sushi lunch and ironically feels like a beached whale
17:08.15*** join/#asterisk timeshell (n=timeshel@76.70.209.246)
17:08.22Nilzaoi'm using the *0 to disconnect calls
17:08.25*** part/#asterisk lanning (n=lanning@173.8.187.197)
17:08.29[TK]D-FenderNilzao: You needs to have your indications.conf properly configured for your zone, and either way, YMMV
17:08.31Nilzao[TK]D-Fender: the *0 can hangup
17:08.43[TK]D-FenderNilzao: Nothing stops you from hanging up...
17:13.59joakoAnyone have any experiance with XO "Integrated" voice/data T1?
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17:14.29*** mode/#asterisk [+o leifmadsen] by ChanServ
17:15.47Nilzao[TK]D-Fender: the only way to detect with * if the calling party has hung up is with CDS?
17:16.25[TK]D-FenderNilzao: No, but it is the proper way
17:17.03Nilzao[TK]D-Fender: thing that happen... the busydetection is on, and count 6
17:17.11Nilzaoif i'm talking with the caller
17:17.18Nilzao[TK]D-Fender: then the caller hang up
17:17.27Nilzao[TK]D-Fender: the busydetection work, and hangup the call
17:17.47[TK]D-FenderNilzao: DISCONNECT tone is not BUSY!
17:17.55[TK]D-FenderNilzao: they are not the same.
17:17.58Nilzao[TK]D-Fender: here in my country is
17:18.06djMaxit just sounds the same
17:18.13[TK]D-FenderNilzao: Make sure your indications.conf reflects this.
17:18.26[TK]D-FenderdjMax: Close in Northe America, but not the same
17:18.36Nilzao[TK]D-Fender: its working when calling with a sip channel!
17:18.45Nilzao[TK]D-Fender: just don work when the caller listening the MOH
17:18.51Nilzao[TK]D-Fender: or ringing
17:18.57djMaxyeah, figured that's what you meant, though is it really the same in other places? Surprising.
17:19.39Nilzao[TK]D-Fender: there is no cds definition on indications.conf
17:19.53[TK]D-FenderNilzao: Makes no sense... if they're listening to MoH, then how can it be hung up?  What does this have to do with SIP?
17:20.22Nilzao[TK]D-Fender: let me try to say what happens
17:21.00Nilzao[TK]D-Fender: the caller call to my line, and the dahdi send to asterisk, and then the Asterisk DIAL(sipxxxx,,mtT)
17:21.07Nilzao[TK]D-Fender: with music on hold
17:21.45Nilzao[TK]D-Fender: the sip ring forever
17:21.54Nilzao[TK]D-Fender: if i answer the sip channel, i talk with the caller
17:22.09Nilzao[TK]D-Fender: if the caller hangup, the telco send the "like busy" signal
17:22.11nunito
17:22.12nunitoI have installed the voices in Spanish and now I have the folder asterisknow sound of a folder that is called, and guess who has voices in Castilian, but an extension framework cuanfo that there is still leaving me a message in English that I have to configure for change language
17:22.30Nilzao[TK]D-Fender: then asterisk detect the busy signal and close the dahdi channel, good
17:22.48Nilzao[TK]D-Fender: BUT, if the sip ring forever, and the caller just hang up
17:22.57nunitoI have installed the voices in Spanish and now I have sound in the folder of asterisknow a folder called "es", and guess who has voices in Castilian, but an extension framework cuanfo that there is still leaving me a message in English that I configured to change the language
17:22.57nunitobuscar
17:23.11Nilzao[TK]D-Fender: the dahdi keep sending the MOH to the busy channel, like someone there
17:23.26ManxPowerNilzao: Then Asterisk is not detecting the busy
17:23.42Nilzao[TK]D-Fender: if i pick up the sip channel, i listen the busy signal, then it count to 6 and works
17:23.53Nilzao[TK]D-Fender: its not working when the MOH playing
17:23.56ariel_Nilzao: your not adding a time limite at the either.  Dial(SipXXX,,mtT,20)
17:24.07Nilzaoarial_: i did it to fix the problem
17:24.08djMaxis there a way to drop some sort of externally generated identifier into the CDR log? Or should I drive it the other way around keying off the * call id?
17:24.26Nilzaoarial_: did DIal(SIPXXX,20,tTm(mybigmusic))
17:24.36Nilzaoarial_: then 20 sec it drop the line
17:24.55ariel_you should have a rule that hangup for you then
17:25.02ariel_n,hangup
17:25.15Nilzaoarial_: have it too, its working
17:25.17ariel_or send it to VM or what ever else you would like to do
17:25.24Nilzaoarial_: i just want to hangup before hit the 20 seconds
17:25.25*** join/#asterisk simNIX (n=simNIX@156-60.bbned.dsl.internl.net)
17:26.08Nilzaoarial_: * just not detecting busy signal when MOH playing
17:26.42Nilzaoarial_: detect only if a sip channel is talking
17:27.01Nilzaoarial_: the 500,500 busy pattern...
17:31.04*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/zeeek)
17:32.53*** join/#asterisk nullable_type (n=nullable@76.77.70.226)
17:34.45nullable_typeI am having problem detectign the DTMF. With g729 codec in use, is it right that SIP Provider should signal
17:39.58*** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be)
17:47.48*** join/#asterisk war9407 (i=war@liquidswords.org)
17:48.25nullable_typehey guys how the DTMF detection is done for g729, is it rfc2833 or sip info
17:49.01*** join/#asterisk seanmh (n=johndoe@198.59.129.24)
17:52.50*** join/#asterisk waltervargas (n=usuario@190.207.69.66)
17:54.00waltervargasanybody test Asterisk1.6 with res_ldap  ?
17:54.34infinity1how can i troubleshoot call quality issues?
17:54.55infinity1i'm using ilbc, voicepulse, and asterisk. sometimes call quality sucks
17:56.41*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
18:01.09*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
18:03.10*** join/#asterisk intralanman (n=lanman@173-102-67-218.pools.spcsdns.net)
18:06.59brunneruhg
18:07.07brunnerI can't find a reliable CNAM provider for the life of me
18:08.47nullable_typeQwell >> Are you there
18:08.58Qwelldepends
18:10.10nullable_typeI was able to send and receive DTMF through the phone, Not through the DialPlan anymore using Read or Send DTMF. It used to work, suddenly stopped after changing provider.
18:11.05[TK]D-Fender[13:48]<nullable_type>hey guys how the DTMF detection is done for g729, is it rfc2833 or sip info <-- yes, OOB for highly compressed codecs
18:11.05nullable_typeAnd i am wondering if its something to do with g720 codec
18:11.10*** join/#asterisk VoipForces (n=kvirc@mail.net-forces.com)
18:11.22VoipForcesHi, anyone familiar with Aastra XML ?
18:11.55nullable_typeD-Fender >> Sorry, did you mean it should be SIP Info and not RFC2833 for highly compressed type like g729?
18:12.11VoipForcesTrying to have a 33i dial via an xml script over the line 4 which is programmed on prgkey7, but it always dials ober line 1...
18:14.09rue_mohranyone know a program to filter out non-relivent settings from sip.conf to make a polycom phone take a few less hours to boot?
18:14.11[TK]D-Fendernullable_type: Either.  Those are 2 different OOB signaling standards.
18:14.41Qwell[TK]D-Fender: 4
18:14.45[TK]D-Fenderrue_mohr: specify the phone's model-specific firmware and not the general sip.ld
18:14.47Qwell(at least)
18:15.09[TK]D-Fenderrue_mohr: And mine took 2 mins at worst,.  Besides, it you do it right you only have to do it once anway
18:15.11rue_mohrhu? specify it where?
18:15.37[TK]D-FenderQwell: I said "those" were 2.  Didn't imply there might not be more :)
18:15.43jplankrue_mohr: upgrade the bootroom to something over 4.0 and use the newest split firmware
18:15.46[TK]D-Fenderrue_mohr: <mac>.cfg
18:16.06Qwelloh, thought you said "there are 2"
18:16.26[TK]D-FenderQwell: Semantic Anti-Virus :p
18:16.27rue_mohrI have to keep rebooting cause I cant get the settings right to make it use distinctive ring via exten => 10,1,SIPAddHeader(Alert-Info:Internal)  via http://www.technicallyamusing.com/?p=44 cause the god damned documentation on polycom phones sucks
18:17.00Qwellumm.  the documentation on polycom phones is amazing.
