IRC log for #asterisk on 20080314

00:00.48EmleyMoordraygon: I do
00:07.21EmleyMoorAt work, someone referred to using Asterisk and Festival togeether as "Festerisk" the other day
00:07.25*** join/#asterisk nighty^ (n=nighty@210.188.173.246)
00:07.41EmleyMoorThis keyboard seems to be doing odd things
00:10.11*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
00:10.23EmleyMoor"Festerisk" is just the kind of hybrid word my dad would have liked!
00:12.54*** join/#asterisk xenonex (n=xenonex@89.218.237.221)
00:15.46*** join/#asterisk m4sk4r4 (n=m4sk4r4@189.12.64.27)
00:16.14EmleyMoorI actually use Festival for all my non-personal "public-side" announcements, and sometimes for internal ones if I need to set them up quickly with no time to record my own.
00:18.12TJNIII use festival in AGIs because simply typing in a string I want in the prompt is easier than pre-recording and mapping a bunch of sound files.
00:18.35TJNIIPlus it works for my 411 script, which pulls from a SQL database.
00:20.07EmleyMoorMy "known marketing number" trap is a bit of a masterpiece, I think
00:20.12*** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk)
00:21.20EmleyMoorFirst, they get instructed by Festival, then by Jay Benham then they end up leaving Voicemail (if they bother!)
00:21.54TJNIIHahaha.  I do have a "queue" in one script that consists of for($count = 0; $count < 120; $count += 1) {do_command("EXEC playback \"custom/tech_music\"");}
00:22.11TJNIIWhere tech_music is a 30 second clip of bad hold music.
00:22.45TJNII(It also plays some sound files telling you about "priority service," but I didn't want to flood the channel
00:23.23MatBoyI think the airplane with my 2x 410 card crashed or something :S
00:23.32MatBoy*cards
00:23.41jblackDid they also assure you that you're call was important to them while dumping you off on bad music?
00:24.31TJNIIOf course.
00:24.33EmleyMoorI am tempted to add "The importance of your call is yet to be determined."
00:24.48*** join/#asterisk Katty (n=The@adsl-75-59-138-34.dsl.stlsmo.sbcglobal.net)
00:24.58jblackEmleyMoor: Nice!.
00:25.30Kattyallo.
00:25.47EmleyMoorSince I got Asterisk configured correctly I have not had one unwanted phone call wake me up, apart from the necessary evil of work calls.
00:26.07jblackset up a hold that makes them randomly perform simple addition, hanging up if they either don't answer, or answer incorrectly.
00:26.14riddleboxdoes anyone have a Grandstream HT488? I set it to periodically subscribe to MWI, but after like an hour the phone doesnt work, I have to unplug it then plug it back in? If I set MWI to not subscribe the phone works perfectly?
00:26.15*** join/#asterisk cardiff (n=cardiff@76-10-153-160.dsl.teksavvy.com)
00:26.49jblack"Please enter the sum of 3+4 to continue holding."
00:27.31billytwowillyanyone familiar with the asterisk appliance here?
00:28.11TJNIII thought about making the script with the "queue" call the user back if they hung up either in the "queue" or the string of pointless IVR menus beforehand, but I thought better of it.
00:28.50jblackWhy did you decide against it?
00:28.50TJNII"I'm sorry we were disconnected.  We care about your call.  Let me place you where you left off."
00:29.07TJNIIBecause people might come for my head. :)
00:29.09billytwowillyHow hard is it to configure the appliance to auto route numbers to specific phones on the lan? i.e can I have 15 people with 15 phones, get 15 dids and then associate each did with a phone on the net so the asterisk appliance passes calls through automagically?
00:29.22jblackOh, you don't mean as a feature... You mean.... call hold stalking.
00:29.23jblackVicious! I love it!
00:29.34TJNIIYea
00:29.45jblack"I told you your call was important to me."
00:29.53*** join/#asterisk voiceperu (n=al@190.42.38.201)
00:30.03voiceperuhellooo
00:30.14TJNIIIt's not really a feature when the "queue" is just an hour of sounds.
00:30.51jblacktell me the sounds aren't that "the computer is thinking really hard" boo-de-beeps.
00:31.50TJNIIThis is all in my "tech support" script.  It looks at the calling context, and reacts based off that.  If it is from a phone I maintain, it drops you into a real queue.  If it is from outside, if messes with you for at least an hour and a half.
00:32.13*** join/#asterisk kimosabe (n=nat@adsl-69-155-128-143.dsl.hstntx.swbell.net)
00:32.21voiceperucan anybody test my asterisk server?
00:33.49kimosabecan i recieve pri on my ds3 to my cisco router while recieving internet bandwith and then from my cisco send the pri to asterisk box
00:34.53*** join/#asterisk St1ckm4n (i=St1ckm4n@75.145.72.133)
00:37.41St1ckm4ndoes anyone here have much experience with asterisk in a call center enviroment?
00:38.49voiceperucan anybody type 190.43.129.89
00:38.56voiceperuin a browser
00:40.25jblackI'm thinking about using the "youuuuu, you got what I neeeed, but you say he just a friend" song in loop as my hold music
00:40.51voiceperusorry
00:41.03voiceperutype 190.42.38.201
00:41.24TJNIIThat is a good song.
00:41.36TJNIII have American Pie
00:41.44TJNIIAnd Safety Dance
00:42.27TJNIIBut when I really feal cruel, I use this: http://www.amazon.com/Beatles-Bossa-Brazilian-Tropical-Orchestra/dp/B00000G7GF
00:42.33jblackHow about I'm too sexy?
00:42.36*** join/#asterisk Idle (n=brian@S010600a024969312.ed.shawcable.net)
00:42.45Idlewhat variable is used for the timestamp?
00:42.56TJNIIAnd I use the clips I downloaded off Amazon, so you never get more than 35 seconds of a song.
00:43.00jblackPerhaps everything at http://www.blender.com/guide/articles.aspx?id=786
00:43.38TJNIIGod that list is bad.
00:44.54jblackOhhhhh. ice ice baby
00:50.37TJNIII have the entire bible in WAV files, but in never made its way into the moh directory.
00:51.41jblackIf I were rich, I'd hire allison to read /usr/share/dict/words.
00:52.35rkeeneI've got a design question -- I'm going to be migrating from a legacy system and I want to change as little as possible.  I'm basically only migrating *PART* of the legacy system (the part under my control), so users are used to dialing 8-XXXX to get someone else on base, and I want to preserve this functionality (going out to the PSTN if it's not a SIP phone)... Is there any way to do this without enumerating all of my extensions in the dialplan ?
00:52.40*** join/#asterisk m4sk4r4 (n=m4sk4r4@189.12.64.27)
00:53.04jblackI can't believe walk like an egyptian didn't make it onto the list
00:53.30TJNIIrkeene: It can be done with a pattern match and a goto
00:53.44*** part/#asterisk cardiff (n=cardiff@76-10-153-160.dsl.teksavvy.com)
00:54.17*** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net)
00:54.18rkeeneTJNII, How so ?
00:54.36rkeeneAll the pattern matching I've read about was sequential, numerical, or string based
00:55.16ThatKidKelcdr_pgsql.so question..  is the spool=pgsql.spool only available after applying the patch?  what is recommended for catching cdrs when the database is unavailable??
00:55.36jblackOH YES!
00:55.38jblackhttp://youtube.com/watch?v=p-At6wk_fQs
00:55.43TJNIIPut your extensions in one context.  Don't give users access it. Then something like exten => _8XXXX,1,Goto(hidcontext,${EXTEN:1},1)
00:56.00*** part/#asterisk mog (n=mog@c-68-62-172-83.hsd1.al.comcast.net)
00:56.34*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7addb4c2794e0f4d)
00:57.17TJNIIjblack: When video phones become standard.....
00:57.26*** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net)
00:57.54rkeeneTJNII, "Put my extensions in one context" ?
00:58.18voiceperuwhere i can find this in my debian ?
00:58.21voiceperuYou don't have permission to access /html/admin/modules/recordings/popup.php on this server.
00:58.34TJNIII think I may have misunderstood you, what do you mean by "not enumerating all my extensions"
00:58.36rkeene(Enumerate them in a context, or specify that they belong to a particular context when defining them)
00:58.37voiceperui dont see html folder
00:59.47TJNIIOh, you want _8XXXX to transfer out of your system?
01:00.20rkeeneTJNII, i.e., I don't want to have something like:  exten => 81001,1,Macro(blah)     exten => 81003,1,Macro(blah)   in my dial plan to have 1001, and 1003 be local and 1002 be external.
01:00.52rkeeneI guess I could, but I already have the list of extensions in so many places :-P
01:01.20TJNIIOh, yea.  So do you want 81003 and 1003 to work?
01:01.34TJNIIOr just 81003?
01:01.58rkeeneJust 8XXXX
01:02.25rkeeneBut 81002 is external (via PSTN trunk), while 81001 is internal
01:04.08*** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au)
01:04.13*** join/#asterisk zen-froglet (n=zen@starlight.chat.za.net)
01:04.33zen-frogletmorning.
01:04.43TJNIIYou'll need to sort the 4 digit extensions so they, and only they, are accessable through one context, if you haven't already.  Don't include this context in the context of your phones.  Then, put that pattern match in the context of the phones and point it at the "hidden" context.
01:04.57zen-frogletwhat would be the recommended app to use for receiving Faxes with Asterisk ?
01:05.12TJNIISo it will match any 5 digit extension starting with 8, and then look in your already defined extensions.
01:06.34rkeeneSo basically I will have to list out all of my extensions in the dial plan, correct ?
01:07.22TJNIIWell, yes and no.
01:07.38rkeene(I have 300 extensions)
01:07.41TJNIIYou will need to seperate the 4 digit extensions from the 5 digit extensions
01:07.54rkeeneThey will all be dialed as 5 digit extensions
01:07.59TJNIIRight.
01:08.29TJNIIBut unless you want to support 4 digit extensions, you must seperate them from any 5 digit extensions in the dialplan contexts.
01:08.48TJNIII guess, how do you have it set up now?  Do all 300 extensions work?
01:09.04rkeeneAnd to do that I will need to put the list of all valid extensions in the dial plan ?
01:09.13*** join/#asterisk RoyK (n=roy@ip-77-55-149-91.dialup.ice.no)
01:09.35rkeeneThere are over 300 extensions under 8XXXX right now, I want a subset of them (300) to resolve to SIP accounts, and the rest (around 4500) to resolve to PSTN terminations
01:10.10rkeeneSo if I dial 81002 from my SIP phone, I get the PSTN 81002, but if I dial 81001 I get the SIP 81001
01:11.05voiceperuhellooo , why i cant play my recordings in freepbx?
01:11.09voiceperui get this
01:11.10TJNIIIs there a rhyme or reason to which exten goes where?
01:11.13voiceperuYou don't have permission to access /html/admin/modules/recordings/popup.php on this server
01:11.21TJNIIIf not, how were you planning on not enumerating each one?
01:11.26rkeeneNo, they are a random subset
01:11.31TJNIIvoiceperu: #freepbx
01:11.41rkeeneI've already enumerated them in the SIP configuration, I was hoping I could reference that list somehow
01:12.01TJNIIyea, you've done it as 4 digit extensions correct?
01:12.19TJNIIAll in a context?
01:12.53rkeeneNo, in the SIP configuration they are the 5 digit (8XXXX) values.. but merely for convience and I could change them to the 4 if that would be easier
01:13.03rkeeneYes, all in the same context, even
01:13.25*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)
01:13.27*** join/#asterisk stansmith (n=stansmit@d60-65-93-136.col.wideopenwest.com)
01:13.35*** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org)
01:13.54TJNIIOkay, so you've done it in the SIP configs, but you haven't done the dialplan yet?
01:14.15rkeeneRight
01:14.33rkeene(Well, the SIP config was generated from a script, pulling the information from LDAP)
01:14.39*** part/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net)
01:14.40TJNIISo your dialplan is currently blank.
01:15.37rkeeneNo, it just doesn't allow for dialing 8XXXX when you want the legacy system's 8XXXX
01:15.53TJNIIOkay, so you have a dialplan with 4 digit extensions defined
01:15.54rkeeneI can dial 81234 and get SIP phone 81234
01:16.02TJNIIOkay
01:16.05rkeeneNo, it uses the 5 digit values
01:16.20TJNIIAll right, I think I'm getting it.
01:16.49TJNIIYou have extensions for all the sip phones, but not the extensions the go the the PSTN.
01:17.13rkeeneRight (since I don't control the PSTN)
01:17.28rkeeneMy interface to the PSTN is a PRI card
01:18.20TJNIIOkay, so for simplicity put all these in a context, like [siplocal]
01:18.36TJNIIThen define another, say [default]
01:18.46TJNIIinclude => siplocal and then do
01:19.05TJNIIexten => _8XXXX,1,Dial(however you get to the PSTN)
01:19.30TJNIIIt will match your local extens first, and the pattern match will grab the rest and send it to the PSTN.
01:19.44*** join/#asterisk BobLutz (n=stansmit@d60-65-93-136.col.wideopenwest.com)
01:19.48BobLutzHello
01:19.55TJNIIJust make sure asterisk gets to the pattern match AFTER your defined extens.
01:20.09rkeeneBut what will cause me dialing 81001 (where 81001 is a SIP account) to get directed to the SIP account ?
01:20.49TJNIIBecause your defined extensions are first
01:21.14TJNIIso it will run your, already defined, exten => 81001 before ever getting to the pattern match.
01:21.25rkeeneSo in my extensions.conf I would have [siplocal] *with nothing in it* [default] exten => _8XXXX,1,...    ?
01:21.42TJNIIsiplocal will contain all the sip extens you said you already made.
01:21.45rkeene(Since my goal is to avoid listing all of the extensions in the extensions.conf again, since they are already in the sip.conf)
01:22.04rkeeneRight, my goal is to AVOID listing all the extensions in the sip.conf and the extensions.conf
01:22.07TJNIIWell, sip.conf tells the system what phones are connected to it
01:22.34TJNIIsip.conf is not part of the dialplan
01:22.46rkeeneRight, I don't want to enumerate this in my dial pla
01:22.51rkeenedial plan, rather.
01:22.55TJNIIYou're going to have to
01:23.02rkeeneOh, okay
01:23.10TJNIIIf there is no rhyme or reason to which are SIP and which are PSTN
01:23.48rkeeneI just thought there might be some way for the dial plan to examine the list of SIP addresses, for convience
01:24.05TJNIINot that I know of.
01:24.16TJNIIYou could probably make something that does, though.
01:24.31TJNIIWell, are the sip addresses going to change often?
01:24.35BobLutzOh boy
01:24.47rkeeneNo... but I can easily generate the dialplan from a script as I do the sip.conf :-P
01:24.59TJNIIThat would probably be easist.
01:25.26TJNIIMake a script that creates extensions.siplocal.conf and include that file into your extensions.conf
01:26.01rkeeneCan I include files from sip.conf, also ?
01:26.17TJNIII think so.
01:26.55BobLutzWait - You can have an "include" in extensions.conf, but the file that is included can be dynamically created - without having to reload the dialplan?
01:27.23rkeeneBobLutz, I assume you will have to reload the dial plan
01:27.32TJNIIYea, you do
01:27.42BobLutzOh, I misunderstood what you guys were saying  - that would of been extremely powerful though
01:28.21rkeeneBut reloading the dialplan isn't that hard
01:28.22TJNIIThere are dynamic dialplan solutions, I know nothing about them though.
01:28.33rkeene(From a script... that generates the dial plan :-P)
01:28.49TJNIIHeheheh.  There you go.
01:28.56rkeene(asterisk -r -x 'dialplan reload')
01:30.19BobLutzYea...
01:31.32rkeeneThe list comes from an LDAP server, and gets sprinkled into an XML file for the phonebook's directory and the extension list :-P
01:33.07TJNIIYou could potentially roll something with agi, but since it is updating the sip.conf anyways, I don't think it would be worth the effort.
01:33.18TJNIII don't know any means to do a dynamic sip.conf.
01:33.36*** part/#asterisk RoyK (n=roy@ip-77-55-149-91.dialup.ice.no)
01:35.51JackEStormI have my sip.conf and queue stuff (along with cdr) in sql via realtime
01:37.09rkeeneJackEStorm, I would consider it, but from what I've been told it wouldn't help (without "potentially something with agi")
01:37.09*** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net)
01:38.43TJNIIRealtime may do it, voip-info has an LDAP example
01:40.43TJNIINone the less, you can probably do the scripted dialplan to get it working now, and then figure out if realtime will work better later.
01:40.59*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
01:41.03zen-frogletwhat would be the recommended app to use for receiving Faxes with Asterisk ?
01:41.14TJNIII haven't played with LDAP in a couple years, but when I did I thought it was kind of an uphill battle.
01:41.16zen-frogletother than rxfax and spandsp ?
01:41.32kimosabeif not mistaken asterisk cant do faxes yet
01:41.36rkeeneYeah, the scripted dial plan is trivial
01:41.57*** join/#asterisk efort (n=efort@74-86-100-202.lx-vs.net)
01:42.14TJNIIWell, maybe it won't do LDAP.  I don't know.
01:42.25zen-frogletkimosabe: my previous Asterisk box would accept faxes and email to me
01:42.50zen-frogletperhaps not a specific function of Asterisk, but it ran within my system
01:43.43*** join/#asterisk AndyGraybeal (n=AndyGray@128.177.27.78) [NETSPLIT VICTIM]
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01:54.51*** mode/#asterisk [+o Corydon76-dig] by ChanServ
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01:57.40Alpha_AIHello there
01:58.07BobLutzAlpha_AI, Hey!
01:59.10Alpha_AIhey boblutz, how r ya?
01:59.32BobLutzIm thinking if i should modify app_swift.c or not
01:59.40BobLutzAlpha_AI, you know a lot of C?
01:59.42Alpha_AIoh yip
01:59.47Alpha_AIno i dont
02:00.00Alpha_AImostly c
02:00.01BobLutzI know  a little..I think enough to do what I need to do, but im scared!
02:00.02Alpha_AIoops
02:00.04Alpha_AImostly delphi
02:00.15BobLutzI was reading something last night, Delphi is real fast, no?
02:00.23Alpha_AIit is pretty fast
02:00.31Alpha_AIfast enough for me
02:00.39BobLutzFaster than C (gcc) according to that page I read
02:00.48Alpha_AIoh yip
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02:02.36Alpha_AIBobLutz, do you know much about sip server?
02:02.49BobLutz:-/ , I only use sip for testing
02:02.53BobLutzWhat are you trying to do?
02:03.00*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-1ef75968c4ff3bf7)
02:03.34Alpha_AIwell i have to pay $635 a month if i get 128 simultaneous channels
02:03.59BobLutzfor sip?
02:04.05Alpha_AIwhich i believe i need. So instead of paying someone to give me channels i thought i will try and give myself free channels
02:04.07Alpha_AIyeah for sip
02:04.18drmessanoFree channels?
02:04.53BobLutzAlpha_AI, you can give yourself unlimited free channels (depends on your hardware) ... getting that out over the internet is something else though
02:05.13drmessanoNo..