18:17.05[TK]D-Fenderrue_mohr: Docs are jsut fine, there are perfect WIKI pages for this and that line COULD be correct but you aren't showing us your phone's configs
18:17.46rue_mohrno, aastra docs are amazing, polycom docs suck
18:17.48rue_mohr:)
18:18.00rue_mohrI hate to think whats worse than the polycom docs
18:18.54Nilzao[TK]D-Fender: i discovered that my telco don send the CDR signal, just the "busy tone" when the caller hangup
18:19.12rue_mohrok, back to what you said, there is a way in <mac>.cfg to specify the phone model to make it boot faster
18:19.17Nilzao[TK]D-Fender: so i will wait the timeout...
18:19.28[TK]D-Fenderrue_mohr: the only common factor with all of your dysfunctional relationships with otherwise generally loved equipment... is YOU :p
18:19.39rue_mohrsip.ld is the firmware...
18:20.02[TK]D-Fenderrue_mohr: *A* firmware.  that is the COMPOSITE one.  Use the broken-up version <-
18:20.07rue_mohrI'v had almost no problems with the aastra phones, and when i did I called them and got good support
18:20.25rue_mohrI had to try 3 differnt numbers for the polycom phone and got an "i dunno"
18:20.43[TK]D-Fenderrue_mohr: Your reseller is supposed to support you directly, not Polycom.
18:20.48rue_mohryea
18:20.51[TK]D-Fenderrue_mohr: Unless you've cert'd with them....
18:20.53rue_mohrthats what my reseller said
18:21.04[TK]D-Fenderrue_mohr: So go nag them instead :)
18:21.24rue_mohrI'v been waiting over 3 months for an answer to having the phone show digits dialed DURING the call
18:22.01[TK]D-Fenderrue_mohr: Ah, that would fall under "feature request".  They have a forum for that
18:22.17rue_mohrwhat I need is someone who sells voip hardware in west canada without shipping it customer direct from the us
18:22.17[TK]D-Fenderrue_mohr: And like normal, don't hold your breath
18:22.23VoipForcesagree with rue_mohr about Polycom...
18:22.34rue_mohryou sure that the phone cant do it?
18:22.57[TK]D-Fenderrue_mohr: I've never seen an option, and it sure doesn't do it by default.
18:23.07rue_mohrat any time I can call aastra and get a proper staight answer, but not with polycom
18:23.21[TK]D-Fenderrue_mohr: But of course if the feature isn't there, it must be a bug right?  One for which they MUST correct on the spot for you.
18:23.33rue_mohrno I just cant find it cause the docs suck
18:23.52[TK]D-Fendercomplains to Chevrolet for the missing wings & rudder for his 1998 Malibu.
18:24.05jplankI love the polycom docs
18:24.05[TK]D-Fenderrue_mohr: You can't find it because the feature DOESN'T EXIST
18:24.12jplank^^^^^^^
18:24.27*** join/#asterisk ruben23 (n=AGENT@124.107.3.178)
18:24.50rue_mohrwell how the heck am I supposed to know what does and dosn't exist!?
18:25.07[TK]D-Fenderrue_mohr: If you can't find it, TAKE THE BLOODY HINT
18:25.15coppice[TK]D-Fender you should have thought of that while they were still in business
18:25.16*** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net)
18:25.26rue_mohrcan this phone do math!? I dont know, the manual says "we dont recommend changing this setting" on almost everything
18:25.47[TK]D-Fenderrue_mohr: You seem to think every product offers every feature that anything similar does.  this sense of entitlement is completely out of whack.
18:26.01rue_mohrif you google you find out that lots of people have problems with polycom volume levels and ... I dotn remmeber now
18:26.44rue_mohrthe aastra phones are that, phones, not web browsing singing mp4 player
18:26.54[TK]D-Fenderrue_mohr: And you're the only one in here with issues on them.  In fact you seem to have issues with just about everything.  I'm seeing a pattern forming here.
18:27.02joakoIn all these years I never even NOTICED the polycoms don't show the digits dialed during a call
18:27.09VoipForcesrue_mohr: carefull, the xml features of the aastra are retty powerfull
18:27.11jplankrue_mohr: people with polycom volume isues, don't know how to setup the phones
18:27.12[TK]D-Fenderhands rue_mohr another square peg
18:27.39rue_mohrthe great thing is the aastra manual tells you in detail about every feature and what it does, with examples of how to use it
18:27.53jplankso does the polycom docs
18:28.23rue_mohrI dare you to tell me what the different volume controlls are and how they relate on a polycom phone from the manual, it just says "there is the setting, dont touch it"
18:28.34jplankthats not what it says
18:28.42jplankit says its not RECOMMENDED to be touched
18:28.47rue_mohrgo ahead, find me a description of "chassis gain"
18:28.53jplankreally
18:28.57rue_mohrreally
18:28.59jplankthats not descriptive enough for you?
18:29.05jplankhow about IP600 gain
18:29.18rue_mohryou tell me then, what sounds are effected by chassis gain
18:29.22Qwellspeaker.
18:29.25jplanktake a guess
18:29.31Qwellwhat else would it be?
18:29.37rue_mohrno, thats the speaker gain
18:29.37Qwellthere's also handset and headset gain...
18:29.38jplankhow many speakers do you see on the chassis?
18:30.23rue_mohrwhen I looked over it, there were contradictions
18:30.28jplanklike that?
18:30.31jplanklike what*
18:31.10rue_mohreverything was there, then there was more, and there was no list of what controlled what, or what context recieve and transmit are (which shouldn't get me but always does)
18:31.16VoipForcesrue_mohr: this is getting no where and you won't get answers like that. I too do not like Polycom, but please tell us what you need and if someone knows the answer they will speak up
18:31.33joakoIf you have the latest firmware and get the latest "administrator guide" from polycom.com I don't see why you have issues. Everything is exmaplained in detail.
18:31.48rue_mohrok, what I need to do now, is get exten => s,1,SIPAddHeader(Alert-Info:Internal) to cause a polycom phone to ring with a different tone
18:32.12rue_mohraastra manual 12000 pages     polycom manual 1100 pages
18:33.33[TK]D-Fenderrue_mohr: Yes, and where is your sip.cfg so we can see where you set up that call?
18:33.38[TK]D-Fenderclass*
18:34.24rue_mohrI do not modify sip.cfg (or I try not to) BECAUSE ITS DUMB when the next version of sip.cfg comes along, you have to start all over again, I have office_sip.cfg, which is all my changes
18:34.27rue_mohri shall post it
18:34.44jplankrue, I'll point you in the right direction
18:34.55rue_mohr:)
18:34.55jplanktake a look at how the alertinfo "Ring Answer" is setup
18:34.56[TK]D-Fenderrue_mohr: You have to.
18:35.19[TK]D-Fenderrue_mohr: go read the WIKI page on Polycom paging <-----
18:35.22joakorue_mohr: If you read the release notes they document the changes. I am not aware in the past few years any changes that caused compatibility issues. For a while I was running 3.x firmware with 2.x config files, very little was changed
18:35.49jplankrunning 3.x with 2.x config files probably isn't the best idea ;)
18:36.04[TK]D-Fenderjplank: Entirely not.
18:36.10barbachahey, any idea why ## would not work to transfer a call ?
18:36.19rue_mohrjplank, this isn't for paging right now... ring answer?
18:36.26jplankright
18:36.36rue_mohrthis is for different ring tone for internal calls
18:36.42jplankbut ring answer is setup exactly how you would do it for something else
18:36.46joakojplank: I did read the release notes, the only changes were for new features added....
18:37.03rue_mohrhttp://www.pastebin.ca/1448957  <-- my changes to the default sip.cfg
18:37.08jplankI've had huge compatibility issues in the past doing that
18:37.14joakorue_mohr: BTW, the answer to your question is on page A-30 and 4-7 of the Polycom administrator manual...
18:38.07jplankmore annoying then anything
18:38.07rue_mohrflips to page
18:38.07rue_mohrI'm on page 4-7
18:38.07jplankI usually use XML notepad and just replace my changes onto the new config
18:38.07jplankplus, I usually don't upgrade the firmware unless needed
18:38.16jplankthe main reason I'm using 3.1.1 is because of the nice new interface
18:38.18*** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-a9e462ad021e83f6)
18:38.23jplank3.2 is going to be sweet though
18:38.43ariel_argh I just love it when my network drops me...
18:39.06jplankmibbit is the devil
18:39.15joakoI actually just checked one install I have config files Revision: 1.67 and firmware 3.1.2.0392!
18:39.30jplank3.1.2 is almost the newest
18:39.36jplank3.1.3 just came out
18:39.45jplank(revb of course)
18:40.11rue_mohris that config missing anything for  the distinctive ring?
18:40.29rue_mohrI'm looking but I suspect I'm not seeing
18:40.42ariel_mibbit might be the devil. But it's sometimes the only way to get to chat via ship's slow cband and most firewalls.
18:41.05jplankrue, is that your whole sip.cfg?
18:41.25rue_mohrno, thats loaded after the default factory sip.cfg
18:41.36[TK]D-Fenderrue_mohr: You have no classes defined because your tried to do it yourself
18:41.48*** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com)
18:41.54rue_mohrI dont modify the sip.cfg
18:41.59jplankwhere are you assigning class 5 to a ringtone?
18:42.00[TK]D-Fenderrue_mohr: And did so with the absolute minimum you thought you could get away with
18:42.10[TK]D-Fenderrue_mohr: the stock sip.cfg is PAGES worth
18:42.14rue_mohrwhat I did
18:42.22jplankyour missing a WHOLE lot
18:42.23[TK]D-Fenderrue_mohr: Yours is no good at all
18:42.34rue_mohrwas to read thru http://www.technicallyamusing.com/?p=44   and try to peel out the stuff about eth custom wave file
18:42.51jplankyour not even uploading a custom wav to the phone
18:43.08jplankyour not defining class 5 to a wav
18:43.10rue_mohr<APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="charles.cfg, office_sip.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="" ....
18:43.31rue_mohrso the user overrdes the office that overrides the factory defaults
18:43.33*** part/#asterisk scooby2 (n=scooby2@pdpc/supporter/active/scooby2)
18:43.51jplankrue, all that config is saying is assign alertinfo Internal to class 5
18:43.57*** join/#asterisk Dr-Linux|home (n=Nothing@221.132.117.17)
18:43.57jplankbut your not saying what class 5 is
18:44.01rue_mohrjplank, from what I could tell, in that webpage, it already exists and they were re-pointing it to a custom file
18:44.05[TK]D-Fender[14:37]<rue_mohr>http://www.pastebin.ca/1448957 <-- my changes to the default sip.cfg <- what do you mean your CHANGES to sip.cfg
18:44.12rue_mohrI dont know what class is
18:44.17rue_mohrits a ring type isn't it?
18:44.33jplankin so many words, yes
18:44.57[TK]D-Fenderrue_mohr: the alertinfo relates to ring types and 2/3 of the way down the main
18:45.12jplankI don't think class 5 is even defined by default
18:45.13rue_mohrtk, understand this, sip.cfg is defualt factory, its overriden by office_sip.cfg which is global to the office phones, and that is overriden by user_sip.cfg for person specific things
18:45.16jplankI believe class 8 is though
18:45.50rue_mohrI could not find a class->ring definition mapping
18:46.40rue_mohrhe is changing se.rt.5. from "4" to "14"
18:46.50jplankI'm sorry, class 5 is enabled by default
18:46.56rue_mohrI assume 4 is the built-in and 14 is a custom
18:47.02jplank<INTERNAL se.rt.5.name="Internal" se.rt.5.type="ring" se.rt.5.ringer="2" se.rt.5.callWait="6" se.rt.5.mod="1" />
18:47.21rue_mohrthats the factory?
18:47.25jplankyea
18:47.28jplankbut by default
18:47.31jplankits set to ring normally
18:47.35rue_mohraaah
18:47.42rue_mohrthat would confuse the hell outa me
18:47.49rue_mohrso what I have might be working
18:47.56jplankno
18:47.58jplankwell yes
18:47.59jplankbut no
18:48.12jplankyou still have to upload a different wab
18:48.13jplankwav
18:48.15rue_mohrso I want to change.... .ringer to something else?
18:48.19rue_mohrwhat!?
18:48.20jplankand assign that class to the wav
18:48.28rue_mohrwhy cant I use another built in one?
18:48.34jplankI guess you can
18:48.38rue_mohrok...
18:49.17rue_mohron page A31 I think they have a list of built in patterns
18:49.31rue_mohr12-24 are empty, so they say
18:49.49jplank13-24
18:50.03rue_mohrok
18:50.07jplanklook in <RINGER></RINGER>
18:50.37rue_mohrok, so my record points to the 5th item of some list...
18:51.03jplankyour pointing it to the same ring tone as the default
18:51.21rue_mohrwhich is by factory called "Internal" and points to ringer type 4, maybe
18:51.24jplanklook inside <ringType>
18:51.33jplankringer type 5
18:51.39rue_mohrisn't 2 default?
18:51.45jplank2
18:51.56jplankringertype and ringer are different
18:52.09rue_mohrarg
18:52.12jplankclass 5 points to ringtype 5 which points to ringer 2
18:52.42rue_mohrso I want ot change ringtype 5
18:52.47jplankyea
18:52.47rue_mohrk
18:52.56rue_mohrthis is helping muchly, thankyou
18:52.56jplankwhich is <INTERNAL>
18:54.55infinity1what happened to ilbc in asterisk 1.6 ?
18:56.06rue_mohr<INTERNAL se.rt.5.name="Internal" se.rt.5.type="ring" se.rt.5.ringer="2" se.rt.5.callWait="6" se.rt.5.mod="1"/>  <-- thats from sip.cfg, I need to override that to be ringer 5
18:56.12rue_mohrk
18:56.16jplankyea
18:56.20jplankor whatever
18:56.40rue_mohrnow I have to figure out where in the horribly formatted xml doc that is
18:56.43jplankringer 5 would be SHORT_DOUBLE_C3E3_MAJOR_WARBLE
18:56.52[TK]D-Fenderrue_mohr: And that ringeriself is difined higher up
18:56.54jplankwhere is what?
18:56.58rue_mohras long as its different I dont mine
18:57.09rue_mohr<INTERNAL> is under something
18:57.11hardwireany way for asterisk cdr_odbc to include the hostname of the asterisk box?
18:57.14rue_mohrthats under something else
18:57.14hardwireper row
18:57.28jplankrue, use an XML notepad if you are having trouble reading the XML doc
18:57.35[TK]D-Fenderrue_mohr: Under the blatant <ringer> key
18:57.56mmlj4hardwire: set a global variable in extensions.conf... you know, hard wire it :-)
18:57.58jplankactually isn't it tone/ringer/ringertype
18:58.10rue_mohrno ringType, thats under....
18:58.24jplankno
18:58.25*** join/#asterisk metfan2007 (n=jc@189.146.139.21)
18:58.33jplankpatterns/callprogress
18:58.46jplankpatterns/ringer
18:58.51hardwiremmlj4: haw
18:58.53hardwiremmlj4: haw
18:58.58hardwiremmlj4: haw.......
18:59.01hardwirethanks tho.. u roick
18:59.02jplank(ignore callprogress)
18:59.02[TK]D-Fenderjplank: No, not callprogress
18:59.02hardwirerock
18:59.19[TK]D-Fender<RINGER>
18:59.20jplankyea, I don't know what I said callprogress
18:59.32mmlj4seriously, about setting a variable...