02:05.19Alpha_AIno according to the sip providers, they are the ones that provide the channels
02:05.27*** join/#asterisk youngproguru (n=root@cpe-76-180-239-199.buffalo.res.rr.com)
02:05.27drmessanoGetting them terminated with an ITSP is something else
02:05.37BobLutzIs what I meant ^
02:05.44drmessanoYou can have unlimited free channels on the internet
02:06.35Alpha_AIlets say i got a did number, to get more people to call that one did number i will need more channels. they cost $5 per month with the group im looking at
02:07.09drmessanook...
02:07.16drmessanoSo how do you propose to do that for free?
02:09.50Alpha_AI.
02:10.10drmessano?
02:10.15Alpha_AIi dont know
02:10.22drmessanoSounds like a plan
02:10.27Alpha_AIim looking at a free open source sip server at the moment
02:10.28drmessanoLet me know how it turns out
02:10.35drmessanoOk
02:10.37drmessanoThats great
02:10.42drmessanoYou have SIP
02:10.45drmessanoThen what
02:11.01TJNIIStep 1: Require hundreds of simultaneous PSTN lines.  Step 2: ??? Step 3: Profit!
02:11.10drmessanoFTW
02:11.12TJNIIStep 2 is always a bitch.
02:11.17BobLutzLOL
02:11.43drmessano~freelines
02:11.44billytwowillystep 2 is hire phone sex operators to man the hundreds of simultaneous PSTN lines
02:12.01BobLutz:-o
02:12.05*** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net)
02:12.08drmessano~freelines
02:12.09jbotStep 1: Require hundreds of simultaneous PSTN lines.  Step 2: ??? Step 3: Profit!
02:12.09TJNIII like the cut of your jib.
02:12.52drmessano~freelines
02:12.54drmessano~free lines
02:12.55jbotStep 1: Require hundreds of simultaneous PSTN lines.  Step 2: ??? Step 3: Profit!
02:12.56drmessanobetter
02:12.59drmessanoneeded the space
02:13.52drmessano"ATT wants $500 a month for a T1"  "I am trying to work out how to do it for free"  "?????"
02:14.13BobLutzdrmessano, Hey, come on, we dont want no black-on-black in here
02:14.29drmessanoblack on black?
02:14.39BobLutzum..
02:15.22*** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id)
02:15.24billytwowillyBobLutz only goes for the inter-racial shenanigans..
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02:17.01drmessanoI see
02:17.13BobLutzast_exists_extension() is deprecated?
02:17.34drmessanoWhere did you see that?
02:17.41BobLutzhttp://lists.digium.com/pipermail/asterisk-dev/2008-January/031466.html
02:18.10ZPerteehow do I find an asterisk consultant in my area?
02:19.05droopswhere is your area?
02:19.27plikhaha, voip manufacturer Snom moved offices last week and their phone lines still aren't connected!
02:19.41BobLutz0wn3d?
02:19.50drmessano0wn3d?
02:19.54ZPerteedroops: North East Ohio
02:20.07drmessanopwn3d <-- the non-lame spelling
02:20.11pliknsh, just a typical teecom cock-up
02:20.47droopsZPertee, http://www.voip-info.org/wiki/view/Asterisk+consultants+USA#OHIO
02:20.58SteveTotarothey should have prepared a little better for a move
02:21.23SteveTotarorent the telco closet for another month and send the calls via sip
02:22.02SteveTotaroor just forward the calls, but don't move until the circuits are in place to forward to
02:22.45SteveTotarosnom is nice but too european looking in style for my tastes
02:23.00plikyeah, we all know how it should be done, or could be resolved - you'd think theyd have a clue too :)
02:23.47SteveTotarowell they are a phone maker, so it doesn't really surprise me
02:24.07SteveTotarothey don't deal with the telcos all the time like some of us
02:24.59drmessanoWhere is Snom located?
02:25.10SteveTotarogermany i believe
02:25.10plikgermany somewhere
02:25.11TJNIIBerlin, I believe
02:25.26SteveTotarotear that wall down
02:25.40BobLutzha
02:25.44droopsi think they did
02:25.49SteveTotarohow are you doin DRM?
02:25.50plikberlin, yes... TJNII Wins!!
02:26.02SteveTotaroI was correct too
02:26.11drmessanoAh
02:26.12SteveTotarojust a little broader in terms
02:26.20drmessanoSpreken Ze Deutsch?
02:26.31drmessanoIm good, Steve
02:26.35drmessanoBeen working my new job
02:26.55*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-61864d40278afa84)
02:26.59eforthow can I tell if asterisk is registering with my ITSP?
02:27.04SteveTotarofalling in like a well oiled cog?
02:27.24drmessanoYeah.. and I made twice as much this week as I did last week working for my old company
02:27.25SteveTotarosip show registry
02:27.27drmessanoHow is that? lol
02:27.37SteveTotaroor iax show reg
02:28.28SteveTotaromoney is one thing, but is the fit right do you think?
02:28.39efortSteveTotaro: thanks, it's a sip trunk
02:29.05SteveTotaronp
02:30.15drmessanoI think it is.. Seems to be a good bunch..  They all work like a big family.. and I see to be right in there
02:30.52SteveTotarocan you slip in your clandestine asterisk agenda?
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02:31.15SteveTotarosoon they will all be addicted
02:31.32drmessanoI am going to
02:31.51drmessanoI plan to wait a few weeks and push the issue
02:31.53efortwhat would keep it from registering?  what needs done to get * to register?  it's static ip on both ends.  using Vitelity which I've been happy with so far
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02:33.13DonAlexEvening all :)
02:33.55SteveTotaroso vitelity worked and now it does not?
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02:34.52macaruchiHi!
02:35.20SteveTotarohas anyone used druid?  is it better than the other free GUIs in your opinion?
02:35.24macaruchi<PROTECTED>
02:35.39AsterlinktechyI use vitelity for long time and its works well on m box :)
02:35.47BobLutzdrmessano, you like C?
02:35.59Alpha_AIit seems that IAX can deliver more channels than SIP can
02:36.19macaruchii use install_prereq and everything fine
02:36.33SteveTotaroalpha, i think you are going to run into audio issues with that iax attitude
02:37.13drmessanoBobLutz: C is not my favorite.. I like Thiamin and E is a close second.. Zinc is up there too
02:37.16efortno I have yet to get the trunk configured. they tell me I'm not registering and that's what I see also but I've been happy with them so far because their support has been good and I can generally get someone on the phone if I really need to
02:37.26BobLutzLOL
02:37.33drmessanoIAX can not deliver more channels than SIP
02:37.42SteveTotaroi have used iax.cc/vitelity for years, i just use their forward function to my cell nowdays
02:38.12SteveTotarosip debug
02:38.19SteveTotarosee what is going on
02:38.32SteveTotaroand then pastebin
02:38.36SteveTotaro~pb
02:38.36jbot[~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
02:40.26DonAlexHey peeps... would anyone like to make a guess why when making dialplan rules via AsteriskNow interface my config is just ignoring them? It saves it to extensions.conf and reloads but just ignores anything I put.
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02:41.06efortsip debug shows only "remote unix connection"  then "remote unix connection disconnected"
02:44.21efortI don't think that's related though, I see that all the time like being scanned
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02:46.24BobLutzast_exists_extension() --> Where could I find such a method in doxygen?
02:46.50droopsDonAlex, you might try #asterisknow or #asterisk-gui
02:47.38BobLutzNevermind ---> pbx.c
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02:50.49DonAlexdroops: Well not 100% sure it is related to the gui or just me mucking up the file somehow ;)
02:51.04DonAlexdroops: but I will anyway , thanks
02:51.55Alpha_AIis iax ready for production use yet?
02:52.46drmessanoI use it in production
02:52.50drmessanoSo do others
02:54.42DonAlexAhhh and another thing to pick someones brains about but just what IS the difference between q931 and 931e ?
02:55.29Qwellan e
02:56.22drmessanoq in the front, e on the back
02:56.29drmessanooh
02:56.44drmessanoBe all technical about it, Qwell
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03:06.05znoG_does anyone know how I could possibly lookup the IP address of the originating call? the idea is to only allow outbound calls if the source IP of the IAX/SIP user is within the local subnet.
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03:22.03TJNIIznoG_: wouldn't it be easier to do that on a per-user basis with contexts?
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03:37.54TJNII~free lines
03:37.54jbotStep 1: Require hundreds of simultaneous PSTN lines.  Step 2: ??? Step 3: Profit!
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04:00.34Alpha_AIHello
04:01.24Alpha_AIIm installing Asterix now through VMserver and its asking me to create a new partitition saying this will cause the loss of ALL DATA on this drive
04:01.24Qwellwhat is asterix?
04:01.25Alpha_AIdoes that mean i will lose everything on the drive or just everything in the VM?
04:01.29Alpha_AIasterisk
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04:02.06Alpha_AIi hate it how someone says 'what is asterix?'
04:02.10Alpha_AIya know what im talking about
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04:03.16Qwellif you can't put forth a little effort to spell the name correctly, why would anybody want to put forth effort to help you?
04:03.36lmadsenQwell: but I didn't even ask my question yet
04:03.43Qwelllmadsen: good!
04:03.52lmadsenquestion time! :)
04:03.59Qwellonly if you spell it right :p
04:04.10lmadsenwhat causes Asterisk to write a disposition of ANSWERED when looking in a SIP trace?
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04:13.38russellblmadsen: the other side answering the call ... 200 OK
04:13.45lmadsenrussellb: thx!
04:13.47russellbnp
04:13.55lmadsenwhat should I expect... just 180 Ringing?
04:14.30russellbmaybe :-p
04:14.31drmessanoHmm
04:17.40lmadsenhrmm... I think a reinvite is screwing up my CDRs
04:28.22lmadsencodefreeze: long shot... but... ping?
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04:57.47b11d`hey chaps
04:58.36b11d`.
04:58.41b11d`whoops
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05:26.49TripleX1anyone up ?
05:27.01BBHossdepends
05:27.17TripleX1hehe
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05:39.55TripleX1which v you running ?
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05:50.03illuminiHello, I'm a first time user setting up AsteriskNow (latest). When I call another extension it goes straight to voicemail, what have I done wrong?
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06:33.13illuminiwhere can I download sample Asterisk config setups, they're mentioned on the website but no links provided
06:33.39TripleX1make samples
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06:50.11mattman99illumini - re your voicemail problem, sounds like the phone you are calling is not registered
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07:35.03pascalsGoodmorning.
07:36.00pascalsI have a quadBRI card and can accept incoming calls, but outgoing calls give the following error: chan_zap.c:8800 zt_pri_error: 4 Write to 30 failed: Unknown error 500
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07:45.07awkhmm, any sugestions as I can determine if 30 channels on a PRI are used..
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07:45.13awkanyone have a script ?
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07:45.25pascalsFound my problem
07:46.10pascalszapata.conf listed too many channels available. Asterisk 1.0 didn't mind, 1.2 doesn't seem to handle that the same way
07:47.07pascalsFYI: I had the gsm optimization bug on my Suse 10.3 box with gcc 4.2.1, solved it by setting -O6 to -O2 in codecs/gsm/Makefile.
07:49.07awkI need some script that will tell me when i've used up all available channels on the PRI... i'm passing everything to a quintum on sip trunk 8000 and 7000 so I could use some like, asterisk -rx "show channeks" | grep 8000 and pass that to a placement and 7000 and then get the sum of X and Y... and pass to Z and if Z = 30 then sendmail
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07:52.37rkeeneAnyone have any thoughts on the voicemail password being the same as the SIP password ?
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07:55.17cmantitohmm ... sip show peers shows ~95% of the peers "UNREACHABLE"...can you say upstream transit problems? ;p
07:55.38awkrkeene depends on what type of security or what type of pols are in place...
07:55.50awkmost of my clients share the same...
07:56.07awkcmantito how is the peers connected
07:56.17cmantitoover teh interwebs ;p
07:56.25cmantitofrom various locatiosn
07:56.26awkok and half of them are down?
07:56.29awkerr 95%?
07:56.34cmantitoactually, specifically, all the comcast users are down
07:56.35cmantito*cough*
07:56.41cmantito...and a few Covads users
07:57.06awkwell its simple can you see reg attempts, if not can you ping their ip's or g/w?
07:57.27awkif I was you I would use something like vqmanager and monitor endpoints...
07:57.46cmantitoI can see them re/deregistering, and I'd bet money any second now I'll get an SMS from Zabbix letting me know of transit problems somewhere
07:57.57cmantitoerr, not deregistering
07:58.02cmantitobut, become unregistered ;p
07:58.08awkunreachable :)
07:58.13awkafter a timeout of 2000ms, or something
07:58.19cmantitoyeah, that's the word
07:58.27cmantitosorry, 4 am and I'm brought outta bed for this ;p
07:58.42awkgo back to bed, nothing you can do.. send the NOC a mail and go sleep
07:59.12cmantitohaha, and the Jabber server's AIM transport just died
07:59.23cmantitoand the winner is ... Failur(3)!
08:00.09cmantitowow, 50% packet loss from here. Yayyy..
08:00.14cmantitognight lol
08:01.52awkjust run a mtr on your endpoint
08:02.00awkand when you wake up see what type of packet los you acing
08:02.01awkfacing
08:02.08cmantitothat's what I'm doing
08:02.14cmantitothis datacentre has been nothing but problems.
08:03.17awkheh, :) looks like its time to change..
08:03.31cmantitowe're certainly trying ^_^
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08:03.43awkjust make sure you don't make the same mistake next time!
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08:03.56cmantitoyeah.
08:04.01cmantitothat's for damned sure.
08:04.12tengulre11Oh,yes !@#$#^^
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08:19.22DonAlexAnyone any idea why asterisk would ignore my dial rules. They are these : [ Context 'numberplan-custom-1' created by 'pbx_config' ]
08:19.22DonAlex<PROTECTED>
08:19.22DonAlex<PROTECTED>
08:19.22DonAlex<PROTECTED>
08:19.22DonAlex<PROTECTED>
08:19.56DonAlexJust ignores them completely and thinks any number dialled is an extension?
08:21.20DonAlexIt is very frustrating. There is only one trunk set up and it is pointing to the right zap interface.
08:27.20cmantitoand I'm going to bed. Happy piday guys.
08:28.30DonAlexOk moving on then..
08:28.55DonAlexanyone can fill in the details of the difference between q931 and q931e ?
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08:43.41FabiOnegood morning
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09:20.05Chris-NBhi
09:20.21Chris-NBanyone tried to setup a mail to fax gateway with asterisk?
09:20.56Chris-NBreceive an mail, convert to fax/tiff and send a fax via ISDN
09:23.18FabiOnei've a problem with outgoing call through a HFC-S PCI ISDN card
09:23.29FabiOnei receive and make call
09:23.57FabiOnebut only 1 at a time
09:24.14rkeeneawk, The SIP clients are restricted to a private subnet
09:25.33FabiOnei think it's a conf problem in zapata.conf or dialplan
09:25.57FabiOnei use dial(zap/1/${EXTEN})
09:26.12FabiOnei think the "1" is wrong..
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09:36.25eric2Chris-NB   I've tried.. but have yet to succeed
09:36.44Chris-NBeric2, so no stable/reliable solution?
09:36.53eric2not with 711
09:37.10eric2any network jittering and the fax get's messed up
09:37.29eric2people claim to be able to do it on an internal LAN but over the internet is unreliable
09:37.42eric2best way is with t.38 from what I've read
09:37.53Chris-NBI try to send a fax via ISDN, not SIP/IAX/.....
09:38.22Chris-NBwith tx_fax, hylafax, spandsp, asterfax ... or whatever
09:38.30eric2I'm all SIP
09:38.35eric2look at callweaver
09:38.47eric2another software piece that might help
09:39.38Chris-NBokay, thanks! I'll try
09:39.42Chris-NBor look
09:40.40eric2If you do get som'n going, let me know
09:41.01eric2but then again, I don't have any ISDN stuff here
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09:54.55rofferHello, i have a question, i run Asterisk 1.4.4 and got a problem vith making a connection out with my sip provider. i can make internal calls and answer incoming calls from my sip provider but not make any calls out. i get this error. chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"unknown" anyone that can help me with this problem ?
09:55.15*** join/#asterisk Akke (n=andy@78-21-74-48.access.telenet.be)
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10:15.38rofferHello, i have a question, i run Asterisk 1.4.4 and got a problem vith making a connection out with my sip provider. i can make internal calls and answer incoming calls from my sip provider but not make any calls out. i get this error. chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"unknown" anyone that can help me with this problem ?
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10:43.02DarKnesS_WolFanyone having problems for snom behind NAT ?
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11:03.19Tilihas anyone used asterisk in hong kong?
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11:09.31kodomohi folks
11:09.40*** join/#asterisk duckz (n=duckz@81-180-102-217.etth.opensys.ro)
11:10.18kodomodid he update to the newest zaptel/asterisk versions and now I'm getting loads of lost interrupts, when I load ztdummy
11:10.53kodomoany pointers appreciated (currently heavy googling ;) )
11:16.10*** join/#asterisk Worf (n=worf@84.119.67.68)
11:17.56nixguyif i wanna have som statistics for calls from asterisk
11:18.16nixguyanyone have any tips on how to get me started?, the google search words are to wide to give any good results..
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11:20.27Rienzillawooyah this channel is large :)
11:21.34Worfnixguy: hmm ... /var/log/asterisk/cdr-csv/ contains a spreadsheet with your calls ... i guess you have to parse that ...
11:21.56nixguyWorf: k thnx
11:22.43JTfinally
11:22.50JTafter 3 months, this pri is lit
11:22.51JT:)
11:23.46nixguyany recomendations?
11:23.55nixguywhat are you guys using to parse those logsfiles? :)
11:24.42Worfhehe - i could dump my thoughs here now, but i'm a asterisk noob. there should be some solutions allready :)
11:24.59RienzillaHey everyone, I have a question which I cannot find on google. I have a frontoffice here with 4 people who are responsible for both answering phonecalls and answering real-life questions at the desk. Therefore I am looking for a queueing solution for asterisk which will not immediately connect callers to an agent's handset, but rather have the handset ring for people that are present (So one of the employees can pick the call up). I do need other fea
11:27.40DrAk0Rienzilla, what?
11:27.53WorfRienzilla: from what i found out you can log in with the same account multiple times, and all phones will ring simultaneousely then. and the first one who picks up, wins...
11:29.19Rienzillabut if I log in, i'm permanently off hook, no?
11:29.38Rienzilla(I found a callbacklogin thing, but that is deprecated according to documentation)
11:30.29DrAk0Rienzilla, i don't get your point
11:30.41DrAk0Rienzilla, you don't want the call to ring on all desks?
11:30.45Rienzillano
11:30.54DrAk0Rienzilla, in which one you want it to ring?