18:59.57rue_mohrso far I have sound_effects.patterns.ringtype    <INTERNAL
19:00.09metfan2007Hi all, I'm trying to config an Asterisk box for inbound/outbound campaigns for the same agents, at the same time, so an agent can receive a call from a queue and generate calls without login/logoff
19:00.27rue_mohrso far I have tones.sound_effects.patterns.ringtype    <INTERNAL
19:00.30jplankI was just about to say that
19:00.51*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
19:00.52[TK]D-Fenderrue_mohr: "se.pat.ringer.5" <- SEARCH
19:00.54rue_mohrso far I have sip.voIpPort.tones.sound_effects.patterns.ringtype    <INTERNAL
19:00.56metfan2007the problem is that the agent is receiving calls from queue while he is in a outbound call, how do I tell Asterisk that Agent/XXX is in a outbound call and don't send calls to him?
19:01.02jplankyou want to change sound_effects.ringtype
19:01.08jplank.internal
19:01.14rue_mohryea, its not written in such I way I know what the hiarchy is to it
19:01.19jplankse.rt.5.ringer
19:01.25jplankrue, use XML notepad
19:01.30rue_mohr!! they abbreviated it!
19:01.59ariel_metfan2007: your agent needs to pause out of the queue to make the outbound calls
19:02.02[TK]D-Fenderruehow DARE they!
19:02.18rue_mohrbuts its under sip.voIpPort.tones.... isn't it? or did my inder finger slide while I was scrolling?
19:02.42metfan2007ariel_, I already tried the pausememberqueue and unpause, pause just before the Dial() for outbound and Unpause in the h extensions
19:03.02[TK]D-Fendermetfan2007: make your dialplan pause them before dialing out, and unpause on disconnect
19:03.17[TK]D-Fendermetfan2007: Ah, well taht's what i'd suggest
19:03.23jplankwhat are you looking for again?
19:03.32jplankyou want to make Internal point to a different ringtone right?
19:03.45*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
19:04.28jplankrue just search for <INTERNAL se.rt.5.name="Internal"
19:04.37jplankand change se.rt.5.ringer="2"
19:04.39metfan2007[TK]D-Fender, the problem is that some Agent's are still receiving calls :S
19:04.41jplankto whatever you want
19:04.50jplankbrb
19:05.16rue_mohrI ahve it, but I WILL NOT edit the sip.cfg, I need to know wehre it is so I can add it to my office_sip.cfg
19:05.31[TK]D-Fendermetfan2007: Do show us your code, the call, the status before and the call itself
19:06.14*** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com)
19:06.22metfan2007[TK]D-Fender, http://pastebin.ca/1448981
19:07.11rue_mohrxml notepad from m$ ?
19:07.14*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
19:07.41*** join/#asterisk oversillo (n=overdriv@190.207.69.66)
19:08.31rue_mohrhahah no .net framework
19:09.57*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
19:10.00KavanSany recommendations for an osx asterisk client?
19:10.22[TK]D-FenderKavanS: What is an "asterisk client"?
19:10.31KavanSdrr sorry still asleep
19:10.33[TK]D-FenderKavanS: "asterisk" isn't a protocol
19:10.35KavanSosx sip client :P
19:10.37KavanSI know...
19:10.55KavanSsometimes we type things that we do not mean...that was one of them for me
19:11.16jplankrue, xml notepad was the best free one I found, I usually use notepad+, but using a real XML program will show you any typos
19:11.24rue_mohrok, sip.voIpPort.tones.sound_effects.patterns.ringtype    <INTERNAL
19:11.36jplankright
19:11.46rue_mohrI cant use m$ xml notepad cause I dont have .net cause I donthave sp3
19:11.48jplankchange se.rt.5.ringer="2"
19:11.50KavanSanyone, osx sip client recommendations? looking at the list on voip-info but it looks pretty shaky
19:11.59rue_mohrhave to create it first...
19:12.15jplankyour causing yourself more trouble that way
19:12.22jplankwhy not just edit the default one
19:12.29jplankthats the recommended way of doing it
19:12.38oversillohi, sorry for my bad english, somebody here has test asterisk 1.6 with OpenLDAP ????
19:12.42rue_mohrcause when they come along with another version I dotn want ot lose my changes
19:13.05jplankrue, you know you don't HAVE to upgrade the firmware everytime a new one comes out, right?
19:13.17rue_mohrok, sip.voIpPort.tones.sound_effects.patterns.ringtype    <INTERNAL
19:13.24rue_mohrit happened once, that was enough pain
19:14.19metfan2007[TK]D-Fender, any clue?
19:15.21[TK]D-Fendermetfan2007: You showed me aa thrifd of what I asked for.
19:15.26[TK]D-Fenderthird*
19:16.22metfan2007[TK]D-Fender, oh, ok ok
19:20.46rue_mohroh its not in patterns
19:20.52rue_mohrok, sip.voIpPort.tones.sound_effects.ringtype    <INTERNAL
19:21.26rue_mohror tones
19:21.30rue_mohrok, sip.voIpPort.sound_effects.ringtype    <INTERNAL
19:22.16jplankedit the main sip.cfg and you wont have this issue
19:22.26rue_mohrnot aprt of voIpPort either :)
19:22.32rue_mohrok, sip.sound_effects.ringtype    <INTERNAL
19:22.38rue_mohrthis is getting easier
19:24.39rue_mohr10 mins to wait for hte thing to reboot and I get my answer!
19:25.50jplankrue, did you follow my suggestion earlier on upgrading your bootrom to 4.0 (or higher) then using the split firmware?
19:28.01[TK]D-Fenderjplank: LOL... *as if*
19:28.55rue_mohrits upgraded
19:29.14rue_mohriirc
19:29.26rue_mohrbut that was what bit me, I dont want to upgrade anymore :/
19:29.41rue_mohrok, the distinct ring didn't work
19:29.51[TK]D-Fenderrue_mohr: unload chan_burnout.so :)
19:30.27[TK]D-Fenderrue_mohr: pb your call attempt, and your new provisioning info
19:31.30rue_mohrhttp://pastebin.ca/1449004
19:31.47rue_mohrand asterisk is setting the info
19:32.00rue_mohr<PROTECTED>
19:32.00rue_mohr<PROTECTED>
19:32.07rue_mohrdamn, sorry
19:32.35rue_mohrout for lunch, back in half hour
19:32.57*** join/#asterisk fskie (n=shiltron@66.252.6.2)
19:33.54*** join/#asterisk AD2008 (n=RatX@wbs-41-208-227-229.wbs.co.za)
19:35.04fskieHi guys, i've got a conspiracy theory about how Sangoma pays google to hide searches that make it possible to comparison shop for their products :) and I devised an experiment to prove my theory
19:35.47fskieI posted this on June 1st http://www.pastebin.ca/1444018?srch=sangoma+s519
19:36.09fskieit is number 4 search on pages from Canada: http://www.google.ca/search?hl=en&q=sangoma+s519&meta=cr%3DcountryCA the link is no longer accessible?
19:36.46Qwellwhat?
19:38.11fskiewell people pay for preferencial ranking right?
19:38.40Qwellno..
19:39.30fskiewhat are all those advertisements at the top of the google page?
19:39.35Qwelladvertisements.
19:40.10fskiewell people also pay for keywords to be blocked
19:40.36QwellIn ads.
19:40.47Qwellads are not results.
19:42.07fskieif you sell something, you pay google not to show your competitors
19:42.22[TK]D-FenderQwell: They are the result of marketers paying Google :)
19:43.48fskieanyway I posted a pretend advertisement on pastebin.ca and google has blocked it?
19:44.57QwellFirst, google didn't block it.  Second, it isn't an advertisement.
19:46.05fskieit gives a description + price for a product that somebody has payed to have blocked
19:46.09*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
19:47.07QwellSo...what do you want it to do?  List every URL on every domain that lists the keywords you searched for?
19:48.13oversillohi, sorry for my bad english, somebody here has test asterisk 1.6 with OpenLDAP ???
19:48.19fskieQwell you will see eventually
19:48.53QwellYou always want...what?  them to index unlinked URLs?
19:49.35fskieit is the fourth search result on the page but it is blocked
19:49.42Qwellhow is it blocked?
19:50.05fskieit is number 4 search on pages from Canada: http://www.google.ca/search?hl=en&q=sangoma+s519&meta=cr%3DcountryCA the link is no longer accessible?
19:52.17fskieQwell anyway sorry for ranting i'll shut up now
19:52.32QwellHow do you figure it's blocked?
19:53.06fskiebecause it was accesable for three day now it is not?
19:53.23Qwellit's perfectly accessible.
19:54.35fskieare you sure what does it say?
19:54.47QwellWhat does what say?
19:55.05fskienever mind man
19:55.21fskiegoogle is evil
19:55.23Qwellgo put your tinfoil hat back on
19:55.43fskieone day you will see
19:55.54jplankI'm in the middle of swapping out PBXs at a restaurant that has two locations, guess what I'm swapping out...
19:56.11jplankboth have trixbox appliances :)
19:56.25Qwelljplank: people actually buy those?
19:56.29jplankI guess
19:56.34jplankthey said we could have them
19:56.37QwellI thought they were an April Fool's joke
19:56.39jplankthey were going to throw them away
19:56.49*** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net)
19:57.05jplankI'm taking it back to our labs an dissecting them
19:57.15*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
20:05.19rue_mohrin that howto, they point the Internal ringer to a sampled audio file that custom, but I dont see where you would use one of the normal internal ones
20:05.31rue_mohrhe sets up saf.2
20:06.04rue_mohrbut then he sets ringer to 4
20:06.25rue_mohrsorry, from 4 to 14
20:06.37rue_mohris saf.2  #14?
20:06.46rue_mohrthat would make 1 13
20:07.15rue_mohrwhcih would exaplain what jplank said
20:11.33rue_mohroh I had the wrong table, ok
20:13.07rue_mohranother nice thing about the aastra phones is that you can download the working config, and see what the results of its config files and local files was
20:20.24[TK]D-Fenderrue_mohr: Yes, each maker does some things better than others.  As much as I love Polycom for all that it lets me do, they don't hold a candle to Aasrta's soft-key programmability which is GODLY
20:23.37rue_mohryay! it was just a typio!
20:24.04rue_mohrI dont liek the web interface, you can really do anything, you can do almost everything from an aastra interface
20:24.42rue_mohrhaha, they pointed out they want a different ring for every line :)
20:24.44rue_mohrsigh...
20:24.46QwellSo write a GUI for it.
20:24.50jplankweb interfaces are almost pointless on phones
20:24.51Nilzaothanks for all help, i will do some workaround then i come back
20:24.54Nilzaogood bye
20:24.59rue_mohrmaybe I should make recordings of each line name :)
20:25.02*** part/#asterisk Nilzao (n=nils@189-68-191-170.dsl.telesp.net.br)
20:26.06*** join/#asterisk layne (n=layne@c-68-37-72-150.hsd1.nj.comcast.net)
20:29.52[TK]D-Fendercheckout time, bbiab
20:39.19*** join/#asterisk Nunners (n=james@82.147.18.2)
20:40.59NunnersCould anyone suggest a piece of kit that will allow me to put in a Mobile SIM card and then allow me to channel calls through that SIM card?  I know the hardware exists, but can't remember what it's called!
20:42.37rue_mohrok
20:42.38rue_mohrso
20:42.54rue_mohrwho has the dtmf codes for the song "camptown races" ?
20:43.24*** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-206.phlapa.fios.verizon.net)
20:43.37rue_mohrI only have "hot cross buns"
20:43.55rue_mohr(32132111112222321)
20:45.50NunnersOk - so it's a GSM gateway, but can anyone recommend one?
20:46.10rue_mohrhmm?
20:46.50rue_mohrQwell, do you have dtmf for the song "camp town races"?
20:47.15AD2008tries to remember the tune to camp town races and fails
20:48.27rue_mohr"camp town races sing that song, doo dah, doo dah, camp town races, 5 miles long, oh doo dah day
20:48.35rue_mohr?
20:48.45AD2008okay, now i got it
20:49.01rue_mohryou want to work on it for me while I rebuild my dahdi drivers?
20:49.04AD2008i just sang that to myself, i really shouldn't be left on my own
20:49.32rue_mohrits phone work, everyone will suspect your busy doing something important
20:49.46rue_mohr604-958-2091
20:49.51rue_mohrsilent termination
20:50.08rue_mohrthey will never know
20:50.29rue_mohryou might have one more local, check the e-slut directory
20:50.51AD2008wow
20:55.30ariel_is just sitting making his patch panel for the setup of 2 E1' and 1 E1 X/O, while he listens to Pink Floyd....
20:58.16*** join/#asterisk cvnet (n=dahitler@24.156.134.228)
20:59.40cvnethello
21:00.14cvnetafter sending calls to Virizon and didnt go through, i have contact them and they send me this back Your calls are showing multiple timers that are incompatible. (For example packetization timer of 150 ms on g711Ulaw64k and g711Alaw64k calls). You are sending in incompatible packetization timing on your OLC (open logical channel)'s codec list" i cant make any sense out of it, any suggestions?
21:01.03rue_mohrvirizon provides the sip?
21:01.08cvnetfull answer http://pastebin.com/m16aba82e
21:01.11cvnetno h323
21:01.20cvnetalso sip but you have to do ipsec
21:04.45cvnetany suggestions?
21:05.59ariel_disallow=all, allow=ulaw
21:07.25ariel_But in the asterisk setups I do, I only work with Sip, & Zap/Dahdi
21:08.33cvnetdisallow=all, allow=ulaw i have taht already
21:09.40_ShrikEcvnet: Verizon is going to want to see your packetizaion at 20ms.
21:10.23_ShrikEAt least that is what they want to see with SIP
21:10.30cvnet_ShrikE: how do i change/conbtrol pocketization?
21:15.32watchyim having an issue with odbc in realtime not showing connected, i don't even get connected = with odbc show
21:15.35_ShrikEcvnet: I believe the default is supposed to be 20ms
21:16.44watchyodbc show Name: asterisk DSN: asterisk is all i get
21:16.47*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
21:17.29cvnetapprently not, is there a way i can change that?
21:20.06rue_mohrAAAAAAAAAAHHHHHHHHHHH I rebuilt my dahdi drivers and there is no audio now!
21:20.18rue_mohrTHIS IS A PRODUCTION SYSTEM!
21:20.36QwellBetter fix it then, eh?
21:20.52rue_mohrwhats going on!?
21:21.34cvnetcan you change the packetization timing on your OLC ?
21:21.41rue_mohrwait, its a sip problem it worked on a fxs phone
21:24.22bmoracawhy are you rebuilding drivers on a production system?
21:24.48ariel_ok 3 new e1 connections made and tested.  Green lights are so nice to see on the digium boards...;-)
21:26.09*** join/#asterisk scarecrowed (n=fabiano@200.101.115.243)
21:28.10rue_mohrOK Incomming fxo calls have no audio from the dahdi card
21:28.11rue_mohr!?!??!?!
21:28.42rue_mohrbut if we make an outgoing call it works fine
21:29.21rue_mohrHELP!
21:29.23bmoracadid you screw your gain up?
21:29.28rue_mohrNO!
21:29.32rue_mohrbut just a sec
21:29.35rue_mohrthat gives me an idea
21:29.59AD2008if it was H.323 I'd tell you to check your UDP port range
21:30.07AD2008for what that's worth
21:30.11rue_mohrits all a local network
21:30.17AD2008lurks in the corner some more
21:30.25rue_mohrif a call comes in on the fxo channel, there is no audio
21:30.36rue_mohrif we go out on the same channel, there is
21:30.51AD2008that is odd, analog should just work
21:30.53ariel_rue_mohr: are you calling into a menu?
21:31.03rue_mohrno phones direct
21:31.16ariel_the phones are sip ?
21:31.21rue_mohryes
21:31.34ariel_setup a menu on your system and send the call there see if you hear it?