11:30.56RienzillaI only want the call to ring on the desks of users that are present
11:31.05DrAk0Rienzilla, ok
11:31.07DrAk0thats easy
11:31.35Rienzilla(I have snom 360 sip phones and asterisk 1.4.9 pbx, if that helps)
11:31.50DrAk0Rienzilla, agentcallbacklogin
11:32.24RienzillaDrAk0: yes I saw that on voip-info but it is deprecated according to the docs there
11:32.36DrAk0Rienzilla, i use it
11:32.40Rienzillaok
11:32.40DrAk0on asterisk 1.4.x
11:32.46DrAk0and works pretty good
11:32.51Rienzillaok good
11:32.56RienzillaI'll try that then
11:33.04Worftime for my noob question ... i set up asterisk on my router but i think i messed something up badly, because when dialing out nobody can hear me. it works when being called and it works when configuring the phone to not log in on my own asterisk but on sipgate directly ... any hints where to start looking? firewall issue? misconfigured asterisk? how do i actually track down such a problem?
11:33.33Rienzillaand one other thing. Is there an easier way for users to park calls and pick them up than to manually transfer them to an extension?
11:33.58DrAk0Rienzilla, hold button on the phone
11:34.14Rienzillayes but the users here want to be able to pick up each other's call
11:34.34tzangerRienzilla: program a speed dial
11:34.39Rienzillaright now they can press hold, and then someone else can pick up that line by simply choosing the line x button on their handset
11:35.00tzangerRienzilla: that's because you've got a key system
11:35.10Rienzillatzanger: the speed dial is one thing, but it would be nice if the handsets indicated that there were calls parked on a specific extension
11:35.10tzangeryou want shared line appearances, which are (I think) still beta
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11:35.41tzangerRienzilla: you can do that with my long-ago merged patch that allows you to get the parked extension with ${PARKEDAT}
11:36.06Rienzillaok
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11:36.15Rienzillaand how do you announce that to handsets?
11:36.26Rienzillaor is that handset specific or impossible :)
11:36.53tzangerthat's up to your specific implementation
11:37.19DrAk0that sounds not simple:P
11:37.32Rienzillabah :)
11:37.39DrAk0im wondering if there is any 3 way call easy implementation
11:37.50Rienzillawell I'll give it a shot :)
11:37.54DrAk0and a good CTI for a call center
11:38.14RienzillaI might be able to program my phones to poll where calls are parked regularly
11:38.22Rienzillaor something like that
11:38.38DrAk0Rienzilla, which phones are you using
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11:50.47Worfhmm ... i suspect my problem has to do with the various possible nat settings in asterisk ...
12:01.59JT~SIPNAT
12:02.00jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
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12:05.44rofferHello, i have a question, i run Asterisk 1.4.4 and got a problem vith making a connection out with my sip provider. i can make internal calls and answer incoming calls from my sip provider but not make any calls out. i get this error. chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"unknown" anyone that can help me with this problem ?
12:06.56rofferand im allso confued, i used asterisk gui to setup sip account, and it puts it under user.conf, all other doc i can find they put it in sip.conf ?
12:06.59WorfJT: thanks ...
12:09.19*** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es)
12:09.22casixhello
12:09.35casixI'm making a call back system with an ivr
12:11.15casixwhen the .call file is readed it calls to the destination. After call (when is ringing) asterisk doesn't wait until it is answered. Asterisk continues executing. Can I make asterisk wait until it is answered??
12:14.09RienzillaDrAk0: great. callbacklogin works like a charm
12:14.26Rienzillaonly annoying thing is that my snom's register calls which are answered by another handset as a missed call
12:19.34*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
12:23.27DrAk0Rienzilla, it happens even with soft phones
12:23.53*** join/#asterisk yang (i=yang@static-ip-62-75-255-125.inaddr.intergenia.de)
12:24.29yangIf someone wants to dial +XXXX number, can I simply add that to the dialplan - exten => _+.,1,Dial(SIP/gsm/${EXTEN})
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12:26.58DonAlexGuys..
12:27.08DonAlexwhy is it asterisk is ignoring my dialrules?
12:27.21DonAlexkeeps trying to make every number dialled and extension?
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12:28.23pithenis the xorcom guy here? (sorry, can't remember your nick)
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12:36.05tzangertzafrir I believe works for xorcom
12:36.34DarKnesS_WolFpithen: what is with ur xorcom ?
12:38.35DonAlexYes h does but not here atm..
12:38.40DonAlexwhat seems to eb the probelm ?
12:38.48yangpithen: tzafrir
12:39.03lirakisDoes any one know how asterisk generates the "tag=" for To: responses?   Is it some type of hash on a part of the invite?
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12:51.12pithenthanks guys- actually i think I just got the answer I was looking for from one of their vendors
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13:03.41*** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk)
13:08.00tzangercoppice: oh hai, I solved that weird 3rd harmonic problem, although I still dont' understand why it was consistently a 3rd harmonic
13:08.08*** join/#asterisk xenonex (n=xenonex@92.47.14.21)
13:08.20coppice3rd rate, of course
13:08.51tzangercoppice: the 32 timeslots coming back were consistently missing the LSB
13:09.20LsodiHi, I would like to record calls on asterisk server, can someone recommend software for recording?
13:09.25*** join/#asterisk SteveTotaro (n=root@pool-71-166-110-9.bltmmd.east.verizon.net)
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13:09.57coppiceis this ulaw or alaw?
13:10.25tzangercoppice: alaw
13:10.26SteveTotaroanyone try druid yet?  i have tried so many GUIs, I am curious if this is worth the time and effort of testing?
13:10.41tzangercoppice: basically I send to the codec and it sends back one bit time late... for all 32 timeslots
13:11.01tzangerI was rotating the bits *in* each timeslot, instead of rotating the entire stream
13:11.35coppicemissing the LSB is normal on robbed bit T1s, and it just reduces the quality a little. I'm not sure if alaw coding falls apart. alaw does something tricky with the first 2 segments of the pseudo-log curve
13:11.37tzangerso it was 0765432107654321076543210, and I was rotating it per-byte instead of per-stream :-)
13:11.47*** join/#asterisk svenna_ (n=svenna@p548D35CF.dip0.t-ipconnect.de)
13:11.49tzangeryeah I read a lot of alaw theory
13:12.02tzangerthey linearize the first bit of the code
13:12.44tzangerand they invert the even bits, as you described :-)
13:14.02tzangerafter they fixed the fpga, audio I got abck from the codec on the far end of the tdm bus was clean :-)
13:14.25coppicethe inverting the even bits is really odd. they need inverting, but it should have been an aspect of the comms channel, and not the codec
13:14.27tzangerthe fpga sees it as a stream of bits, whereas the blackfin sees a stream of 8 bit words
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13:15.00coppiceblackfins seem to be flavuor of the month :-)
13:15.04tzangeryeah that blew me away when I read that, but you explanation was very good
13:15.44tzangerI like 'em
13:16.11tzangerIf I spent the time I'm sure I could have bit-rotated the stream nicely or played with the dma settings to delay the bit correctly, but fuck it, there's an fpga guy, make him work a little :-)
13:16.29tzangerI already had to deinterlace the damned bit stream
13:16.34tzangergotta love old hardware
13:16.53coppicethey are the only really successful attempt to made a general purpose + DSP core that does both jobs well. however, people complain that it looks like ADI ran out of cash when Intel pulled out, and the thing never got finished
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13:17.28tzangerOMAP was interesting
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13:18.06coppiceOMAP is a general purpose core + a DSP core. that's the route a lot have taken. the blackfin is the only successful combined core
13:18.58tzangeryes... what I don't like about OMAP is the dual cores.  it was interesting, but looks like a bigger pain in the ass than it's worth
13:19.16*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)
13:19.17tzangertrance first thing in the morning... nice
13:19.33Kattymorning!
13:19.43tzangerwoot
13:19.55Kattytzanger: quick! tell me if you can record a zap line.
13:20.01coppicethere is good and bad. you can run Linux on the ARM, while the DSP core is not handcapped by the timing vagary of linux
13:20.03Kattytzanger: from start to finish of the call!
13:20.33tzangerKatty: yes of course you can
13:20.42tzangercoppice: indeed
13:20.52tzangercoppice: actually I've been buried in realtime all week
13:20.54Kattytzanger: is it record, monitor, or mixmonitor?
13:21.04tzangermonitor or mixmonitor is what I use
13:21.07coppicehm. it looks like the UK courts have finally fallen into line with the rest of .eu over patentability
13:21.41tzangeractually had a design specc'd out that recorded every single call, internal or external, catalogued it, converted it and archived it for sbarnes-oaxley or however you spell it
13:22.14tzangercoppice: on x86 anyway I was able to see max 32us latency on a shitty via c7, throwing everything short of a kitchen sync at it
13:22.33coppicesarbanes-oxtongue
13:23.21coppiceat the interrupt level, maybe. not at the apps level. also, that latency is based on big assumptions about other peripherals
13:23.32*** join/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com)
13:24.08tzangercoppice: that is at the apps level
13:24.38BobLutzMorning all
13:24.38tzangerthat's a userspace app running latency tests with network, interrupts and flat-out CPU
13:26.01tzanger1.2GHz Via C7
13:27.56coppiceyou are imposing some severe constraints on the machine's activities if it can always respond so crisply
13:28.05Kattytzanger: can i pastebin some stuff and have you give it a quick look see?
13:28.11Kattytzanger: in regards to mixmonitor
13:28.11tzangerI can try
13:28.37tzangercoppice: naturally, it's an embedded system, but I'm not doing anything overy weird to it
13:28.45tzangerobviously zero power saving
13:28.57tzanger(although I have a hard time calling any x86 "embedded" :-)
13:28.58*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)
13:29.10BobLutzlol
13:29.25coppicegeode?
13:30.06BobLutzQwell: Should I not ask questions in #asterisk-dev ?
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13:31.06tzangereven geode I don't refer to as embedded
13:31.08tzangertoo PC like
13:31.30DonAlexAwww crap..
13:31.35DonAlexnow why is this happening..
13:31.36DonAlexhandle_request_invite: Call from '' to extension
13:31.42*** join/#asterisk atb_ (n=d@196.25.36.129)
13:31.53DonAlexwith every bloody number..
13:32.06DonAlexwhat macro got deleted or something?
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13:33.56atb_hi there, i know it aint the right place but possibly sombody can assist as ive been searching around for the past two days and cannot fix such a smalle problem, anybody here had any experience with a2billing integration and care to assist ?
13:34.49Kattytzanger: [from-pstn-recorded]
13:35.15*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)
13:36.09Kattytzanger: http://angela.sleekgeek.org/2008/03/14/record-incoming-zap-line/
13:36.18tzangerhahah sleekgeek
13:36.33Kattyyes, that's our website.
13:36.44Kattytzanger: that's what i've got in mind.
13:37.28tzangerthat looks about right.  I would probably sanitize the caller id first, checking for empty or invalid and replacing iwth 'unknown'
13:37.49tzangeralso you're mixing , and | but it's not a federal crime, only a misdemeanor
13:37.57yangIf someone wants to dial +XXXX number, can I simply add that to the dialplan - exten => _+.,1,Dial(SIP/gsm/${EXTEN})
13:38.55*** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net)
13:39.07ZaVoidmorning guys
13:39.10Kattytzanger: good idea.
13:39.14ZaVoidi got a strange thing herei can't get right
13:39.24Kattytzanger: why is mixing a misdemeanor?
13:39.37ZaVoidi'm trying to dial out... and do exten => s,5,Background("beep") after the call is answered
13:39.39Kattytzanger: i thought if you told someone first... :/
13:39.45ZaVoidbut it doesn't always play the beep at the right time
13:39.57ZaVoidi'm guessing i'm using the wrong command to wait for supervision
13:40.46De_Monyang _XXXX would be the better way to handle 4 digit extensions _+ is just plain dangerous
13:41.11yangDe_Mon: it can be any amount of digits I made it _+.
13:41.26tzangerKatty: it's just inconsistent
13:41.32Rienzillahmm
13:41.34tzangerlike telling someone you like them but then ignoring them
13:41.39tzangerdrives people batty :-)
13:41.41Rienzillacan you detect in your dialplan whether a queue has agents logged in?
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13:42.05De_Monyeah, I just realized + isn't a valid pattern, still half a sleep I spose.
13:42.25yangDe_Mon: its not valid?
13:43.11yangDe_Mon: any idea how can he otherwise dial the number from his mobile phone, which aere enterred as +XXXXXX
13:44.01*** join/#asterisk iulius (n=iulius@mail1.technologieshq.com)
13:44.34ZaVoidits as if the BEEP never plays sometimes
13:44.48hmmhesays~beep
13:44.48jbotbeep is probably the protocol formally known as bxxp
13:45.38BobLutzIn app_read.c, What does "int res" represent?
13:45.58BobLutz"int res = 0" line 89 to be exact
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13:49.02awkhello putnopvut, with bug 12127 you had any issue with v3 patch? v2 was shoddy, and caused locks
13:49.20awkbusy building a rpm package with the new v3, and see if it works...
13:49.30*** join/#asterisk javar (n=javar@69.79.134.24)
13:49.48putnopvutawk: v2 caused locks? Did you mention it on the bug?
13:50.47awkwe gave the info to jvandal... if he doesn't report it i will...
13:51.07putnopvutawk: wait...
13:51.10putnopvutdo you mean 12098?
13:51.31awkalso, you closed that thread on 1.4.17 iax io threads... I can re-produce that, just instaled a version now on a few clients with debug info...
13:51.58putnopvutawk: I have no idea what you're talking about with regards to iax io threads.
13:52.06*** join/#asterisk oej (n=olle@ti400720a080-9600.bb.online.no)
13:52.09awkhmm, iax runs out of threads..
13:52.17awkwill show you bug id, wait...
13:54.05awkhttp://bugs.digium.com/view.php?id=11790&nbn=3
13:54.22awkrussel was dealing with it.. sorry..
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13:54.43putnopvutawk: Okay.
13:55.01putnopvutawk: So v2 on 12127 was causing locks? Are you sure you don't mean issue 12098?
13:55.21awkI have the lock file...
13:56.32putnopvutCould you upload it to the issue, please? That would be very helpful.
13:56.56awkhmm, sorry the lock file I have is for issue 12098.. only 1 client has issue with v2
13:57.12putnopvutawk: Ah, that makes more sense.
13:57.35awkdo you also need that?
13:57.52putnopvutawk: no, I think I've cleared up the locking problem with v2.
13:58.03awkgreat, let me try that patch...
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13:58.09putnopvutjvandal reported that v3 doesn't crash or lock, but he has a lot of stuck channels.
13:58.16putnopvutI'm not sure if it's related to my patch or not though.
13:58.23awklet me quickly ask him on msn
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14:00.21awkjoel: I'm doing more test
14:01.11awksimplify dialplan and will send on ticket, with the patch asterisk doesnt crash, no locks, etc, but have 'zombie' channels,
14:01.15awkthats what he says...
14:02.18putnopvutI'll try to see why there could be zombie channels with that patch. If I find out why, I'll report on the bug.
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14:03.00awkthanks... will keep updates on ticket too.. just doing some more tests myself..
14:03.14putnopvutawk: Thanks for testing :)
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14:19.58*** join/#asterisk jks2 (n=jks@87.57.88.86)
14:21.04jks2using * 1.4.x how can I store voicemail message in a mysql database? (the mysql vm interface seems to be gone and the realtime interface seems to allow me only to store voicemail users in the db)
14:21.11ZaVoidso how do i wait for an answer? or progress? to move on in my context?
14:21.14*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582425.dsl.bell.ca)
14:21.44*** join/#asterisk op3r (n=Op3r@222.127.88.164)
14:22.11op3rhello whats the command again to show how many channels you have for the e1? zap show status?
14:22.20op3ror zap show channels?
14:22.55sysreqjks2: you have to compile 'odbc voicemail storage' support and then use mysql through odbc..
14:23.03sysreqjks2: http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
14:24.04*** part/#asterisk jivco (n=jivco@85.187.217.6)
14:24.30jks2sysreq, okay, I was looking at that -- no way to go directly to MySQL instead of through unixodbc, right?
14:25.17sysreqjks2: i don't think so, no.
14:25.36sysreqalso, beware that (taken from that page, at the bottom).. "I tried adding ODBC message storage to a 1.2.5 system already using MySQL for RealTime... Not a good idea, but using ODBC for both Realtime and msg storage seems good so far. -X1Z"
14:25.52*** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net)
14:26.06jks2hmm, but 1.2.5 is a long time ago
14:26.07sysreqi don't know if it still applies to 1.4 though.
14:26.30sysreqyeah, i agree.. but just keep that in mind if you ever go through problems ;p
14:26.40jks2hehe, I'll do that :)
14:27.45*** join/#asterisk jarod14 (n=jarod14@ns2.viatelecom.com)
14:28.09*** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com)
14:28.28JayTee52I have 2 SIP phones setup on a new server to test, extensions 5154 and 5146. I can call 5146 from 5154 but when I try to call from 5146 to 5154 I get a fast busy and a 603 on the phone display.
14:28.37ZaVoidanyone got any ideas? i want to dial using SIP.. and when the far end users picks up run a macro.. but not till they pick up the phone
14:29.05rofferHello, i have a question, i run Asterisk 1.4.4 and got a problem vith making a connection out with my sip provider. i can make internal calls and answer incoming calls from my sip provider but not make any calls out. i get this error. chan_sip.c:11848 handle_response_invite: Failed to authenticate on INVITE to '"mynumber" anyone that can help me with this problem ?
14:29.08*** join/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
14:29.11[TK]D-FenderZaVoid: use a "call file" or "AMI Originate"
14:29.19*** join/#asterisk fedya (n=fedya@75.112.143.226)
14:29.21methodsanyone use the perl agi lib's ?
14:29.32methodssorry i mean Pg
14:29.32*** join/#asterisk ddunavant (n=David@pool-71-178-115-160.washdc.east.verizon.net)
14:29.41ZaVoidi am doing that fender
14:30.15ZPerteeIf the first line of my dialplan is exten => s,1,Wait(10) does that mean that it will wait 10 seconds before answering?
14:30.18DarKnesS_WolF[TK]D-Fender: hello dude :) how are u ?
14:30.41[TK]D-FenderZPertee: Depends on what kind of channel you are answering
14:31.07[TK]D-FenderZPertee: Actually.... I should say "yes" to that///
14:31.15[TK]D-FenderDarKnesS_WolF: Getting by
14:31.29JayTee52On the server console I get the message: Warning[10793]: app_dial.c:1115 dial_exec_full: Dial argument takes format (technology/[device:] number1)
14:31.40ZaVoidfender http://pastebin.com/d20a92db4
14:31.45ZPertee[TK]D-Fender: ok thanks.  it will be a zap channel.
14:31.58DarKnesS_WolF[TK]D-Fender: have u ever had a snom phone behind nat ? i'm become unreachable in 1 min and 4 sec
14:32.12[TK]D-FenderDarKnesS_WolF: Read up :
14:32.13[TK]D-Fender~sipnat
14:32.14jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
14:32.54[TK]D-FenderZaVoid: Depends on what your channel is doing...
14:33.04ZaVoidi'm playing Background BEEP
14:33.06ZaVoidthen reading digits
14:33.15[TK]D-FenderZaVoid: the one you're CALLING.
14:33.15ZaVoidbut someitmes the BEEp plays before teh call is answered
14:33.20ZaVoidyep
14:33.30ZaVoidsometimes it plays after its answered.. seems kinda random
14:33.36DarKnesS_WolF[TK]D-Fender: i did read it :( nop the phone is NATED the asterisk on public
14:33.53[TK]D-FenderDarKnesS_WolF: And you're not showing me anything...
14:34.05[TK]D-FenderZaVoid: And you still aren't showing much either.
14:34.12ZaVoidyeah i know one sec
14:34.34*** join/#asterisk anonymouz666 (n=anonymou@201.19.227.137)
14:35.17JayTee52ok, disregard my question. I figured it out. damn typos! grrrrrrr arrrggghhh
14:35.30DarKnesS_WolF[TK]D-Fender: sure ask to show u anything :-) the phone register after 1 min exactly it become unreachable
14:35.50[TK]D-FenderDarKnesS_WolF: Try ahrder.
14:35.53[TK]D-Fenderharder*
14:36.16ZaVoidfender: http://pastebin.com/d7ce6b298
14:36.32rofferneed some help with auth, i get Failed to authenticate on INVITE to, i have tried auth in sip.conf and register, but cannot make a outgoing call. i can call intern and call inn to the sip privider configurd with asterisk gui inside users.conf ? anyone ?
14:36.39DarKnesS_WolF[TK]D-Fender: i think if itis 1 min then i need to add option for like defaultexpire or so
14:37.09[TK]D-FenderZaVoid: I don't see the context for your Channel....
14:37.23[TK]D-FenderDarKnesS_WolF: Show first, comment second....
14:37.38ZaVoidline 7
14:37.49*** join/#asterisk JunK-Y (n=junky@modemcable153.55-201-24.mc.videotron.ca)
14:37.59ZaVoid[anicallback-leg2]
14:38.34DarKnesS_WolF[TK]D-Fender: yes what u wanna to see ? there is nothing on logs :-D tell me show me that and this and i'll do
14:39.29[TK]D-FenderZaVoid: and your call-file says : Channel: Local/17328530514@anicallback-leg1/n
14:39.37ZaVoid#
14:39.37ZaVoidContext: anicallback-leg2
14:39.38[TK]D-FenderZaVoid: Leg1 <-
14:39.45*** join/#asterisk draygon-w (n=draygon@gateway5-pnap.exigo.com)
14:39.47ZaVoidahh
14:39.59[TK]D-FenderZaVoid: I said your CHANNEL, not where you dump it after it ANSWERS!
14:40.23*** join/#asterisk VJFROMGT (n=vjfromgt@pool-96-232-13-228.nycmny.east.verizon.net)
14:40.31ZaVoidso i'm still answering in anicallback-leg1 then
14:40.31DarKnesS_WolF[TK]D-Fender: Mar 14 16:13:39 NOTICE[4134]: chan_sip.c:10078 handle_response_peerpoke: Peer '187' is now REACHABLE! (372ms / 5000ms)
14:40.34DarKnesS_WolFMar 14 16:14:43 NOTICE[4134]: chan_sip.c:11867 sip_poke_noanswer: Peer '187' is now UNREACHABLE!  Last qualify: 372
14:40.51VJFROMGTsuddenly i hear no audio , nat looks ok
14:40.54VJFROMGTany suggestions?
14:41.23[TK]D-FenderZaVoid: I want to see what it dials....
14:41.29ZaVoidok
14:42.02VJFROMGTif i restart the machine problem gest solved for a few hours
14:42.04rkeeneAnyone else have a problem with the "tor2" module causing a kernel panic ?   http://www.rkeene.org/viewer/tmp/asterisk/zaptel-causing-crash.txt.htm
14:42.37*** join/#asterisk axisys (n=axisys@155.70.141.45)
14:42.41yanghey [TK]D-Fender can my dialplan start as exten =>_+.    for numbers starting with +
14:43.11[TK]D-Fenderyang: I might think so, but have never had to work with "+".  Go try.
14:44.12ZaVoidLocal/17328530514@anicallback-leg1-aff9,2"
14:45.09ZaVoidbecause if read callfiles correctly... If the call answers, connect it here      * Context: <context-name> Context in extensions.conf
14:45.29ZaVoidso it sholdn't do anythign in the context until after the call actually connects.
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14:46.36jasonWootdoes anyone have any experience with intuitivecreation's AMA suite?
14:47.09rofferneed some help with auth, i get Failed to authenticate on INVITE to, i have tried auth in sip.conf anything else i can try ?
14:47.38JunK-Yrkeene: which kernel?
14:47.42ZaVoidso this happens before my phone even rings... so i'm getting false progress messages i guess
14:47.43ZaVoid<PROTECTED>
14:47.43ZaVoid<PROTECTED>
14:47.45[TK]D-FenderZaVoid: "Channel: Local/17328530514@anicallback-leg1/n" <- show me what this is calling.....
14:48.05ZaVoidyou mean the DIAL command fender?
14:48.33[TK]D-FenderZaVoid: I want to see the whole damn dialplan for EVERYTHING this thing is supposed to be using AND the CLI output to match
14:48.38rkeeneJunK-Y, 2.6.21.5-smp
14:49.02*** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net)
14:50.18JunK-Yrkeene: report this with OS on bugs.digium.com please.
14:50.56*** join/#asterisk CunningPike (n=arodgers@204.239.12.183)
14:51.40BobLutzDoes anyone here know where AST_FRAME_DTMF is defined?
14:52.06*** part/#asterisk methods (n=daquino@c-68-36-237-152.hsd1.nj.comcast.net)
14:52.59coppicetzanger: what about an OC768 PCIE card?
14:53.12JunK-YBobLutz: frame.h
14:53.28tzangercoppice: nah, that means I'd need something to connect it to :-)
14:53.53coppicemake 2
14:54.24putnopvutBobLutz: it's in include/asterisk/frame.h
14:55.07BobLutzJunK-Y: putnopvut: Thanks!
14:57.50rkeeneJunK-Y, Done:  http://bugs.digium.com/view.php?id=12213
14:58.57Qwell~nowwhat
14:58.58jbotSo you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E
15:00.16rofferneed some help with auth, i get Failed to authenticate on INVITE to, i have tried auth in sip.conf anything else i can try ?
15:00.45*** join/#asterisk RoyK (n=roy@box36.fortel.no)
15:06.54anonymouz666Ashley Dupre is just....amazing.
15:07.28anonymouz666I'd install an asterisk box for her for free.
15:07.29anonymouz666lol
15:08.05tzangerhaha
15:08.11tzangershe is not really good looking
15:08.26tzangershe's got a nice enough face and yeah she'd draw second looks,but amazing?
15:09.16anonymouz666yeah, perfect.
15:09.20anonymouz666IMHO
15:09.49BobLutzIs there a specific reason app_swift.c was never merged into the Asterisk project?
15:10.10QwellBobLutz: because the author chose not to contribute it
15:10.54*** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
15:11.01BobLutzQwell: Interesting ...
15:11.07*** join/#asterisk bmg505 (n=leon@196-209-76-155-tbnb-esr-2.dynamic.isadsl.co.za)
15:11.13filerkeene: that is already fixed in the latest Zaptel
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15:12.03Qwellfile: the tor2 thing?
15:12.14Qwellmight want to poke sruffell before he looks into 12213..
15:12.34filepretty sure kpfleming fixed it with revision 3863
15:12.57filebut... could be wrong
15:14.34tzangerstruffel?
15:17.47jks2sysreq, seems to be a problem yes...asterisk just crashes every time I run the voicemail app :-|
15:20.21*** join/#asterisk ddunavant (n=David@pool-71-178-115-160.washdc.east.verizon.net)
15:26.46casixI'm making a auto-dial with ivr syste. When the .call file is readed it calls to the destination. After call (when is ringing) asterisk doesn't wait until it is answered. Asterisk continues executing. Can I make asterisk wait until it is answered??
15:26.51*** part/#asterisk C4colo (n=DJpyro@67.41.154.214)
15:26.58agxI've 2 xDSL and 2 public's IP to the asterisk box: but there is seems no way it accepts 2 IP in externip= for using SIP trunks
15:28.18casixbut i'll have the same problem
15:28.27rkeenefile, I'm using the latest Zaptel
15:28.37rkeenefile, And so I must conclude that it is not fixed
15:29.05fileyou mentioned 1.4.9, the latest is 1.4.9.2
15:29.05sysreqjks2: err.. and that's with odbc storage and mysql realtime? i guess you're going to have to use mysql through odbc for your realtime as well ;\
15:29.34*** join/#asterisk ming_zym (n=ming_zym@220.181.54.88)
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15:29.43*** mode/#asterisk [+o russellb] by ChanServ
15:30.41jks2sysreq, yes, odbc storage and mysql realtime :-|
15:31.00rkeenefile, Arg, regression
15:31.37rkeenefile, Close the bug with extreme prejudice
15:32.08jks2sysreq, just a bit odd that voicemail simply makes asterisk crash... mysql realtime works fine, but asterisk crashes when the voicemail app is started
15:32.57sysreqjks2: .. but then using odbc everywhere will allow you to have a more homogenous solution; plus if you decide to switch to another database system along the way, you'll only have to modify you odbc configs.
15:33.11jks2oh well, I don't think I'll ever do that
15:33.20jks2doesn't odbc introduce an overhead?
15:34.44sysreqi don't know, i've rarely used it
15:34.55*** join/#asterisk jpeeler (n=jpeeler@216.207.245.1)
15:35.09sysreqbut i wouldn't think so
15:35.22jks2I would think so :-|
15:37.14*** join/#asterisk outtolunc (n=me@c-67-174-216-60.hsd1.ca.comcast.net)
15:38.30BobLutz"if(f->frametype == AST_FRAME_DTMF) { ... }", if I want to read in more then 1 DTMF, I should change "AST_FRAME_DTMF" to "AST_FRAME_DTMF_BEGIN", right?
15:42.35filebegin is a frame to indicate the start of a single DTMF, and an end is a frame to indicate the end of a single DTMF (the same DTMF as the begin), if you need to read in multiple DTMF digits you need to probably store the digits from end frames in a buffer
15:43.56*** join/#asterisk jivco (n=jivco@85.187.217.6)
15:44.04BobLutzfile: I am trying to modify app_swift.c (not official * code i know) to have functionality similiar to app_read.c, I am combing back through app_read.c, and I think I see my mistake, thank you
15:46.22*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
15:46.50ZeeekAnyone feel like talking? Maybe asterisk 1.6? I dunno. http://voipusersconference.org
15:47.20ZeeekCall (724) 444-7444 and enter 22622# 1#
15:47.35QwellZeeek: are the commercials over?
15:47.39Zeeeknever
15:47.44Zeeekhow can I pay for gas?
15:48.03Zeeekor the vegetarian restaurant where I get gas?
15:49.00JunK-YZeeek: can we call via IAX2? :)
15:49.05Zeeeksure
15:49.10Zeeekvia your provider :)
15:49.29JunK-Ywhat about if my providers are zap only? :)
15:49.38ZeeekCall (724) 444-7444 and enter 22622# 1#
15:49.48Zeeekit's worth it
15:50.03BobLutzI called a couple weeks ago
15:50.21ZeeekBobLutz and were you treated with the proper respect ?
15:50.35BobLutzSomeone was like "Who is this from Ohio?"
15:50.41BobLutzI got scared and hung up
15:50.47ZeeekBecause we can't see who you are
15:50.57KattyHORAY!
15:50.59KattyMY PRI WORKS!
15:51.00Zeeekwe see what you telco reports as an area code
15:51.09*** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
15:51.09*** mode/#asterisk [+o lmadsen] by ChanServ
15:51.31ZeeekToday, Barak Obama shares his dialplan secrets with us
15:51.45Strom_CZeeek: usually, if you're running a conference, etiquette dictates that you don't reveal information about the callers
15:52.11ZeeekI have no other way to ask the person to speak UNLESS they register with Talkshoe
15:52.23ZeeekIf they do, I see a pseudo and have NO other info about them
15:52.31Zeeekwhich is a GoogThing (tm)
15:52.50Zeeekas asterisk freaks, you are all able to change your CID anywway, no?
15:53.01Strom_Cthats not the point
15:53.11Strom_Cthe point is you don't announce "Oh look, someone from Ohio"
15:53.19Zeeekok here is the point: I HAVE NO WAY to ask someone to speak
15:53.36Strom_Cgood.  Let them speak when they're ready.
15:53.52JayTee52Katty, I'm getting ready to use 2 PRI circuits on our Asterisk box. Is your PRI an incoming PRI from your telco provider?
15:54.08Strom_Cdoes a "telco provider" provide telephone companies?
15:54.14KattyJayTee52: yes, it is.
15:54.19KattyJayTee52: i'll be blogging it today, too.
15:54.20Strom_Cor did you mean just 'telco'?
15:54.22Strom_C:P
15:54.24KattyJayTee52: would you like the link when i'm done?
15:54.30*** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
15:54.30*** mode/#asterisk [+o blitzrage] by ChanServ
15:54.30JayTee52yes, please
15:54.32Kattyk
15:54.35Kattyhi blitz (=
15:54.48coppiceonly the desparate would blog about a PRI
15:54.48blitzrageKatty: hi!
15:54.57blitzrageKatty: you know that blitzrage == lmadsen right? :)
15:54.58Kattycoppice: i blog everything, dear.
15:55.09Kattycoppice: my memory requires it
15:55.13coppicethat's even worse
15:55.17Kattyhehehe
15:55.20Kattyoh well.
15:55.21Kattyit's handy
15:55.27Kattyit will continue.
15:55.33Strom_Cwhat the hell ever happened to notepaper? :P
15:55.42Kattyi can't ready my own handwriting.
15:55.47Kattyi type fast that i write anyway ;)
15:56.02Strom_Cand clearly not very accurately...
15:58.05anonymouz666Katty: I can't just write without a keyboard
15:59.11*** join/#asterisk CrashSys (n=kumba@216-199-37-76.tpa.fdn.com)
16:00.09*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
16:00.10CrashSysSimple question about global var's. If I have status=1 as a global car, from a channel can I change it to status=0, and will it save that?
16:00.15*** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it)
16:00.41CrashSyss/car/var
16:00.55BobLutzCrashSys: I beleive so...let me check something
16:01.14DarKnesS_WolFmmm been hours i'm wih this problem :-s
16:01.17CrashSyswell I can do it the amish method... just try it on a test DID...
16:01.33BobLutzCrashSys: I do something like that in my extensions.conf and it works
16:02.47Kattydoes the [from-pstn] bit in extensions.conf handle incoming calls the same way with a pri as regular analog ports?
16:02.54Kattys,1, s,2, s,3 etc?
16:03.06BobLutzCrashSys: status will only be 0 for that channel that changed it though
16:03.07DarKnesS_WolFKatty: yes i think so
16:03.19KattyDarKnesS_WolF: do you know about DIDs?
16:03.30CrashSysBobLutz: What if I want to change the global var for all channels after that?
16:03.48DarKnesS_WolFKatty: nop :-)
16:03.49CrashSysisn't there a setglobal?
16:03.56KattyDarKnesS_WolF: okie dokie
16:04.03CrashSyskatty: replace the s with the DID the carrier is sending
16:04.09DarKnesS_WolFKatty: what i thikn u'll need to create siperated gorups for ur DID's PRI channels
16:04.19CrashSysso if the carrier is sending the last 4 digits as the did, you would replace S with the last 4 digits...
16:04.22DarKnesS_WolFthen u make context for each one
16:04.36DarKnesS_WolFand treat them the way u like
16:04.46BobLutzCrashSys: For some reason, I think you can NOT change the global var for all channels..
16:04.47CrashSysso it would be <did>,1,command; <did>,n,command
16:05.09CrashSysBobLutz: well that makes the possibility of a night button difficult...
16:05.17BobLutznight button?
16:05.35BobLutzCrashSys: You were correct
16:05.39BobLutzpage 434
16:05.40CrashSysyes... old-school KSU/PBX feature...
16:05.45*** join/#asterisk seanbright (i=seanbrig@65.207.74.18)
16:05.48CrashSyshit a button, and the system plays a "night menu"
16:06.00CrashSysI wish I had a better memory sometimes :(
16:06.06Strom_CCrashSys: don't do it with globals
16:06.10BobLutzSet(GLOBAL(var)=val)
16:06.10Strom_Cdo it with database settings
16:06.25BobLutzCrashSys: Are you talking about a GotoIfTime()?
16:06.27Strom_Cthat way, night mode will stay on until you explicitly turn it off
16:06.34CrashSysstrom: you mean astdb?
16:06.43CrashSysor mysql?
16:06.45Strom_Castdb
16:06.46KattyCrashSys: dankou.
16:06.53CrashSysok...
16:06.58Strom_Cmysql is clearly overkill for your application
16:07.04CrashSysyes, I was about to say :)
16:07.13CrashSysI've never tried to store anything within astdb
16:07.19Strom_Cit's super-easy
16:07.24CrashSysI shall now search voip-info.org for enlightenment :D
16:07.45CrashSysdbput/dbget?
16:07.54*** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep)
16:08.46Strom_Cjesus, that's ancient syntax
16:08.49Strom_Cuse the DB() function
16:09.08CrashSysone sec, let me fire up a CLI
16:09.12Strom_Cis the wiki seriously still that out-of-date?
16:09.22CrashSysYes, there is still pre 1.0 stuff on the wiki :)
16:09.31CrashSysthat's what, 4 years old now?
16:09.33Strom_Cdoes it still say things like "NOTE!!!!  As of 0.8.4...."
16:09.54CrashSysthe astdb page is pre 1.2 release :)
16:10.01Strom_Cwigh
16:10.04Strom_Cer, sigh
16:10.06Strom_Cwhat a mess
16:10.25CrashSys:) Has digium hired a guy to do nothing but correct voip-info.org yet?
16:10.43CrashSysImagine the money you'd save on tech support calls by having accurate 1.2/1.4/1.6 pages in there
16:10.50CrashSysor they
16:13.34BobLutzWhen you Google anything Asterisk, voip-info.org is the first always, lol
16:14.49*** join/#asterisk comprookie2000 (n=david@adsl-065-012-210-216.sip.bct.bellsouth.net)
16:16.22*** join/#asterisk robeph (n=robf@router.asteriasgi.com)
16:16.32bkruseBobLutz: That is a huge reference, even moreso before there was a lot of documentation
16:16.39bkruselike 'the book', asteriskguru, etc
16:16.48robeph[TK]D-Fender: I found it today... hahah the problem was so...simple yet not readily apparent
16:16.49BobLutzI frequent the site a lot, people in here tell me not to
16:17.17x86why would someone want to use SER (or OpenSER) with Asterisk?
16:17.20seanbrightrunning 1.4.18 - i have a SIP phone set up and working, making calls out over a PRI via Zap.  when the party on the other end answers the call, i get 1 CDR record.  when the party on the other end does not answer, i get 2 CDR records.  i would like to only have 1 in either case...