21:31.41AD2008having said that, it brings back memories of setting up activevoice voicemails in the 90's with analog pbx integration - you had to set the DTMF parameters and various tone cadences on every system
21:31.57ariel_or send the call to vm
21:32.04rue_mohrno vm set up
21:32.09ariel_see if you hear that and get the recordings
21:32.27rue_mohrI see where your at, I can set the incomming calls to the milliwatt sec
21:32.29ariel_wow production system, no VM direct to phones
21:32.29rue_mohrsrc
21:32.46rue_mohrno, the vm is via the telco cause were short on lines
21:33.10bmoracaewwww
21:33.15rue_mohryup
21:33.33bmoracafuster cluck
21:33.50rue_mohrI need the system to check the co message indicator and play "camp town races" on the recpetionists phone if there are messages
21:34.12rue_mohrthat'll be part of my paging type programming
21:34.15ariel_rofl
21:34.15AD2008waah haah, so you actually really need camp town races?  classic
21:34.45PagautasHi, i've a problem with mixmonitor and app_queue
21:34.48AD2008what i'm liking here is that you're actually working to match the business requirements
21:34.50rue_mohrno, but she said it jokingly and since I told them this system can do anything I intend to follow thru
21:34.58AD2008i give you a 10 on your performance review
21:35.02Pagautascall comes to a queue, agent answers and mixmonitor starts monitoring
21:35.11Pagautaswhen an agent tranfers a call mixmonitor stops
21:35.12rue_mohrright, we just need the audio to work now
21:35.20Pagautasthe last version where mixmonitor worked corectly is 1.4.17
21:35.30Pagautaswhere could be a problem?
21:35.38*** join/#asterisk goodjoke (i=1827a8fa@gateway/web/ajax/mibbit.com/x-66bc5c857e947656)
21:35.41rue_mohrI'll bet you I know
21:35.53AD2008covers his eyes
21:36.02ariel_Pagautas: that as far as I know is normal as it's a new channel that gets transfered too.
21:36.08rue_mohrits prolly the memcopy function, the same error that caused the dahdi_monitor tool to stop working right
21:36.43rue_mohrthey changed the nature of how it worked, so it would copy zeros on the end of strings
21:36.57watchyanyway to turn on more logging with my ODBC ?
21:37.06watchyso i can see if asterisk is even trying to do anything
21:37.20AD2008me 's avaya never has zeros at the end of strings.  oh wait...
21:37.39AD2008insert a / before that
21:37.44Pagautasariel_: but on the call center it is not acceptable
21:37.54Pagautasi need to have all calls recorded
21:38.51AD2008Pagautas - this is a very good question, i am interested in the same functionality.  have you looked at the changelogs from 1.4.17 onwards?
21:38.53ariel_Pagautas: that is an issue we have always had.  There are other ways around them but you have to join them up in your dial plan.  Or you set them up via meetme or conference and record the conference.
21:39.10rue_mohrAAAARRRGG the phonesystem must have just crashed!?
21:39.19*** join/#asterisk elitecoder (n=liq@apollo.bullethost.com)
21:39.21rue_mohrDAMNIT stupid freaking upgrades
21:39.28rue_mohrdamnit
21:39.29AD2008whoops
21:39.40AD2008what happen?
21:39.41ariel_rue_mohr: why upgrade during working hours?
21:39.45bmoracaadministrators
21:39.47bmoracaerm
21:39.50elitecoderI need to set the caller id before I queue a call so it shows the number I dialed out to, (auto dialer)
21:39.58PagautasAD2008: there is not much changes
21:39.59bmoracaadministrators' mantra:  if it ain't broke, don't fuck with it.
21:39.59Pagautashttps://issues.asterisk.org/view.php?id=11811
21:40.13elitecoderit calls a person, they press 1 and then get put into a queue. I need it to show THEIR phone number on the hand sets, not ours haha
21:40.25Pagautasi think there they changed something
21:40.35cvnetafter sending calls to Virizon and didnt go through, i have contact them and they send me this back "Your calls are showing multiple timers that are incompatible. (For example packetization timer of 150 ms on g711Ulaw64k and g711Alaw64k calls). You are sending in incompatible packetization timing on your OLC (open logical channel)'s codec list" i cant make any sense out of it, any suggestions?
21:40.54Pagautasariel_: by then what do monitor=... function ir queues.conf do?
21:41.11*** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman)
21:41.46Pagautasuuups i meant monitor-type = MixMonitor
21:42.39rue_mohrcvnet, is virizon your sip provider then?
21:43.53*** join/#asterisk intralanman (n=lanman@173-102-39-245.pools.spcsdns.net)
21:44.35watchyanyone mind helping with a res_odbc issue?
21:44.37rue_mohrdigium gave me an svn copy of the dahdi-linux-complete  hwo do I know what version it was and if there were issues with it?
21:45.03rue_mohrwatchy, I just need to get my system working again first
21:45.07cvnetue_mohr yes, but we are sending calls h323, since we couldnt figure how to do ipsec, so the calls are getting send via h323
21:45.12watchytry a different version of dahdi
21:45.37rue_mohrmy old files worked, kinda
21:45.48watchyhaha
21:45.52leifmadsenrue_mohr: it'd just have been the latest version at the time
21:45.52watchywhat the hell is kinda
21:46.06rue_mohrwell that was the reason for hte upgrade, there were a few audio problems
21:46.20watchywell if you have 0 audio now
21:46.25watchytry to back down 1 version
21:46.28rue_mohrlike dtmf with the polycom phones
21:46.32rue_mohris really screwed
21:46.32watchysee if itts some how the issue
21:47.04rue_mohrI'm gonna try to go back to 2.1.0.3 with the dahdi stuff
21:47.07watchytry another version
21:47.12watchysee what happens
21:47.49rue_mohrI'm trying ! i'm trying!
21:48.11[TK]D-Fenderrue_mohr: Why did you just change in the first place?
21:48.16rue_mohrslides down in his seat to escape the gaze of the 15 people behind him
21:48.41rue_mohrcause polycom dtmf is screwed up, and digium said it was fixed in the new dahdi drivers
21:49.57elitecoderI have a question. I set my callerid before I call an outgoing number, but then that call is added to a queue once it's answered and locally the person sees the same caller id. I need the local person to see THEIR phone number, the number that was dialed. Any help?
21:50.10rue_mohrthey also said problems with non-duplex audio were fixed
21:50.17rue_mohrsiplex
21:51.37*** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home)
21:55.43rue_mohrmaybe I shoudl try 1.2.25 from 1.2.22
21:55.48rue_mohrlater
21:57.51*** join/#asterisk [acer]lanman (n=lanman@173-102-39-245.pools.spcsdns.net)
21:59.20watchy[Jun  5 16:57:43] WARNING[3969]: loader.c:478 ast_unload_resource: Firm unload failed for res_odbc.so
21:59.20[TK]D-Fender[17:48]<rue_mohr>cause polycom dtmf is screwed up, and digium said it was fixed in the new dahdi drivers <-- PARDON?
21:59.24watchyanyone know whats uop with that?
21:59.46[TK]D-Fenderrue_mohr: What does SIP DTMF have to do with DAHDI?
21:59.58rue_mohrlibtonezone
22:00.10rue_mohrseems to be the relation
22:00.21[TK]D-Fenderrue_mohr: NONE.
22:00.30[TK]D-Fenderrue_mohr: completely different channel drivers
22:00.39rue_mohr?
22:00.45rue_mohrhmm
22:00.57rue_mohrstarts to wonder why digium tossed him this update
22:00.59[TK]D-Fenderrue_mohr: DAHDI only fixes how * deals with DAHDI communication.
22:01.20rue_mohrI didn't mention our volume levels
22:01.21*** join/#asterisk phy (n=phy@funky.monkey.org)
22:01.25rue_mohrI did mention echo
22:03.42watchyuse oslec
22:04.11*** join/#asterisk thuddwhirr (n=wolthuis@mimezine.com)
22:04.18rue_mohrI have a hwec
22:04.41watchythen you have a major fin issue somewhere
22:05.29*** join/#asterisk viq (n=viq@unaffiliated/viq)
22:06.02watchyanyone got a good res_odbc howto?
22:06.10rue_mohrI was able to do it
22:06.13Qwellwatchy: it's in use.  don't force an unload of something.
22:06.15rue_mohrbut the system is rebooting
22:06.34watchyqwell: my odbc show isn't showing what examples show
22:06.41watchyand i'm getting no debug info
22:06.49rue_mohrI did mine with gpsql
22:07.05watchyin every example people have Connected = something
22:07.07watchyi don't even have
22:07.08thuddwhirrhas anyone ever seen a situation where you Dial an IVR via SIP, the IVR starts sending back audio imediatly, and Dial does not awknowledge that the call was answered, hanging up after timing out?
22:07.08watchythat
22:07.37Qwellthuddwhirr: 800 number probably?
22:07.40thuddwhirryeah
22:07.55Qwellearly media
22:08.05thuddwhirrQwell: exactly. you know any fixes?
22:08.26thuddwhirris hopeful
22:08.35Qwellnope
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22:10.59thuddwhirrhrm
22:14.35rue_mohrk, now we have major echo problems
22:14.47rue_mohrwith default levels
22:15.24rue_mohr[Jun  5 15:13:44] WARNING[2774]: chan_dahdi.c:1499 dahdi_train_ec: Unable to req                                                            uest echo training on channel 5: Invalid argument
22:15.26rue_mohr!?
22:15.39rue_mohrso something went wrong with the recompile!?
22:18.36[TK]D-Fenderrue_mohr: Said invalid arg... sounds like config issue, and we don't see any
22:18.37rue_mohrwhere do I loook?
22:18.43QwellInvalid argument is probably errno from the ioctl()
22:19.13rue_mohrman do I hate...
22:19.33rue_mohrI have an hour to get the system working again
22:19.48rue_mohrmust never never never use svn version
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22:24.05watchywhat the shit is up with this res_odbc
22:25.08watchyits liek not attempoting to connect to my sql or something
22:25.19rue_mohrI can hel you when I get s
22:25.25rue_mohr[Jun  5 15:24:40] WARNING[2720]: chan_dahdi.c:1499 dahdi_train_ec: Unable to request echo training on channel 5: Invalid argument
22:25.30rue_mohrok I need ideas FAST
22:25.44rue_mohrthis is a hwec
22:25.58watchythen turn off echo training
22:26.14watchyare you running asterisk or freepbx
22:26.17rue_mohrhows it gonna cancel echo?
22:26.19rue_mohrasterisk
22:26.28*** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.1.1 (2009/06/05), Asterisk 1.6.0.10 (2009/06/05), 1.4.25.1 (2009/06/05), *-Addons 1.6.1.0 (2009/04/28), 1.6.0.2 (2009/05/21), 1.4.8 (2009/04/28), dahdi-linux 2.1.0.4, dahdi-tools 2.1.0.3 (2009/02/03), Libpri 1.4.10 (2009/04/18) -=- Related channels: #asterisknow #asterisk-gui #switchvox #freepbx #asterisk-bugs #asterisk-dev #asterisk-commits
22:26.35watchyhmm
22:26.41watchyyou google that error?
22:26.47leifmadsenAsterisk 1.2.33, Asterisk 1.4.25.1, Asterisk 1.6.0.10, and Asterisk 1.6.1.1 are now available!
22:27.00Qwellturn off echo training.
22:27.10leifmadsenSee http://www.asterisk.org/node/48598 for more information
22:27.17rue_mohrwill it still be able to cancel echo if I turn training off?
22:27.33watchyi think thats for software echo trainers
22:27.36watchyi could be wrong though
22:28.01watchythis res_odbc is making me want to shoot myself, so i'm not thinking very clearly
22:28.08watchyand i'm far too lazy to look it up
22:28.11rue_mohrechotraining=yes was all that was set
22:28.26watchythen put no?
22:28.34Qwelljust remove it
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22:28.43rue_mohrI commented it out
22:28.46watchyits not asking for the capital of california
22:28.50watchyits asking for a yes or not
22:28.51rue_mohrhow do I know if someone is on the phone?
22:28.54watchyor just delete it
22:29.03watchysip show inuse
22:29.09Qwellif calls aren't working...  chances are, nobody is on the phone.
22:29.14watchyhaha, yea
22:29.47rue_mohrthere is just a HUGE echo on the sip phone side
22:30.01watchyyou have echo between 2 sip phones?
22:30.10rue_mohrdahdi->* ->sip
22:30.23watchywhat kinda card?
22:30.42watchyit better not be some generic poor people shit
22:30.56watchyyou didnt buy it off ebay for $4 did you
22:30.59rue_mohrok, commenteing out echo training worked
22:31.11rue_mohrwhy? I had it set to yes before
22:31.19QwellBecause your config was wrong.
22:31.27rue_mohrbut it was working before
22:31.32QwellYour config was wrong.
22:31.38watchywell apparently your config was wrong
22:31.48Qwellit working was a bug.
22:31.58rue_mohroh wait, there is one difference now, my libtonezone matches the dahdi drivers, it didn't before
22:32.00watchyi'm sure qwell doesn't care dude. he told you how to fix it
22:32.09Qwellactually, you told him how to fix it
22:32.15watchywell
22:32.19rue_mohrok, let me check some things and I'll go over the database with ya
22:32.20Qwellbut let's not argue semantics.  I'll take the credit if you really want :P
22:32.23watchyfine, i don't care why it worked before
22:32.27watchyit works now
22:32.44watchymy res stuff is setup fine from every aspect i can tell
22:32.50watchythe isql connect works etc
22:33.10watchybut in asterisk i see nothing at all really to with res_odbc having issues
22:33.45cvnetanyone here who would help me out sending a h323 call to their box for testing?
22:33.48watchyi really should of set this up before
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22:46.01rue_mohrok so, what I idieally need is for asterisk to generate dtmf out the dahdi card for the duration the user holds their finger ont eh button, I'm using rfc2833 for dtmf on both asterisk and the polycom phones
22:46.12elitecoderAnone know how to fix this? [Jun  4 13:42:07] WARNING[23500]: app_queue.c:3014 try_calling: The device state of this queue member, SIP/britani, is still 'Not in Use' when it probably should not be!
22:50.26rue_mohrshoudl I have a /dev/dahdi interface?
22:51.06*** join/#asterisk buttons840 (n=buttons8@c-98-202-249-126.hsd1.ut.comcast.net)
22:51.47rue_mohrfxotune dosn't work anymore
22:51.52rue_mohrwhat did I do!
22:52.40buttons840asterisk is asking if my card is single or duel span, what is the difference?
22:52.55rue_mohr1 or 2 port
22:53.26buttons8404 jacks (4 pin jacks, what are they called?) on the card
22:53.37rue_mohrtdm400
22:53.55buttons840so what will i answer to the question, single duel or logging?
22:54.06rue_mohrwhat arey ou trying t set
22:54.21buttons840i'm just doing the asterisk installation, and it is asking these questions
22:54.29buttons840the card is a sangoma a200
22:54.43rue_mohrI dontknow
22:58.34QwellAsterisk doesn't ask questions.
22:58.37rue_mohram I in danger of having to change config files to work again if I take asterisk from 1.4.22 to 1.2.24
22:58.38Qwellbuttons840: What are you really doing?
22:58.48rue_mohrer
22:58.50rue_mohram I in danger of having to change config files to work again if I take asterisk from 1.4.22 to 1.4.24
22:59.28ectospasmrue_mohr: the configuration files do not change from 1.4.22 to 1.4.24, so the changes should be minimal if any
22:59.38rue_mohrthanks
23:00.27rue_mohrwhats the lasest stable? 1.4.24 or .25?
23:00.35ectospasm1.4.22 was the first to use DAHDI, but if you're already at that version it shouldn't make a difference
23:00.40cvnetanyone here who would help me setup a IPsec with virizon? I'm willing to pay you for your help please msg me
23:00.45ectospasmrue_mohr: you should check the Changlog for that info
23:00.50Qwellrue_mohr: What does the topic say?  What does the download site say?  What did lmadsen say less than an hour ago?