16:17.52robeph[TK]D-Fender: their dns was broken,  but this didn't effect any normally dns requisite services,  since host file or direct IP use bypassed this problem all together,  what the problem was is that for some reason SRV was enabled in sip.conf and every time it'd try to dial it would send srv requests like they would soon be out of style.
16:18.10robeph[TK]D-Fender: turning this off...everything works fast and new.
16:18.57robephthough I'm not particularly sure of the byproduct of turning SRV requests off for sip will be.  does anyone?
16:20.03coppicejameswf: http://www.xkcd.com/
16:32.36*** join/#asterisk Entr4nced (n=IMG001@67-129-213-39.dia.static.qwest.net)
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16:33.24[TK]D-Fenderrobeph: Seems to cause miraculous speed-ups for people with maligned DNS servers ;)
16:34.19*** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com)
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16:35.07robeph[TK]D-Fender: :p..  well the 36 second timeout must be set somewhere,  which would account for that exact time wait that each dial received..
16:35.19robephI just wondered what the purpose of using it at all would be in the case of SIP
16:35.42robephin this particular case the failover/static load balancing of SRV records seems silly
16:36.40robephthough I assume it may have some uses if using sip trunking,  but i guess its not very bright in assumptive reasoning :p
16:49.42draygon-wyang did you get the + thingy resolved?
16:50.21yangdraygon-w: i wasn't able to test it
16:51.14yangdraygon-w: from what i saw on the CLI the customer succesfully connected, but i am not sure
16:51.59*** join/#asterisk af_ (n=getsmart@88-149-230-191.dynamic.ngi.it)
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16:53.18dikdusthi
16:54.31dikdustis there a way to look if I have putted mysql support in * ? guess I haven't compiled asterisk-addons package .-.
16:54.59dikdustgrep sql on log => [Mar 14 17:50:10] VERBOSE[10958] logger.c: cdr_sqlite.so => (SQLite CDR Backend)
16:57.54*** join/#asterisk xenonex (n=xenonex@92.47.14.21)
17:01.14JunK-Ytype cdr status and look if you see mysql?
17:04.25*** join/#asterisk minthome (n=mintee@c-68-45-231-166.hsd1.nj.comcast.net)
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17:06.35blitzragecodefreeze: ping
17:07.54dikdustthanks JunK-Y ;) there isn't mysql support ... and with asterisk-addons can't see mysql support ...
17:09.13dikdustdamn guess I have taked 1.2 asterisk addons package .-.
17:10.19BobLutzIs it possible to have a SIP channel go through SSH?
17:12.25*** join/#asterisk ZaVoid (n=zavoid@66-95-182-90.client.dsl.net)
17:12.44codefreezeblitzrage: here am i
17:12.53blitzragecodefreeze: !!
17:12.58blitzragecodefreeze: I have a CDR question for you............
17:13.26*** part/#asterisk quigon (n=matias@190.3.121.15)
17:15.13blitzragecodefreeze: ok, so I've got a call that comes in from a CCM to asterisk, and then asterisk does a call forward (just another Dial()) back out to another CCM gateway. I have the exchange:  INVITE, 100 Trying, 180 Ringing. If the incoming leg to Asterisk then hangs up, I send the CANCEL, get a 488 back, then send a couple more CANCELs, and the exchange is done - however... in this scenario, my CDR disposition shows as 'ANSW
17:15.13blitzrageERED'
17:15.37blitzrageI have two CDR records, one for the incoming that shows NO ANSWER, and the 2nd leg shows as ANSWERED... it's almost like they are swapped...
17:16.49codefreezeblitzrage: CCM = Corba Component Module?
17:17.09fileCisco Call Manager
17:17.17blitzrageheh... what file said :)
17:17.33blitzrageI don't get a 200 OK or anything... at least not from the 2nd call leg
17:17.56codefreezeOK, I was hoping for Cute, Cuddly... something, but that'll do...
17:18.08blitzrageheh
17:19.00*** join/#asterisk xenonex (n=xenonex@82.200.211.5)
17:20.05codefreezeblitzrage: I have no quick answer to that one. This sounds like the code that does this is in chan_sip somewhere.
17:20.42blitzrageya... :(
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17:20.48blitzragenot entirely sure where to start debugging that one
17:20.52codefreezeBecause of masquerading, I'd be not shocked if some sort of swapping might occur in edge cases
17:21.16blitzrageya... I did canreinvite=no because I thought that was causing it, but same issue
17:21.31blitzrageseems like the first leg should have been set to ANSWERED, and the 2nd to NO ANSWER (at least that would make more sense to me)
17:22.45codefreezeIt all comes down to the philosophy that CDR's store 3 events, and it gets real tricky sometimes picking out which channel & CDR an event applies to.
17:23.21*** join/#asterisk roffer_ (n=roffer@39.84-234-228.customer.lyse.net)
17:23.58roffer_need some help with auth, i get Failed to authenticate on INVITE to, i have tried auth in sip.conf anything else i can try ?
17:24.03codefreezeespecially down in the channel drivers, where it's juggling all that stuff on an event-driven basis.
17:25.14*** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com)
17:26.25blitzrageya... I'm wondering if I should file a bug with the traces to at least determine if it's a bug, or something I'm doing stupid, or if this is normal....
17:29.40dikdustok solved I' m a jerk .-.
17:29.52dikdustthanks a lot to everybody :=
17:29.54dikdust:)
17:32.06roffer_i have problem with auth when i try dial out with sip provider. log here http://pastebin.com/d3a776261 every thing else vork local, inbound.
17:32.33codefreezeblitzrage: it won't HURT to report this bug, it sounds like one to me. I'm torn: I spent a week or two in the chan_zap code, trying to straighten out transfer issues; but there's dozens of overlapping scenarios, and fixes to one can throw another off...
17:32.36*** join/#asterisk esaym (n=user@72.183.198.134)
17:33.33anonymouz666codefreeze: what to do then? :)
17:33.56codefreezeblitzrage: I've worked on the CEL stuff (no, Juggie don't close 10099 yet), but that was classed as 'recreational' time, so I threw myself into just trying to straighten out what's there...
17:35.40codefreezeThe whole thing has grown to be a mess, and needs to be re-engineered, but community input is definitely needed, and we need a spec, and agreement as to direction.
17:36.24anonymouz666re-engineered means rewrite most of code?
17:36.38*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)
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17:37.16blitzragecodefreeze: sounds like a very important discussion for astridevcon
17:37.55blitzragecodefreeze: ya, I think I'll file a bug and see if it's just the way I'm interpreting the data, or if it's some other underlying issue. Both channel legs are SIP, and there is no transfer going on. Just a call comes into Asterisk, and then Asterisk doesn't another Dial() back out
17:38.25blitzragehowever, I DO  a Playback() which might be causing the one ANSWERED disposition, but I think the bug is it being swapped on the CDR channels. I'll let you know when I've filed the bug.
17:38.43blitzragemight try taking out that playback to see if I end up with a disposition of NO ANSWER on both then
17:38.54*** join/#asterisk R0land (n=Roland@193.227.191.90)
17:39.03codefreezeanonymouz666: Well, sort of. The CDR struct as it is now, probably should go. CEL is fundamentally the right route, I think, but I haven't thought all the detail through for it. What info you need to tie the segments together, and how to store/provide that info is what chiefly concerns me.
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17:40.06R0landhello all!
17:41.24R0landim facing a certain issue at hand, 1-Zap/7 was active, and asterisk was continually logging status of zap/7 without connecting any cable to line No:7
17:41.45R0landwhat might be the cause of that!
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17:44.47BobLutzfile: The light bulb just turned on, thanks again!
17:45.44fileBobLutz: hmm?
17:47.09BobLutzfile: I was asking about AST_FRAME_DTMF vs AST_FRAME_DTMF_BEGIN earlier
17:47.12BobLutzhttp://www.pastebin.ca/942632
17:47.17fileah
17:47.38fileAST_FRAME_DTMF is really AST_FRAME_DTMF_BEGIN in disguise, it's there for old times sake
17:47.42BobLutzI format my code like a no0b...but I graduated only a couple months ago
17:47.45fileor is it AST_FRAME_DTMF_END...
17:47.58fileyeah, it's end
17:48.20JunK-YEND yeah (based on the define)
17:48.21BobLutzWell..For some reason, AST_FRAME_DTMF wasnt picking up the first DTMF...AST_FRAME_DTMF_BEGIN does
17:49.17BobLutzI need to clean that code up big time, but I would like to submit it (contribute?) to the Asterisk project
17:49.32*** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net)
17:49.50fileare you the original author of what you are going to submit?
17:50.22BobLutzeh...I took will@loopfree.net's app_swift.c and combined it with app_read.c from Asterisk
17:50.37BobLutzDynamic TTS + Read()
17:51.09BobLutzI dont know if app_swift.c is maintained anymore..I need to e mail him
17:51.59*** part/#asterisk R0land (n=Roland@193.227.191.90)
17:52.01filethat would be no then... which makes it more difficult, as in order to contribute you must sign a license agreement for what you are contributing... and since you are not the original author that is unhappy
17:52.07robephwhat I should do is stick a small bit of code to warn on SRV timeout,  so no one will spend 10 hours of trying to figure why asterisk receives Dial,  waits 36 seconds,  then sends invite..
17:52.20robephwhen dns isn't around.
17:52.30BobLutzfile: bah..oh well, im too young with my eyes too wide to know any better
17:52.40BobLutzfile: thanks for your help though
17:52.46robephyou have to sign license agreement for submitting code thats already gpld?
17:53.04robephodd..but okay.
17:53.29fileDigium has business edition which is not open source, in order to use contributions in that they have to be properly licensed
17:53.55BobLutzHmm.. I wonder if that is why will@loopfree.net never submitted app_swift.c originally
17:54.07fileit is possible
17:54.14robephBobLutz: do what a lot do and simply toss your patch up on a seperate server with a what and why,  if people find it useful they'll grab it,  stick it up on some of the message list / newsgroups,  people will flame you and praise you if its good... or both if not,  or sometimes niether and ignore you =)
17:54.28BobLutzprobably ignore me
17:54.29BobLutzlol
17:55.01robephI can't tell you how many "after market" patches I shove down the throat of my open source apps on my machines =\
17:55.25robephprobably 10% of what I build is patched in some form with various tweaks. =s
17:55.34robephprolly why all my stuff is always falling apart but hey....
17:55.52filethe positive thing about having it put in is that it's no longer up to you to maintain it, and it potentially gets a wider audience... but some individuals still dislike licensing it, which is fine
17:56.26robephfile: do not underestimate the audience of asterisk users ML and newsgroups... those are usually the people that count anyhow ;)
17:56.30BobLutzI kinda always felt like a chump in the back of my mind for using all open source software, I had always wanted to contribute back
17:57.08fileBobLutz: hehe
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17:57.30robephBobLutz: hahah well don't do that,  you're being gracious often times just using it ;).... the idea behind it is often just to create something useful and by it being useful you're using it to its purpose.   but if you CAN contribute,  doing so can do no harm,  unless you write code like me then you get bricks through your window late at night....
17:57.48BobLutzhaha
17:58.46Unihey all, got a problem, trying to add support for "if you know the ext of your part you can dial it..." and so I used Background(greeting) followed by a WaitExten(10,) but if I dial any numbers asterisk just hangs up the call, any ideas?
17:59.54*** join/#asterisk RobH (n=RobH@216.207.245.1)
17:59.57Uniguess the relevant config section may help: http://pastebin.com/m711a6906
18:01.08robephseriously though,  if you aren't the original licensee or you use multiple licensed libs etc.   Get yourself a lil host,  a free one for all it matters (not geocities... )   make a lil index.html with links to the directories with your tarballs for each lil project and an explination of what each is,  why its done,  what functionality it provides,  why its useful,  etc.   then post the link in your sig on newsgroups / mailing lists during your normal
18:01.23robephpeople will see and be curious,  I often follow links from sigs to such things
18:01.36robephfound lotsa neat stuff ,  others do too,  its free,  and easy to get noticed if you do good work
18:04.28BobLutzrobeph: Yea this works out cause I have a domain name, but nothing on the site
18:05.04*** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep)
18:05.25robephI mean nothing complex,  I'd model it like cpan for example,  just your name,  explination of what it is,  cut and dry,  no pretty graphics,  this is what you get download it or don't,  here's how it works,  have fun good day,  gpl, bye =)
18:05.39BobLutzhaha
18:06.01*** join/#asterisk DonAlex (n=DonAlex@host86-137-212-179.range86-137.btcentralplus.com)
18:06.08DonAlexWaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa
18:06.14DonAlexI want to cry.. :(
18:06.23robephyou already have.
18:06.31*** part/#asterisk TripleX1 (n=TripleX1@modemcable132.108-83-70.mc.videotron.ca)
18:06.38DonAlexWhat the hell am I doing wrong that asterisk is ignoring my dialrules?!
18:06.52robephcare to be a bit more informative?
18:06.58DonAlexEvery number I type it thinks is an extension.
18:06.59BobLutzDonAlex: `dialplan reload`
18:07.16BobLutzDonAlex: Misread, sorry
18:07.17DonAlexBobLutz: Done and Done.
18:07.31robephpaste the dialplan?
18:07.33*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)
18:07.43jameswf~pb
18:07.44jbot[~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
18:07.57robephyeah not in here,  that'd be bad. do what he says.
18:08.13robephin sial or rafb pastebins? :'(
18:08.21robephoh nm there is rafb
18:08.24DonAlexsure the snippet it is liek 3 lines
18:08.25DonAlex;)
18:08.52robephwell if its ignoring it,  the rest may be revealing as to why,  ie things overriding it etc.
18:09.31DonAlexhttp://pastebin.com/m4ee84f70
18:09.40DonAlexoh the owhole thing huh
18:09.43DonAlexno worrie
18:11.37DonAlexhttp://pastebin.com/d34785fc6
18:12.28DonAlexThat any help ?
18:14.33DonAlexIt is not like they are complicated rules.. I only JUST started configuring things!!?!
18:14.39robephheh
18:14.50DonAlexrobeph: *sighs*
18:14.55robephI still am not sure what you're passing and what you're getting back,  and what you expect to get back
18:15.32DonAlexrobeph: Well essentiall I am trying to dial an outside line. 123 is the speaking clock here int he UK
18:15.44DonAlexrobeph: a good way to make sure we are outside..
18:16.21DonAlexrobeph: so _123 is if it matches 123 right?
18:16.30DonAlexrobeph: add a 9 and send it out.
18:16.38BobLutzDonAlex: You might want to try setting timeouts
18:16.54BobLutzSet(TIMEOUT(digit)=3) for example
18:16.57robephk that 123 made no sense to me sense 123 here in us is ...well nothing.. or part of a (cc)-npa
18:17.04DonAlexbob: Timeouts for what exactly? It is recognising 123 as an extension
18:17.12BobLutz1-2-3
18:17.17DonAlexin fact it recognises ANY number as a bloody extension?!
18:17.37robephit is an extension... heh sort of
18:17.41DonAlexwait lemme show a real number
18:17.42DonAlex:)
18:18.14DonAlex[Mar 14 18:17:56] NOTICE[464]: chan_sip.c:13753 handle_request_invite: Call from '500' to extension '07050653748' rejected because extension not found.
18:18.21DonAlexGrrrr
18:18.25DonAlex;)
18:19.10DonAlexopppps before you lambast me..
18:19.23DonAlex[Mar 14 18:18:59] NOTICE[464]: chan_sip.c:13753 handle_request_invite: Call from '500' to extension '907050653748' rejected because extension not found.
18:19.36DonAlexdoes the same with a 9 as well.. as per the dial rules.
18:19.40DonAlexHmmms
18:20.24DonAlexwonder if parked calls is to blame it is the only other bloody thing that is included I don't quite understand.
18:20.49robephwhy include it?
18:21.02rkeeneI'm having a weird problem with my voicemail after the most recent crash
18:21.28robephalso yo u even sure you're in the right context?
18:21.37DonAlexrobeph: Dunno.. it did that by default. That's asterisk now for you?
18:21.46robephit did what?
18:21.51DonAlexcustome-numberplan-1 no?
18:21.52rkeeneNo voicemail menus have output ... It *ACTS* like it's working, as far as I can tell.. but I can't hear things like "Enter password"
18:22.18robephDonAlex: yeh yio usure its dialing in that context
18:22.31DonAlexrobeph: I mean   [numberplan-custom-1]
18:22.36DonAlexrobeph: of course
18:22.37robephyes
18:22.38robephok
18:22.47DonAlexrobeph: How do I make sure it is doing that?
18:23.01DonAlexrobeph: I mean it is the Only Dialplan there is?!
18:23.28BobLutzcontext != dialplan
18:23.31robephfrom console try dial 07050653748@numberplan-custom-1
18:23.34robephoh
18:23.36robephdur
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18:23.47robephheh
18:23.51*** part/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
18:24.12robephi guess I dunno then :p
18:24.32DonAlexrobeph: The asterisk console?
18:24.38robephDonAlex: well not from bash
18:24.55bkrusegui questions belong in #asterisk-gui, stop flooding #asterisk because you are not understanding :]
18:25.10DonAlexrob : Hmmmmm "No such command 'dial 07050653748@numberplan-custom-1' (type 'help' for help)"
18:25.30DonAlexrobeph: Or am I being dense and do no need dial ?
18:25.32robephno dial command?
18:25.45DonAlexrobeph: apparently not.
18:25.52robephodd.. dunno the internals well enough to know why you don't have that heh
18:25.58*** join/#asterisk GBR_ (n=gbr@200.103.96.98)
18:26.03bkruseyou do not have a console driver loaded.
18:26.10bkrusealsa/oss
18:26.37DonAlexbkruse: Ahh could be.. but this is and embedded box so not sure how to go about doing that
18:26.53DonAlexXR1000 if anyone is interested ;)
18:26.57bkruseDonAlex: it is it.
18:27.10bkruseDonAlex: What 'embedded box' ?
18:27.23DonAlexbkruse: Xorcom XR1000
18:27.26bkrusean AA50?
18:27.29bkruseoh....no idea then
18:27.32*** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
18:27.43bkrusemy guess is it does not have alsa/oss (that would be normal in an embedded environment)
18:27.49robephthey really should include a console debugging dial app
18:27.53DonAlexbkruse: ugg fortunately not.. ;) Had even more headaches with that one :P
18:28.05robephthat isn't related to sound drivers heh
18:28.15bkruserobeph: nah, you can use originate I believe....
18:28.21robephah?
18:28.24DonAlexrobeph: Well I am guessing they think it will just 'work' and of course they are short on space :)
18:28.35DonAlexrobeph: originate?
18:28.42robephDonAlex: he said it not I
18:28.44bkrusetype 'originate' at your asterisk cli.
18:28.53robephbkruse: I do have dial,  don't have originate =s
18:28.53*** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net)
18:28.56DonAlexAhh yes..
18:28.58robephdoes it take same args?
18:29.02DonAlexthat's there..
18:29.04bkruserobeph: you do not have originate on your asterisk CLI?
18:29.08DonAlexany idea of the synatax?
18:29.31bkruseoriginate iax2/bkruse extension 9037whatever@numberplan-custom-1
18:29.40bkrusetype 'originate' at the CLI! It tells you the syntax!
18:29.44robephhahah
18:29.53robephpbx*CLI> originate
18:29.54robephNo such command 'originate' (type 'help' for help)
18:29.57robephahhh ok =)
18:30.02robephwho does originate belong to
18:30.03DonAlexbkruse: It tells me a lot more than that ;) Hadly a man page though
18:30.18bkruserobeph: not sure
18:30.31robephwhat does it do,  route call between two devices?
18:30.41robephor device / ext
18:30.43DonAlexbkruse: How are extensions specified then ? 500@<ip>
18:31.08bkruse500@context, it has to be on the local box
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18:31.37robephbkruse: what did BobLutz mean that I was wrong in calling that a context
18:32.11DonAlexbkruse: local, what like a FXS port?
18:32.17znoG_is there some way to tell the IP address of a call made by a SIP/IAX user?
18:32.31BobLutzrobeph: I just said a dialplan isnt a context really
18:32.34BobLutza dialplan has a context
18:32.48robephBobLutz: I meant point the ext at the context in his dialplan
18:32.49robephok
18:32.50robeph:p
18:32.54robephie ext@context
18:32.54BobLutzsorry haha
18:33.12robephI was confused I was lik I couldv'e sworn I knew what I meant to say here...
18:33.47BobLutzhappy pi day everyone
18:34.26robephanyhow DonAlex you wanna ' originate proto/device 07050653748@numberplan-custom-1 ' I'm guessing without having the help text for that function
18:34.41robephBobLutz: 3.1415926535898?
18:34.48robephI think thats it anyhwo
18:34.58BobLutz:)
18:35.27robephI have an ex gf who knew it to a disturbing 80 something places,  why I do not know,  it was likely just a display of deeper rooted mathmatical obsessions
18:35.35*** join/#asterisk flush (n=SYN_SENT@ip216-239-88-173.vif.net)
18:35.53DonAlexrats.. being kicked out of the office.. Will be back in a few hours..  Thanks for your help guys.. I will be looking up the originate when I get home.
18:36.02DonAlexLaters all
18:36.32BobLutzhaha wow, 80?
18:36.51robephyeh
18:36.59robeph+ - some
18:37.44robephin truth it was really odd,  cos aside from that lil nuance,  she was a rather normal girl,  worried more of fashion and chick flicks than science et al.
18:37.50robeph:p
18:37.58robeph*shrug*  just one of those oddities..
18:41.06rkeeneHmm, it seems that Playback() no longer works on my Asterisk system.. any ideas ?
18:41.20robephwhen'd it stop working?
18:41.53CrashSysHey strom, is AstDB persistent?
18:42.00rkeeneAfter the asterisk box crashed and came back up
18:42.28BobLutzrkeene: is the module loaded?
18:42.31rkeene(I had restarted the asterisk daemon recently, and all my configuration files are version controlled...)
18:42.53rkeeneBobLutz, I would guess so.. it worked this mornig
18:43.30rkeene(Which module)
18:43.36robephapp_playback2.so
18:43.44BobLutz2?
18:43.53robephI guess,  thats what I use
18:44.00robeph:p
18:44.04BobLutzhaha
18:44.09robephit works
18:44.17robephso.. I mean i dunno if there's another version heh
18:44.19BobLutz1.4.17 --> app_playback.so for me
18:44.31robephheh I'm using 1.2.24
18:44.37rkeeneModule                         Description                              Use Count
18:44.42rkeeneapp_playback.so                Sound File Playback Application          1
18:44.47rkeene(I'm using 1.4.18)
18:45.51*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
18:46.56BobLutzrkeene: Does modules.conf autoload/
18:47.18rkeene[modules]
18:47.18rkeeneautoload=yes
18:47.26BobLutzword...
18:47.41BobLutzNo kind of error?
18:48.55rkeeneNo error that I've been able to find
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18:56.25robephrkeene: not sure if it'll help,  but turn debug on?
18:56.45*** join/#asterisk docelmo (n=vircuser@h74.196.180.12.cable.rstb.cablerocket.net)
18:58.39BobLutzWhat exactly is an ASTOBJ?
19:00.02russellban object management api ...
19:00.57BobLutzReading through the source.. just trying to wrap my head around it
19:01.08russellbwhich has been deprecated in favor of astobj2
19:01.49BobLutzd'oh
19:02.01russellb:)
19:02.09russellbastobj2 is documented pretty well
19:02.33BobLutzIs there better documentation then the doxygen?
19:02.50russellbinclude/asterisk/astobj2.h
19:02.51rkeenerobeph, Seems to be related to the Zaptel module
19:02.59rkeeneBecause I unloaded that, and it is working now
19:03.07russellbthere are some things in the header file that aren't showing up in doxygen for some reason
19:03.08robephah
19:03.11robephstrange
19:03.18BobLutzrussellb: thanks
19:03.27CrashSysall hail the mighty russell
19:04.02russellbBobLutz: no problem
19:04.06russellbCrashSys: greetings sir
19:04.17CrashSysNow I need a beer
19:04.17rkeenePerhaps even the "tor2" module, since I can load many Zaptel modules, but not "tor2"
19:04.26rkeene(Well, I can.. but then playback() quits working)
19:05.17*** join/#asterisk codejunky (i=jan@88.198.12.5)
19:05.24CrashSysCan we write an app_beer module? monitors my stock of stella artois in the fridge, and re-orders if it gets low
19:06.06CrashSysSome sort of predictive ordering algorithm...
19:06.33Qwell~nowwhat
19:06.34jbotSo you just installed Asterisk now what? http://www.youtube.com/watch?v=ULgwbvj768E
19:06.56codejunkyHello, I have connected my asterisk to my voip provider with the sip protocol and connected a phone (hardware) to my asterisk. Now the problem is that if I dial out I do not hear the one I am calling for 5-10 seconds after he answers the call. Any ideas what I can do?
19:08.09*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)
19:08.17*** join/#asterisk nny_1 (n=Scott_My@64.203.239.83)
19:08.36*** join/#asterisk svenna_ (n=svenna@p548D35CF.dip0.t-ipconnect.de)
19:09.52nny_1ok kids.. here's a loaded question.. Does have anyone have an opinion on the price of this system:
19:09.52nny_1170 Snom 300 Phones
19:09.52nny_1Quad T1 Card (Digium)
19:09.52nny_1Server - Dual Core, 4Gb Ram
19:09.52nny_1System load average 20%
19:09.53nny_165,000.00
19:10.15[TK]D-FenderSnom... BLEH
19:10.20CrashSys65K?
19:10.22CrashSys....
19:10.24nny_1yes
19:10.35jasonWootis that Yen?
19:10.40*** join/#asterisk esaym (n=user@72.183.198.134)
19:10.45nny_1[TK]D-Fender: USD
19:10.47nny_1oops mt
19:10.49nny_1USD
19:11.07jasonWoot¥
19:11.09Qwellerr...what's a snom 300 cost?
19:11.12CrashSysSeems like a high price to me...
19:11.23nny_1[TK]D-Fender: heh only for the web interface, easier to test remotely
19:11.27nny_1CrashSys: installed
19:11.36CrashSysIncluding cabling?
19:11.37nny_1CrashSys: basically we are installing it
19:11.45nny_1CrashSys: existing infrastructure
19:11.55[TK]D-Fendernny_1: BLEH <-  And massively over-priced.
19:11.58CrashSysSo your just deploying to the desktop and server rack?
19:12.06nny_1CrashSys: yes
19:12.07bkruseQwell: for some strange reason, I heard the music video from ~nowwhat, but it was not from my pc..... ha ha :]
19:12.16Qwellbkruse: heh
19:12.42[TK]D-Fendernny_1: Snom 300 = $99 USD (retail).  170 x $99 =16830$ USD for phones.  So basically you're selling a server for almost 50K$
19:12.51Qwell...
19:12.58nny_1[TK]D-Fender: 2 hours per phone setup and testing
19:13.04Qwell2 hours?!
19:13.05*** join/#asterisk snapple42 (n=snapple4@h216-18-80-132.gtconnect.net)
19:13.05QwellPER PHONE?
19:13.08Qwellno
19:13.19Qwell5 minutes - tops
19:13.20[TK]D-Fendernny_1: Man I gotta get me some customers like that ;)
19:13.21nny_1Qwell: 1 hour in house testing, physical deployment in room
19:13.24nny_1[TK]D-Fender: hehe
19:13.27CrashSysYou are doing something wrong if a large deployment like that takes you 2 hours per phone set-up/install
19:13.28Qwell...you've gotta be kidding me
19:13.29nny_1it's for a hotel
19:13.30DeeewayneQwell, maybe its mini-mall style testing
19:14.02*** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net)
19:14.17CrashSys65K is high
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19:14.25bkruseQwell: Agreed, we should start a business
19:14.29nny_1Qwell: ha good luck
19:14.51nny_1Qwell: this isn't in HHI, and if you have ever been here, you'd know it's a deal compared to the rest of the industry here
19:15.18Qwellnny_1: feel free to send some of that money to the developers :)
19:15.28nny_1Qwell: heh
19:15.33CrashSysan IPOffice on 8 phone with sip trunks was $10K...
19:15.38Qwellsmall, unmarked bills would be best
19:15.52nny_1CrashSys: yeah our 10 phone system is 6k
19:15.57nny_1installed
19:15.58St1ckm4nwe payed about 15k for our nortel switch that capped out at 25 phones
19:16.05CrashSysnny: Well the definition of good customer service is they are happy when they leave... so if they are tickled pink to pay $65K for it and you can deliver their needs, sell it...
19:16.25nny_1CrashSys: yeah this is all prelim.. we mainly deal with smaller systems atm
19:16.39nny_1we are probably going to knock down the installation time
19:17.00nny_1we have a metric that doesn't scale well, and that's one of the things I am looking at
19:17.16CrashSyscant imagine you'd have more then 40 work hours in laying the system out and setting up provisioning files
19:17.47CrashSysSpecially for a simpler hotel set-up... most PITA thing is billing/wake-up calls
19:17.47nny_1CrashSys: true
19:17.51bkrusebesides, snom's auto provisiong ROCKS
19:17.53nny_1CrashSys: yeah indeed
19:17.56nny_1bkruse: agree
19:17.57*** join/#asterisk quigon (n=matias@190.3.121.15)
19:18.01CrashSysPolycom's...
19:18.03nny_1hey i haven't sent an invoice yet, lol
19:18.06nny_1yeah
19:18.15nny_1polycom is on the slab too, (we normally use them)
19:18.15bkrusenny_1: if you can do it, do it, we are all just jealous
19:18.31CrashSysIf you want to sub it out, let me know :)
19:18.38nny_1:)
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19:18.40bkrusei would say 95% is putting the phones in place, provisioning is 5% with snom's awesome auto provisioning, you just have to be efficient
19:18.47nny_1bkruse: fully agreed
19:18.57nny_1bkruse: i have the ability to do the provisioning
19:19.00Qwellwalk to a room.  plug it in.
19:19.08Qwellthat's...err...not hard
19:19.11nny_1yeah lol
19:19.37bkruseQwell: Like I said, we need to start a business. QwellKruzTeleJoshComVoxMart - Install asterisk and phones, only 70k!
19:19.53CrashSysbkruse: you hiring?
19:20.02nny_1well my main concern is how this stacks against 1.) say someones trixbox deployment (shudder) 2.) a traditional analog system and 3.) nortel/mitel/etc
19:20.19nny_1analog, er --> FXS
19:20.40bkruseCrashSys: of course, it takes a lot of man power for provisioning! or...I could be completely lying. :P
19:20.43nny_1and heh, hell yeah start the damn business, the market is horribly inflated, obviously :)
19:20.53CrashSysI'm good at phone deployment!
19:21.04CrashSysI even make sure the cords dont go under the phone stand!
19:21.06bkrusenny_1: asterisk > nortel/mitel/avaya in analog deployment
19:21.08bkruseCrashSys: hired
19:21.16nny_1bkruse: agreed
19:21.25CrashSyssahweet
19:21.25bkrusenny_1: actually, the market is not, you have to know HOW to market
19:21.26Qwellbut, you probably aren't beating them on price
19:21.33bkruseQwell: never
19:21.40Qwellnot at 65k
19:21.50nny_1right now ALL of our SMB systems kick the shit out of the local market
19:21.52bkrusenortel systems (analog) for that many are under 30k easy
19:21.59nny_1mind you we are also in the process of doing an 800 phone system
19:22.06bkruseSo.....
19:22.10nny_1we even beat the local telco on pricing
19:22.13CrashSysFor a straight analog switch you wont be beating the mitel's...
19:22.22bkruseCrashSys: In that price range, no
19:22.24nny_1er mitel IP rather
19:22.29Qwellexcept that his solution *would* be less than 30k
19:22.38Qwellif he wasn't charging 4x for the phones
19:22.45CrashSysTrue
19:22.59nny_1whos charging 4x per phone?
19:23.02bkruseQwell: Absolutely.
19:23.07nny_1fender hit it on the head
19:23.10nny_1170 per phone
19:23.15nny_17k server
19:23.20bkruseNot to mention that snom does annual pricing agreements, which you should look into....
19:23.26nny_1heh and 34k for install
19:23.39nny_1yeah we get the snoms for less than 99
19:23.41CrashSys7
19:23.49CrashSys7K for a server? You aren't buying a dell are you?
19:23.54bkrusenny_1: exactly. That what 'mass purchasing' is.
19:23.56nny_1no thats retail
19:24.01CrashSysCause that's one way you can guarantee some service calls :D
19:24.03nny_1bkruse: indeed i know this
19:24.08nny_1CrashSys: no dell
19:24.10bkruseCrashSys: Even so, 2950's are ~4k with 4 gigs of ram and rails
19:24.15nny_1CrashSys: and thats not what we pay for those items
19:24.26nny_1CrashSys: we are a business :)
19:24.32bkruseCrashSys, Qwell: so basically he gets even more money in pocket getting them even cheaper
19:24.51nny_1bkruse: precisely
19:24.53CrashSysI'm just joe blow trunk slammer... but i'd ethically have an issue with myself charging $65K for that job...
19:25.07CrashSysPlus at $65K you wont win the bid compared to a TDM Interconnect
19:25.08nny_1CrashSys: you or a company?
19:25.18nny_1CrashSys: no bidding here
19:25.24nny_1CrashSys: we are their provider
19:25.41Qwellso then charge $800k
19:25.45CrashSysIf they'll pay, take 'em...
19:25.45nny_1CrashSys: and remember, this is a discussion, no invoice estimate or anything has left my desk
19:26.04nny_1so you would charge what?
19:26.32nny_1that's why i am having this discussion, i am not out for someone's bloood or to overcharge, however, our 2 hour metric usually includes training, which this doesn;t need
19:26.45nny_1and we are evolving the process
19:27.28bkruseQwell: You know how hard we could hit the market and what an advantage we would have? lol. qwell + bkruse > all
19:27.40nny_1bkruse: what are you waiting for?
19:27.42nny_1bkruse: do it
19:27.47bkrusenny_1: I work for digium.
19:27.49nny_1bkruse: we need more asterisk companies out there
19:27.51nny_1bkruse: lOL
19:28.01CrashSysnny: shooting from the hip i'd ballpark a job like that around 30K
19:28.06nny_1bkruse: well.. then you guys know what the deal is
19:28.06bkruseI will wait for the buyout, then do something like that :P
19:28.12bkruseyes, we do :]
19:28.15bkruseI have a couple things up my sleeve....
19:28.15CrashSyswithout spending 3-4 hours and a site-survey crunching the hard numbers...
19:28.32*** join/#asterisk pa (n=pa@unaffiliated/pa)
19:28.52nny_1bkruse: you guys have siwtchvox systems priced pretty damn close to ours
19:29.08bkruseCrashSys: Honestly, with competing with cisco, I have heard of many cases where the company went with cisco instead of an asterisk install because the price was TO low, we cannot criticize nny_1 for having some of the market figured out.
19:29.24nny_1bkruse: indeed
19:29.36nny_1bkruse: we lost the local township system to Cisco JUST for that reason
19:29.42CrashSysWell, then just say $500/phone
19:30.05CrashSysThat's on par for digital KSU/PBX
19:30.07bkrusenny_1: People will be people. Thinking logically is not always the standpoint to view from. In the marketing biz, perception is your best friend.
19:30.26nny_1FWIW in the precise local market we have one competitor, and we undercut their NOrtel BCM prices by 10% at least
19:30.32bkruseFiguring people out is a gift and a talent that you must possess
19:30.41CrashSysJust hope there's no ESI's or Telrad's up in your neck of the woods
19:30.49keith4_what are my options for boot servers for polycom phones? all examples I see seem to be ftp
19:31.00bkrusekeith4_: how about res_phoneprov.c in trunk?
19:31.03nny_1heh we sell enough, I'll invite you all down.. this market is it's own unique metropolis
19:31.14bkrusekeith4_: You can do it all from the GUI with minimal tweaking.......
19:31.23keith4_GUI?
19:31.24nny_1my business partner is an old Fed/ GIS guy and the statistics here are mind blowing
19:31.28bkrusenny_1: You haven't even seen the market outside the US.
19:31.30bkrusekeith4_: web interface
19:31.32nny_1bkruse: indeed
19:31.44[TK]D-Fenderkeith4_: Go read the admin guide...
19:31.46keith4_bkruse: looking to do provisioning using dhcp
19:32.00nny_1bkruse: mind you, we are looking at *this* one vs* the global market
19:32.12nny_1bkruse: don't get me wrong, 65k was prelim, we hope to get it down to under 50
19:32.15keith4_[TK]D-Fender: polycom's admin guide?
19:32.22[TK]D-Fenderkeith4_: Clearly.
19:32.27keith4_ugh
19:32.42*** join/#asterisk Yourname`` (n=chatzill@unaffiliated/yourname/x-837320)
19:33.02CrashSysnny_1: In the florida area an average of $500/station is a pretty good "dip-stick" to measure an install by...
19:33.04bkrusenny_1: Sure, but I am talking about business outside the US. I do not think you fully realize that people take what is made, and sell it for hundreds of thousands to countries that do not have information like you do. I know people in india that make hundreds of thousands / year because they understand the market and can read english and translate.
19:33.05CrashSysunless your cisco...
19:33.06keith4_I started to, but it's full of so much useless crap
19:33.12nny_1CrashSys: thank you
19:33.51*** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net)
19:33.53CrashSysBut for $500/station, it's a full-features KSU/PBX from Telrad or ESI or Samsung, etc...
19:34.09CrashSysHotel systems are a lot cheaper... almost half usually
19:34.23CrashSysBecause it's all SLT's and just a billing modules/wake-up call
19:34.31nny_1bkruse: I understand, thank you... i hope to learn that first hand soon :)
19:34.44*** join/#asterisk bkruse (n=bkruse@216.207.245.1)
19:34.44*** mode/#asterisk [+o bkruse] by ChanServ
19:34.48bkrusesorry about that
19:34.55CrashSysMitel's big into hotel systems...