23:01.03watchyundefined symbol: mysql_odbc_escape_string
23:01.04rue_mohroh it might eb relivent
23:01.10watchyi'm getting that when i start asterisk now
23:01.15watchyso i guess i'm getting somewhere
23:01.17rue_mohrstops eating his pen for a sec
23:01.58watchyi'm further in my odbc issues
23:03.01rue_mohrsee .22 was the first to use dahdi, so I'm thinking there are prolly fixes in the subsiquent versions
23:03.59ectospasmrue_mohr: right, but the configuration files have not changed syntax
23:04.06ectospasmafaik, anyway
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23:04.41rue_mohrgood stuff, know off hand if the dahdi-linux-complete-2.1.0.4+2.1.0.2.tar.gz has the fix to dahdi_monitor to make the guage lengths come out right?
23:05.41ectospasmrue_mohr: what do you mean?
23:05.45watchyisnt dahdi monitor in tools?
23:05.51ectospasmwatchy: yes
23:06.15watchydoes dahdi linux complete include tools?
23:06.22ectospasmwatchy: yes
23:06.32watchyah ok
23:06.59ectospasm(the 2.1.0.2 is the tools version)
23:08.08buttons840Qwell, I received a asterisk server from a friend, I formated the drive to start from scratch, it has a sangoma a200, i installed linux (ubuntu server), and asterisk, during the asterisk installation the terminal reported 1 isdn card found, and asked some questions about it.  I know little about the card, I just want to practice the scripting programming elements of asterisk.
23:14.41rue_mohryea I rebuilt it and it says it cant find a /dev/ entry
23:15.27[TK]D-Fenderbuttons840: Sangoma cards require to to install their wanpipe driver first <-
23:15.29ectospasmrue_mohr: what says it can't find the device?  Are there any errors in dmesg
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23:15.46elitecoderI'm trying to prevent asterisk from trying to send a call to queue members who're already on the phone. Any help?
23:16.09rue_mohrroot@debian:/usr/src# fxotune -v -i 6   /dev/dahdi/1 absent: Device or resource busy
23:16.16rue_mohrit worked beofre...
23:16.23carrarlitecoder, if only there was a way to get the status of phone!
23:16.33ectospasmrue_mohr: is Asterisk running?
23:16.37rue_mohroh wait
23:16.38rue_mohrhahaha
23:16.43rue_mohrthanks
23:17.26buttons840hmmm, where can i find this driver?
23:17.37rue_mohrthe complete?
23:17.42rue_mohror the isdn
23:17.52*** join/#asterisk fnordus (n=dnall@70.71.225.48)
23:17.54buttons840what do i need to get this a200 working?
23:17.56[TK]D-Fenderelitecoder: pastebin the complete CLI output around it, not just the error
23:17.56rue_mohrthe isdn support will be in the linux kernel
23:18.06[TK]D-Fenderbuttons840: WANPIPE <--------------------------
23:18.37carrarSMOKIN WAN
23:18.46*** join/#asterisk fnordus (n=dnall@70.71.225.48)
23:19.04ectospasmheh
23:19.53carrarWANHOOKAH
23:21.01ectospasmcarrar: WANVAPORIZER?
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23:22.01rue_mohris there info on what the paramiters are in fxotune.conf?
23:23.12ectospasmrue_mohr: it's a series of coefficients for an equation
23:23.50ectospasmunless you intimately understand FFTs and such, you shouldn't be setting those values by hand
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23:24.05[TK]D-Fenderectospasm: If they are so efficient, why do their need to be more than one? ;)
23:24.19[TK]D-Fenderthere*
23:24.20ectospasm[TK]D-Fender: uh
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23:25.09ectospasmI don't even know where to begin answering you (yes, I know you're being facetious (aren't you?))
23:26.08[TK]D-Fenderectospasm: MOI?! Never!
23:26.08elitecoder[TK]D-Fender: there's no error, it's just sending a call to a member who's on the phone already.
23:26.38rue_mohrectospasm, I do udnerstand fft's and thats as much as I want know, thanks!
23:26.41*** join/#asterisk fnordus (n=dnall@70.71.225.48)
23:26.58rue_mohrfft...
23:27.00rue_mohrshudders
23:28.04elitecoderok I have it figured out. in sip.conf you need to add limitonpeer=yes under [general], and for the peers/users set the call limit call-limit
23:29.02ectospasmelitecoder: call-limit limits the number of simultaneous calls a given user can make, so you may want to set it greater than 1
23:29.05elitecoderhmm if i set call-limit to 1, it will do it
23:29.08[TK]D-Fenderelitecoder: Yes, because I know that warning also comes with a specific warning about what doc to read to fix that...
23:29.23elitecoderi already read upgrade.txt
23:29.34[TK]D-Fenderelitecoder: Read it again... fior the first time!
23:30.05elitecoderlooks like it may be working now
23:30.15elitecoderit did call a busy phone once but it hasn't again
23:30.42cvnet[TK]D-Fender: after sending calls to Virizon and didnt go through, i have contact them and they send me this back "Your calls are showing multiple timers that are incompatible. (For example packetization timer of 150 ms on g711Ulaw64k and g711Alaw64k calls)" any suggestions? You are sending in incompatible packetization timing on your OLC (open logical channel)'s codec list" i cant make any sense out of it, any suggestions?
23:31.09elitecoderthat should be fine though, cause i'm testing it on the same phone
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23:41.32Qwellcvnet: just setup a VPN, man...  you'll be much happier, I promise.
23:41.37*** join/#asterisk bbryant (n=brett@c-68-59-21-152.hsd1.sc.comcast.net)
23:42.32cvnethum I'll try
23:42.57*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
23:43.58Qwellor...just switch providers and be up in about 10 minutes
23:44.24cvnetvirizon has the best rates, thats why im trying to get this up and running
23:44.33QwellNo they don't.
23:44.41cvnetwho then?
23:44.55QwellI absolutely guarantee that you can find 20 providers that beat their rate by just typing "voip providers" into google.
23:45.03Qwell~itsplist-us
23:45.04infobot[~itsplist-us]  Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
23:45.35cvneti have cheaper poviders, but not very stable, they charge you for FAS and 60% of time no connection, again this is not for North America route, but over seas
23:47.25Qwellbesides - even if a provider is more expensive...  how many hours have you spent trying to get Verizon working?  How much is your time worth?
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23:49.28cvnetlol, this is my goal, I work from home so time i got alot, as long as i get the right guild to do it, i have been searching the net for 4 days trying to find how to do ipsec but so far nothing very usefull, did lots of reading... still trying , im new in voip and asterisk so it takes me 6 days to do the same taks as probably would take you 20 min
23:49.57Qwellthat's because you're using h323.
23:50.52cvnetwell with h323 i dont need ipsec, but it doesnt work, they send me an email saying pocket size are incorrect, so im focusing on sip with ipsec
23:51.45cvnet"Your calls are showing multiple timers that are incompatible" <-- for h323
23:52.17Qwelllike I said...
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23:54.04cvnetthe requiement for SIP are +Equipment must support multiple VPN tunnels , +Equipment must support ONE of these transform sets: esp-3des-md5, esp-3des-sha1, esp-aes128-md5, esp-aes128-sha1, +Perfect Forward Secrecy <== so you sayingi can do all that with VPN ?
23:55.21[TK]D-Fender....
23:55.31[TK]D-Fendercvnet: THOSE ARE VPN SETTINGS <-
23:55.48[TK]D-Fendercvnet: You are now formally invited to join "The Real World" (tm)
23:56.24buttons840i've open nautilus using gksudo nautilus, i then enter the fallowing in the location "ssh://ester@192.168.0.100/" but this returns an error "could not find the loations /home/buttons/ssh://ester@192.168.0.100/" also note, this process works perfectly when i use regular nautilus, but not when i open it as super user
23:56.48Qwellbuttons840: why do we care/?
23:56.55buttons840sry, wrong chat
23:57.07Qwellyou know...
23:57.18QwellI don't want Linux to be a good desktop.
23:57.27QwellI just decided that.  Just now.
23:58.02*** part/#asterisk buttons840 (n=buttons8@c-98-202-249-126.hsd1.ut.comcast.net)
23:58.27Qwelland the reason just left.
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