19:35.29ZPerteeanyone use ipcomms free DID service?
19:36.10[TK]D-Fenderkeith4_: Its the 4th chapter, and blatantly obvious.  Seek therapy...
19:36.29bkruse[TK]D-Fender: haha, try not to be too harsh, but I completely agree.
19:36.36keith4_[TK]D-Fender: if you don't know the answer, just say so :-P
19:36.42nny_1CrashSys: bkruse etc .. going to rewrite this, let you know what craooens
19:36.44nny_1crappens*
19:36.49keith4_I didn't say "where can I learn for myself how to provision polycoms" ;-)
19:37.08nny_1and thank you all for the input, I appreciate the discussion
19:37.25[TK]D-Fenderkeith4_: And no, I will not be so easily goaded into satisfying your sloth-like tendencies :p
19:37.33keith4_awwww
19:37.37[TK]D-Fenderpwned
19:37.38bkrusenny_1: The global market. For example, I know of an asterisk installer in brazil that does outrageous installs, you have to think outside the box of the US government and regulations and think from their perspective, where no one knows asterisk, and pbx is a monopoly, one that can be easily taken over
19:37.52keith4_all I have is this lousy PDF that doesn't have an index in the sidebar
19:37.53bkrusejbot: [TK]D-Fender++
19:39.37keith4_[TK]D-Fender: FYI, it's chapter 3, not 5
19:39.39keith4_er, not 4
19:40.21nny_1bkruse: heh i recently dealt with an install in Panama. The company used trixbox on two boxes (IAX2 to US). The boxes were way subpar (VIA Chipset, Harddrives on static wrap, unmounted) MiniITX systems. 8 or so Snom Phones, $10,000. We ended up wiping Trixbox, setting up proper dialplans from scratch. The hardest part wasn't the price, they had NO recourse for getting some of their money back or the situation resolved before we stepped in
19:40.45bkrusenny_1: My point exactly
19:40.54*** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net)
19:41.02nny_1oh and the US box is no throwing hard drive errors, so I had to give him a quote on preplacing it, as the box itself is basically trash
19:41.07nny_1now*
19:41.43[TK]D-Fenderkeith4_: Depends which version you're looking at, and it sure appears you found it jsut fine...
19:41.46bkrusenny_1: dell 2950's are your friend
19:41.54znoG_sorry to repeat, i just find it hard to believe no-one knows how to do this :) is there some way to tell the IP address of a call made by a SIP/IAX user?
19:42.08keith4_[TK]D-Fender: yah, but now all I have is lots of reading to do, and no easy answers
19:42.14bkruseznoG_: sip debug
19:42.24bkruseIt might even be in cdr-csv
19:43.06[TK]D-Fenderkeith4_: its in bllody big print as to what protocols is supports for the boot server.  Get some new eyes... this is just sad...
19:43.14[TK]D-Fenderashdsdklsd
19:43.15keith4_it's aggravating that polycom abbreviates "Soundpoing IP" as "SIP" everywhere
19:43.44nny_1bkruse: yeah we are a growing company, we aren't trying to rape the market, or inflate it anymore than it already is... My business partner and I had a discussion on how we could lower the pricing on this quote, and what it looked like vs. other quotes we have seen. We came to two conclusions, one is that our quote could* be lowered, although we have played that game before and lost due to the lower rate exhibiting the product as "cheap" and 2.) we are al
19:43.45keith4_oh... yah I found the tftp stuff a while ago
19:43.50lirakiswhen you install a new kernel, do you just need to recompile zaptel stuff? or * too
19:44.10keith4_just zaptel
19:44.14nny_1bkruse: we are looking at dells for the HA setup (800 phones) here as well
19:44.22nny_1bkruse: heheh well
19:44.24bkruse:]
19:44.30bkruseYa, you got it down.
19:44.34*** join/#asterisk quigon (n=matias@190.3.121.15)
19:44.41nny_1bkruse: thanks..
19:44.44bkruseExactly, lower rate = cheap
19:44.56bkruseyou completely understand what 75% of people in the market do not
19:45.02bkrusethe other 25% are making $$$, and lots of it.
19:45.12jasonWootkeith if you figure out how to get rid of softkeys on the polycoms, let me know... going on 6 months now
19:45.15nny_1bkruse: agree
19:45.36nny_1brb
19:45.56keith4_jasonWoot: sure, as soon as I figure out what a softkey is
19:46.20BobLutzIs there some kind of naming convention for Asterisk code? (AST_DECLARE_APP_ARGS(), AST_APP_ARG() --> are these in the same source file?)
19:46.50[TK]D-FenderjasonWoot: Like which?
19:47.41bkrusenaming convention and what file they reside in are two totally different question
19:47.45bkruseBobLutz: ever heard of ctags? it rocks.
19:47.56BobLutzctags?
19:48.26bkruse!google vim + ctags
19:49.00BobLutzbkruse: I see the vim ctags, but I know not how to use vim :-/ sucks to be me i guess
19:49.17BobLutzerr ctags in the source code... I didnt know what they wre called
19:49.37CrashSysHmmm
19:50.05bkruseBobLutz: when you find someone you want to the find the function/origin you can use ctrl+]
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19:51.03BobLutzbkruse: lol that is so useful!
19:51.15bkruseBobLutz: you have no idea.
19:51.17BobLutzlol
19:51.22BobLutzi really dont..
19:51.22bkruse:]
19:51.31bkruseespecially in big projects like asterisk
19:51.52keith4_can I reboot a polycom remotely?
19:52.02bkrusekeith4_: web interface
19:52.07keith4_tried that
19:52.12keith4_web interface sucks
19:52.13bkruseactually, just make a change in the web interface in the admin panel, it has to reboot
19:52.17CrashSyskeith: if there has been a provisioning change, you can use sip notify... otherwise web interface
19:52.18bkruseno it does not, you suck.
19:52.19keith4_oh duh
19:52.24keith4_no, it sucks
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19:52.34bkrusekeith4_: or, you cannot use the web interface and need to admit it
19:52.44*** join/#asterisk retloc (n=retloc@199-117-163-66.dia.static.qwest.net)
19:52.45znoG_bkruse: wouldn't sip debug only apply to SIP clients?
19:52.53[TK]D-Fender"Denial, its not just a river in Egypt"
19:52.55keith4_i prefer the snom web interface to the polycoms
19:53.11nny_1keith4_: the snom's and the polycoms are night and day as far as phones go
19:53.12retlocMy extensions.conf keeps clearing itself after every edit. Permissions are 777 and I am using nano for editing.
19:53.16[TK]D-Fenderkeith4_: Shouldn't be touching the web interface on a Polycom EVER anyways.
19:53.17retlocAny thoughts on this
19:53.27bkruseznoG_: oh, I thought you said sip
19:53.27nny_1[TK]D-Fender: heh agreed
19:53.30keith4_[TK]D-Fender: how else would you recommend rebooting a phone that's several miles away?
19:53.43[TK]D-Fenderkeith4_: the WIKI is your friend.
19:53.44keith4_nny_1: agreed
19:53.46znoG_bkruse: i need to find out the IP of the user in either SIP or IAX (depending on how they logged in)
19:53.50keith4_the wiki is a mess
19:54.10CrashSysKeith: make a silly change in a polycom provisioning file, like changing registration number 8's port to 5061... it'll reboot
19:54.15keith4_as wiki's tend to be
19:54.24retlocEven if I copy extensions.conf off of one of my identical working dialers it still clears the file completely
19:54.29[TK]D-Fenderkeith4_: http://www.voip-info.org/wiki-Polycom+reboot+hardphone+script
19:54.35jasonWootpolycom won't reboot via web interface if ext is in use
19:54.47[TK]D-Fenderkeith4_: Like a  second google search on it doesn't turn it up as the FIRST result, nice and blatant.
19:55.31nny_1heheh damn telemarketers.. "Can I speak to the decision maker?" (Accented voice)... who the hell responds to that??
19:55.33keith4_I need 130 lines of perl to reboot a phone? not acceptable
19:55.43nny_1I need to instill telemarketer hell as defined in the wiki
19:56.11[TK]D-Fenderkeith4_: You have a very serious reading dysfunction....
19:56.19retlocFor the solutions it was found
19:56.27retlocMY HDD was ful
19:56.32retlocClearing recordings
19:56.40*** join/#asterisk FuriousGeorge (n=brian@ool-4354d18c.dyn.optonline.net)
19:56.46FuriousGeorgehey all
19:56.55nny_1retloc: was that a haiku attempt :)
19:57.17keith4_[TK]D-Fender: all of these methods seem to use the fact that the phone is periodically checking its config file for updates.... i need to reboot the phone to have it re-dhcp to tell it which tftp server to use... so that's not very helpful, now is it?
19:58.08[TK]D-Fenderkeith4_: Keep reading, and it might all sink in...
19:58.23FuriousGeorgei was just thinking:  if there were some elegant way to automatically check in on listening channels that have been open for 5 hours or more, somehow determine if there is activity, and if not perform a soft hangup, i could probably go without rebooting asterisk daily
19:59.13FuriousGeorgeits not a bid deal, but it kinda bothers me.  i dont restart postfix or apache daily
20:01.15FuriousGeorgeand it rarely just crashes and dumps a core, that i could file a bug report wity
20:01.17FuriousGeorge*with
20:01.37nny_1something to consider on dealing with pricing vs. cost vs. competition too is what you can do to increase the price with VAR based things, extended warranties, 24 hours support, etc... You can charge X for the system on a base level, but add the things that you have to address in the market.
20:02.09keith4_damn
20:02.13keith4_i'm such an asshole
20:02.15nny_1the reason a cisco phone system is high dollars is because you can call Cisco at 4 am, and tell them what you had for lunch, and for that price they'll listen
20:02.45FuriousGeorgenny_1: is that to me?
20:02.57nny_1FuriousGeorge: heh no, part of a prior discussion
20:03.06FuriousGeorgeheh
20:03.08nny_1FuriousGeorge: what kind of system are you dealing with
20:03.10nny_1?
20:03.38nny_1phones/ channels/ version of asterisk? I haven't seen the issues you mentioned, so I am curious (furious?) as to what is causing your specific reboot needs
20:04.03FuriousGeorgenny_1: one sec, pls...  darned clients always interrupt
20:05.34keith4_[TK]D-Fender: why can't you just say "sip notify" instead of letting me ramble on like an idiot for 10 minutes?
20:05.34nny_1heh
20:05.41[TK]D-Fenderkeith4_: What and prevent this ultimate reality from smacking you upside the head?  Never! ;)
20:06.03keith4_in my defense, you sent me to a wiki page called "reboot hardphone SCRIPT"
20:06.53[TK]D-Fenderkeith4_: Sorry... doesn't hold water.  You're every attempt at reading anything for information shows the same pattern.  Thime to correct the problem, not the symptom.
20:06.53*** join/#asterisk cardiff (n=cardiff@76-10-153-160.dsl.teksavvy.com)
20:07.04[TK]D-Fenderyour*
20:07.04keith4_lol
20:07.09[TK]D-Fendertime*
20:07.13keith4_the solution would be a cup of coffee
20:07.17[TK]D-FenderWOW.... I'm f-ing fried today...
20:07.26nny_1it is Friday
20:07.29keith4_but it's nearly 5 here, so it's not going to happen
20:08.20nny_1beer it is
20:08.42keith4_no
20:08.46keith4_beer is what started this problem
20:08.50nny_1lol
20:08.51keith4_2 of them, at lunch, to be specific
20:09.16keith4_[TK]D-Fender: i'm not usually this dumb, but i've been drinking at work today
20:13.43*** join/#asterisk xenonex (n=xenonex@82.200.211.5)
20:16.26bkruseeither you work rocks, or it sucks horribly
20:16.57*** part/#asterisk cardiff (n=cardiff@76-10-153-160.dsl.teksavvy.com)
20:17.15*** join/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net)
20:17.28ThatKidKelanyone know where one can buy or download a quality NPA NXX list?
20:19.06_ShrikEThatKidKel: nanpa
20:19.10keith4_bkruse: no specific policy about drinking during lunch
20:19.19keith4_and the boss is out today... so....
20:19.28ThatKidKel_ShrikE.. I'm all over there site, and can't find it
20:19.28nny_1bkruse: btw have you heard of any changes in the Digium support offerings? We are under the understanding that you can buy 24/7 support, but only with AstBiz... which I have nothing against, except there are quite a few situations where we would rather use our own base distro
20:19.43ThatKidKel_ShrikE..  Only Area Codes
20:19.55bkrusenny_1: AstBiz? You can purchase 24/7 support for any particular product
20:20.20bkrusenny_1: http://www.digium.com/en/services/maintenance.php
20:20.51nny_1bkruse: yeah they update that, but AFAIK for config support, you have to use AstBiz
20:21.08_ShrikEThatKidKel: you can try maponics
20:21.29BobLutzThis Asterisk programming is mad hard
20:21.40nny_1BobLutz: heh is that sarcasm?
20:21.52BobLutzno
20:22.06BobLutzstruct stuff *ps        !#$#
20:22.09bkruseBobLutz: Programming asterisk? It is if you do not know how to program, or the layout and some of the API's in asterisk
20:22.22QwellBobLutz: what, you've never used structs and pointers?
20:22.23bkrusebut talk about open source, you have all the examples you could ever want!
20:22.32bkruseQwell: That is why programming asterisk is hard :]
20:22.37BobLutzI know a little bit of C
20:22.40*** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net)
20:22.44BobLutzIts kinda weird going from OOP java to C
20:22.47bkruseBobLutz: would it help if I rewrite asterisk in visual basic?
20:22.52BobLutzLOL
20:23.02bkruse:]
20:23.04nny_1bkruse: lol actually, I would prefer it in the original BASIC
20:23.06BobLutzIm a fast learner...I didnt even know what a keyboard was a couple years ago
20:23.07russellbi'm almost done with my bash rewrite
20:23.17bkruserussellb: good good
20:23.47BobLutzapp_read.c is extremely helpful though, if you guys see Mark, tell him I said great job
20:24.25*** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku)
20:24.26russellbtry channels/chan_zap.c
20:24.29russellbthat's the best place to start
20:24.44russellb*** KIDDING ***
20:24.58BobLutzLOL, i was opening it up in my text editor
20:25.04*** join/#asterisk harryv (n=fufufu@0x55508034.adsl.cybercity.dk)
20:25.39russellbnah, apps/ and funcs/ are the best place to start in general ...
20:25.49BobLutzIf I can get this module to do what I want, I will probably try to do some janitor work in my free time
20:25.49russellbsome of them, anyway
20:25.50nny_1bkruse: long story short, and openly stating the fact that additional fees are expected, we want to be able to tell our customers that on top of our already **stellar** support we offer, they can contact digium directly after we have all fled to Mexico and still have a support channel. Shit right now I already have the framework, site and software setup for a proper support channel, I just hate to reinvent such an already prefectly round whee;
20:25.52anonymouz666I wonder if someone really understand chan_zap.c :)
20:25.53nny_1wheel*
20:25.55russellbapp_queue, app_dial, for example, are more complicated
20:26.08BobLutzapp_read is nice...barely at 200 lines
20:27.07Qwell"if you guys see Mark, tell him I said great job"
20:27.26QwellI wonder how much of his code is still left
20:27.30russellbheh
20:27.35*** join/#asterisk JayTee52 (n=jforde@207-67-84-188.static.twtelecom.net)
20:27.49Qwellprobably a crapton
20:27.54russellbprobably
20:27.56BobLutzthe only file i havent seen his name on is astobj2.h
20:28.02russellbhe's the last person who changed 23% of the lines of code
20:28.09Qwellrussellb: damn
20:28.10russellbkpfleming is the last person who changed the most, at 26%
20:28.12anonymouz666Qwell: did he still program?
20:28.20Qwellanonymouz666: occasionally..
20:28.27russellbhttp://bugs.digium.com/svnstats/asterisk/trunk/
20:28.43Qwell1.8%? O.o
20:28.52BobLutzlol
20:28.53Qwell+ 0.1%.  w00t
20:28.59russellbha
20:29.28QwellMarko?
20:29.29Juggie0%, woot :)
20:29.33russellbno, me, heh
20:29.36Qwelloh
20:29.58Qwellrussellb: I'm sure it would be much different in 1.0
20:30.01russellbi wonder if the svn mv to create the main/ dir got kpfleming some stats :-p
20:30.12Qwellhmm, not sure how svn handles moves
20:30.17russellbno idea
20:30.31Juggieif Qwell would stop commiting my patches i'd have 2 :p
20:30.47Qwellwmeadows has 0 lines per change
20:30.51Qwellwtf did he change?
20:31.13bkrusenny_1: Then contact sales@digium.com and strike a support deal!
20:31.23nny_1bkruse: good call, will do :)
20:31.46bkrusenny_1: That, and you can probably get around it by purchasing our hardware (which you will probably have to do) and getting support through those
20:31.48Qwellwow, I did not expect that at all.  33% of my changes are in main/, and 27.8% in channels/
20:31.52Qwell...7% in apps
20:32.10bkrusebut MANY companies have deals to where they have 24/7 support with digium that they offer when they cannot figure things out.
20:32.22nny_1bkruse: by hardware do you mean the card or the entire PBX?
20:32.23russellbi have made over 150 commits between 2 and 3 AM
20:32.25russellbO.O
20:32.32bkrusenny_1: cards
20:32.37bkruserussellb: I do not doubt that at all
20:32.39nny_1bkruse: cool that we already do
20:33.12keith4_do I really have to "contact my polycom reseller" to get the damn firmware?
20:33.46hmmhesaysor google
20:33.53hmmhesayswhich firmware are you looking for?
20:34.06keith4_SIP 3.0
20:34.15keith4_also, the "distribution zip" file
20:34.18keith4_for the boot server
20:34.39keith4_the SIP 3.0 administrator's guide says "Copy all files from the distribution zip file to the phone home directory."
20:34.47keith4_like I'm supposed to a) know what that file is or b) have it
20:35.41hmmhesaysI would guess that is the firwmware file
20:35.51keith4_me too
20:36.02keith4_but to login to the polycom portal, you apparently have to be a reseller partner
20:36.08keith4_which is idiotic
20:36.41keith4_I even tried to open an account, and one of the options under "what are you?" is "phone owner"... and then you can login, but can't download anything
20:36.43keith4_fucking polycom
20:37.01*** join/#asterisk EvilDeshi (n=Skunk@75-135-93-93.dhcp.mdsn.wi.charter.com)
20:37.17EvilDeshianyone around that can help me resolve this issue I am having with realtime using the odbc handler?
20:37.36Qwellkeith4_: so ask your reseller?
20:37.49keith4_I did. they haven't gotten back to me
20:37.57keith4_telephonydepot
20:38.21Yourname``So if manager dialing is crappy and slows everything down like asterisk taking up 100% CPU, loads going upto 70... what am I doing wrong?
20:42.40nny_1good question, is there a way to break down app cpu usage?
20:43.02FuriousGeorgenny_1: im bacm
20:43.07FuriousGeorgeand back too
20:43.17BobLutznny_1: `top` ?
20:43.37EvilDeshido I have to use odbcinst to get odbc to work with mysql and realtime?
20:43.55nny_1BobLutz: lol
20:44.08nny_1BobLutz: well.. yeah.. i knew that :)
20:44.13BobLutzlol
20:45.38FuriousGeorgei have two 'medium volume servers' on running 1.2.twenty-something, thats been around for four years, another one on 1.4.18...  today, they both use tyan motherboards w/ opterons on nforce chipsets, and sangoma cards, but in the past ive used tdm400p
20:45.43FuriousGeorgei use snom phones
20:45.53FuriousGeorgethe 1.2 box tends to have internal calls lock
20:46.01nny_1BobLutz: it may be the crack i just smoked, but doesn't top *just* show asterisk usage? I was thinking something like "app_que" is using X resources
20:46.04FuriousGeorgethe 1.4 tends to have inbound pots calls lock
20:46.30BobLutz`man top` ?
20:46.37nny_1BobLutz: indeed
20:46.39BobLutzlol
20:46.39FuriousGeorgethe former is sip<->sip, the latter is pots->sip.  if i reboot them nightly they will be good for 6 months
20:46.52FuriousGeorgeif i dont they will start experiencing hung channels after a few days
20:47.22nny_1BobLutz: looks like i have some reading to do, my top-fu is only basic
20:47.28keith4_htop
20:47.53FuriousGeorgeerr, i meant to say 'if i restart * nightly'
20:48.52bkruseFuriousGeorge: Bug reports means bugs get fixed (with proper information)
20:49.04bkrusesaying "zaptel doesn't work" does not help....
20:50.04FuriousGeorgebkruse: i started out by saying, before i got interrupted, that im not sure how to bug report these.  there is no core dump
20:50.13*** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net)
20:50.22FuriousGeorgeand when it happens i just quickly softhangup the channels and cron asterisk to reboot at night
20:50.48FuriousGeorgei keep saying reboot instead of restart, but you get the idea
20:51.44nny_1time to flee, later all
20:51.47FuriousGeorgelater
20:51.56*** part/#asterisk nny_1 (n=Scott_My@64.203.239.83)
20:52.18bkruseFuriousGeorge: Console output or anything?
20:53.57FuriousGeorgebkruse: just the occasional 'maximum retries exceeded' but those always tend to happen.  I must confess i dont grasp the lingo of the CLI, but i don't see anything that jumps out at me.
20:54.10*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
20:56.24FuriousGeorgechan_sip.c:1944 retrans_pkt: Maximum retries exceeded on transmission 3c42
20:56.53filethat's bad... it means chan_sip sent out a packet somewhere but got no response
20:58.14FuriousGeorgefile: this is a pretty basic setup, but they do some volume in calling.  any practical cause for that?  bad wiring?  bad phones? all of the above?
20:58.32*** join/#asterisk ZX81 (n=ZX81@202.20.97.211)
20:58.35BobLutzits 5
20:58.40bkrusefile: Right, possibly a slow dns response?
20:58.48bkruseOr his other endpoint is just disappearing?
20:58.50filepossibly configuration... possibly network... possibly routing...
20:58.57filewithout a sip debug it is hard to say
20:58.58bkruseI wonder how much packetloss he has to that host
20:59.13BobLutzfile: russellb: Qwell: bkruse: Thanks for the direction with the Asterisk code
20:59.15*** part/#asterisk BobLutz (n=miles@d60-65-93-136.col.wideopenwest.com)
20:59.20bkruselol
20:59.43russellbfunc_shell is where it's at
21:00.16bkruserussellb: Totally. It is a good answer to "Wtf struct iax2_user *iax2_user, I do not have *'s in java!"
21:00.34Qwellmultiplying a struct times an iax2_user?!
21:00.35bkruseBobLutz messaged me that :]
21:00.36Qwellyou're mad!
21:00.41fileFuriousGeorge: but it is not normal to get those...
21:01.18FuriousGeorgefile: i can prevent it by rebooting asterisk
21:01.20russellbbkruse: lol!
21:01.21FuriousGeorgenightly
21:01.29bkruseQwell: Exactly! a "structure" and "pointer" um, did you include the math library import java.classes.util.io.math.asterisk.string.res.channels.math.Properties.1948.newest.class.math ?
21:01.30russellbpointers are such silly
21:02.49FuriousGeorgei notice if i use answer-after: 0 with my snom phones i can reliably hang a sip channel using 1.2
21:03.35russellb1.2?
21:03.38russellbhmmm ... oh yeah!
21:03.41russellbi remember 1.2 ...
21:03.47Qwell1.2 what?
21:03.48FuriousGeorge1.2.20
21:03.49russellbbut then i moved on like 2 years ago
21:03.56*** join/#asterisk darius_ (n=darius@humility.bourg.net)
21:04.05filemy apartment PBX runs 1.6.0-beta4
21:04.11russellb63 changes to asterisk 1.2 since 1.2.20
21:04.19darius_Who's the iax supporting PSTN terminating voip provider of choice these days?
21:04.42FuriousGeorgei suppose i should upgrade
21:04.58filerussellb: SHARK!
21:05.21russellbOMFG NO!
21:05.23bkruseFuriousGeorge: http://www.voip-info.org/tiki-index.php?page=Asterisk+v1.2
21:05.42bkrusenotice one of the first lines "On Nov. 15, 2005 Asterisk 1.2.0 finally saw the light of day!"
21:05.49bkrusetoday is March 14, 2008.
21:06.07FuriousGeorgebkruse: yeah, i know development stopped, but the 1.4.18 that i run, when its channels lock, its pots and sip
21:06.28*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
21:06.46*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
21:07.04FuriousGeorgeso im scared if i upgrade, since the servers are almost identical, that i will get the worse version of the problem...  i guess it doesnt matter since i restart daily, but i feel like that is just giving up on finding the nature of the issue
21:11.49*** join/#asterisk plasmid (n=noway@c-76-124-171-163.hsd1.pa.comcast.net)
21:13.03fujinanyone know if it's possible to do a silent macro/gosub yet?
21:13.13fujinI'm using a macro for a local channel which is really really noisy
21:14.12plasmidI am having difficulty with my P2PT sipura box not seeing my pbx. I get this on the registration bit: Registration State:Can't connect to login server. I did a vi /etc/hosts.allow and my P2PT of IP 192.1681.107 is allowed. What the devil am I misisng here? Ports 5060-5061,10001~10021 UDP open on router.
21:15.12plasmidi changed the registraion details to that of the provider and it registers fine... but as soon as i change the proxy to my internal pbx (192.168.1.105) I get the above error. What gives?
21:16.20EvilDeshianyone know how i can fix this issue [unixODBC][Driver Manager]Data source name not found, and no default driver specified?
21:16.37fujinconfigure odbc properly
21:16.45EvilDeshiI am not sure how
21:17.00*** join/#asterisk RobH (n=RobH@216.207.245.1)
21:20.14*** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com)
21:21.51*** join/#asterisk rmayorga_ (n=rmayorga@unaffiliated/rmayorga)
21:23.12*** join/#asterisk zobia (n=laurashr@222.212.66.128)
21:23.20zobiaHello everyoen
21:23.23zobiaeveryone
21:23.58zobiaanyone knows where there's zaptel 1.4.8 or 1.4.9 rpm for centos 4?
21:24.11zobiai search it for long time . still no luck/
21:24.39zobiaand if anyone can share the zaptel's .spec for 1.4.8 or 1.4.9 i am so appreciate.
21:24.39_ShrikEzobia: its not that hard to build it from source
21:25.27zobia_ShrikE: i use rpmbuild and checkinstall to build it . both failed. please help
21:26.11zobia_ShrikE: i am not good at the .spec file making. if you have any idea please let me know. thanks.
21:26.44fujinwhat do you need an rpm for? just install from source ;>
21:27.22zobiafujin: i need to rpm to make a autointall cd for zaptel and asterisk kickstart cd.
21:28.03fujinew
21:28.20*** join/#asterisk wordzilla (n=me@d58-106-139-71.sbr4.nsw.optusnet.com.au)
21:29.54*** join/#asterisk CVirus (n=GoD@196.205.192.125)
21:36.14zobiano one knows? or someone can tell me how to make or .spec or find a .spec for zaptel 1.4.8 or 1.4.9?
21:41.06*** join/#asterisk esaym (n=user@cpe-72-183-198-134.satx.res.rr.com)
21:45.53*** join/#asterisk simNIX (n=user@82-204-21-111.dsl.bbeyond.nl)
21:46.02*** join/#asterisk glaz (i=strke@glaciuz.com)
21:46.29simNIXgreetings
21:46.38glazHi
21:46.50glazIs it possible to convert a Cisco 7970 Phone to SIP?
21:47.03glazNot sure I am asking the right question at the right place
21:47.14_ShrikEglaz: the answer is yes
21:47.47glaz_ShrikE: thanks, I can't find any docs on this, but I guess if you say yes I'll find a way to do it.
21:48.43simNIXanone perhaps know url on how to setup Asterisk + spa2100 ? (I asume I dont need zaptel ?)
21:48.50_ShrikEglaz: cisco intentionally does not provide much information on that phone
21:48.58glaz_ShrikE: why is that?
21:49.36_ShrikEI guess they dont want it to be very easy to get it working on other platforms
21:50.03glazyou did it?
21:50.05_ShrikEbut I have one that does work with * and sip.
21:51.03glazok, I read that some people made it working with chan_sccp
21:51.30_ShrikEWith that phone, sccp is easier than sip.  IMHO
21:51.37[TK]D-FendersimNIX, start with :
21:51.38[TK]D-Fender~book
21:51.39jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
21:51.40[TK]D-Fender^^^^
21:52.07[TK]D-FendersimNIX, And as for the SPA, there are dozens of quicky guides for devices like that.  And Indeed you do not require Zaptel for just * + that ATA
21:52.37[TK]D-FendersimNIX, You need Zaptel for TDM hardware, MeetMe Conferences, and IAX2 Trunking Mode.
21:53.10glaz_ShrikE: I guess this is what I need: http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/7970g_7971g-ge/english/5_0/sip/english/administration/guide/70sipag.pdf
21:54.11high-rez<PROTECTED>
21:54.14high-rezerps
21:54.33*** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net)
21:54.49CrazyTuxHey guys has anyone intergrated an AGI script that simply connects to an HTTP (Server) For CNAM,
21:55.01CrazyTuxThat they wouldnt mind emailing me, or helping me out :)
21:55.21CrazyTuxI looked into FastAGI -- Perl, etc, but it does not seem to be to complete.
21:57.03glaz_ShrikE: how did you do it? quickly
21:58.40simNIXFender Ty
21:59.15_ShrikEglaz: i'm not sure that can be done "quickly" :)
21:59.20glazwhen they say United IP phone, are they meaning SIP ?
21:59.37_ShrikEglaz: lemme see if I can find my old configs
22:00.00glaz_ShrikE: great!
22:01.19*** join/#asterisk EmleyMoor (i=phil@topdeck.tinsleyviaduct.com)
22:01.39EmleyMoorAre there known problems with using Festival with cache?
22:02.19EmleyMoorI ask because I have it on and my FXO had a stuck call on it - stuck in a Festival call
22:02.41EmleyMoorOnly noticed because a call that would normally have gone over the FXO went over IAX instead
22:02.59*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
22:03.48EmleyMoorI did read it may be a problem - but short of conducting some more testing, I will be unable to tell
22:04.52EmleyMoorWhat the asterisk console logs - is it kept anywhere?
22:08.03glaz_ShrikE: any luck?
22:09.24*** join/#asterisk BobLutz (n=stansmit@d60-65-93-136.col.wideopenwest.com)
22:10.00plasmidwhen using a regular phone with a sipura box, is there a way to put the caller on hold? I dont' have one of those voip phones yet.
22:11.57*** join/#asterisk russellb (n=mobile@asterisk/developer-and-stable-maintainer/drumkilla)
22:11.58*** mode/#asterisk [+o russellb] by ChanServ
22:25.33[TK]D-Fenderplasmid, yes, go read the users guide to see how.  Its a * code.
22:25.43[TK]D-Fenderplasmid, [flash] + feature code
22:26.03plasmid[TK]D-Fender, i've been trying to find this guide... lol.. because that's not the only feature i would like to use. I also want to transfer, etc...
22:26.13*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
22:26.35plasmidfreepbx.org documentation doesn't show the codes... is there a wiki?
22:27.15[TK]D-Fenderplasmid, its not FreePBX's job to document other hardware...
22:27.41[TK]D-Fenderplasmid, That's like asking Ford for a 1989 Tercel owner's guide...
22:28.07plasmid[TK]D-Fender, ok. Is there a google garage sale for these unsupported codes?
22:28.17[TK]D-Fenderplasmid, And no, Sipura/Linksys does not have a WIKI
22:28.34plasmidso in other words, regular phones cannot put the caller on hold.
22:28.39[TK]D-Fenderplasmid, http://www.google.ca/search?hl=en&q=Sipura+ATA+users+guide&btnG=Google+Search&meta=
22:28.54*** join/#asterisk jay21d (n=jjohnson@pool-71-180-24-188.tampfl.fios.verizon.net)
22:29.37plasmidhmm.. sounds like I have to dwell with PAP2T config codes.
22:29.44plasmid[TK]D-Fender, thanx for the info.
22:30.37[TK]D-Fenderplasmid, And amazingly its onlyt he first link :)
22:30.46[TK]D-Fenderplasmid, You should try a little harder...
22:31.14plasmid[TK]D-Fender, i was trying harder but the other way. Thinking it was a pbx documentation.
22:31.16jay21dPocket Talk is a good SIP softphone that runs on Windows mobile
22:31.49[TK]D-Fenderplasmid, Devices tend to have their own way of doing things.  Read your device's manual first
22:33.47plasmid[TK]D-Fender, something aside... IS there a way to improve the quality of the calls? I heard that u can use a different codec at the expense of more CPU usage?
22:34.09[TK]D-Fenderplasmid, if you're local to the server you should be using G.711
22:34.10plasmid[TK]D-Fender, kinda newbish at this whole VOIP.
22:34.41plasmid[TK]D-Fender, and also Qos I take it. My router is crappy though at handling VOIP calls. Time for a new router I think.
22:35.36plasmidG.711a or G.71u?
22:35.44plasmid*G.711u
22:36.07*** join/#asterisk xenonex (n=xenonex@82.200.211.5)
22:37.15plasmidCodec G.711a is used within Australia and Europe, while G.711u is used within US. nv <--
22:39.33*** join/#asterisk Dexter_81 (n=Dexter_8@host231-112-dynamic.3-87-r.retail.telecomitalia.it)
22:39.46Dexter_81hi i'm italian
22:40.16*** join/#asterisk russellb (n=mobile@asterisk/developer-and-stable-maintainer/drumkilla)
22:40.17*** mode/#asterisk [+o russellb] by ChanServ
22:41.16Dexter_81hi i'm italian, how to i can called a number using spoofing?
22:41.59[TK]D-Fenderplasmid, Either depending on what the other side of most calls will be using.
22:42.38plasmid[TK]D-Fender, my PAP2T 26 pg guide does not mention a single tidbit on these codes u mentioned.
22:42.40*** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk)
22:43.15plasmid[TK]D-Fender, i do remember that to call someone on the network I have to put the extension and then #.
22:43.59*** join/#asterisk paci` (n=paci@cpe-066-057-116-114.nc.res.rr.com)
22:44.03paci`hey you guys around
22:45.27[TK]D-Fenderplasmid, notmal people don't use residential devices like that for business-like functions like transfer/hold
22:45.42paci`how would i go about configuring asterisk to use skype
22:46.20plasmid[TK]D-Fender, agreed.
22:46.38plasmidack.. skype.... they privy into your conversations.
22:46.44plasmidread the small print.
22:46.45[TK]D-Fender~skype
22:46.47jbot[~skype] Skype is a free VoIP software and service using a closed client and propritary protocol.  Only commercial channel drivers exist, all of which are poor implementations of convoluted hacks. In general, forget about using Skype with Asterisk...
22:47.25paci`aw.
22:47.35paci`whats a good VoIP service to use with asteris
22:47.43[TK]D-Fender~itsp
22:47.44jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
22:47.45paci`asterisk*
22:48.00paci`~itsplist-us
22:48.00jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, or http://www.jnctn.com, or http://www.bandwidth.com, or http://vitelity.net
22:51.36paci`[TK]D-Fender, if you odn
22:51.37paci`er
22:51.48paci`don't mind the highlight*, do you know any ones with a fixed rate by hand/
22:52.25[TK]D-Fenderpaci`, most offer a variety of services.  Go read
22:52.25*** join/#asterisk RobH (n=RobH@69.18.84.191)
22:52.49*** join/#asterisk hmmhesays (n=hmmhesay@24-119-176-74.cpe.cableone.net)
23:00.02paci`what would the conf file for using a bluetooth gateway be
23:02.56paci`or better yet, what is a free incoming gatewat
23:02.59paci`gateway*
23:05.13[TK]D-Fenderpaci` : gateway from where?
23:05.23paci`[TK]D-Fender,
23:05.27paci`like, not sure how to explain it
23:05.30paci`I think FWD has it
23:05.33paci`a free incoming number
23:05.41[TK]D-Fenderpaci`, www.ipkcall.com
23:05.46paci`that was it
23:05.48paci`ipkall
23:16.50*** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net)
23:18.25eric2I just had asterisk die on me
23:18.30eric2out of the blue
23:19.07drmessanoDid ya restart it?
23:19.22eric2yes, but luckily I was on the phone when it happened
23:19.31eric2otherwise it would still be dead
23:19.36eric2very strange
23:20.18drmessanoWaht time zone are you in?
23:20.25eric2e.s.t
23:20.33eric2new york/toronto
23:20.37drmessanoWas it a business call?
23:20.44eric2n
23:20.48drmessanooh
23:21.00drmessanoAsterisk deserves the weekend off too
23:21.00eric2but it's been cutting out for a second every now and then too
23:21.11eric2no, residential customers are on the system too
23:21.18eric2so it's gotta work 24/7
23:23.06drmessanoAnyone using a softphone on their blackberry?
23:24.13Qwelldrmessano: buy me one, and I'll set one up
23:24.16drmessanolol
23:24.23QwellYou're a Dr.  You can afford it.
23:24.33drmessanoWhat color?
23:24.40Qwellorange
23:24.50Qwellno such thing?  better make it a Neo1973 then
23:24.55Qwellthose come in orange.  kthx
23:25.16[TK]D-FenderQwell, bkruse wrote an IAX one for it ;)
23:25.23Qwell[TK]D-Fender: I know
23:25.27[TK]D-FenderQwell, That'll be 500$ please ;)
23:25.31drmessanolol
23:26.35drmessanoI don't want a phone built on open specs.. "Open" is overrated
23:26.47drmessanoOMG
23:26.49drmessanoListen to me
23:26.55QwellMr. DRM
23:26.56drmessanoOne day of using Vista.. and look at me
23:27.02QwellGTFO
23:27.17drmessanoActually
23:27.55drmessanoI sit down at my new desktop at work for the FIRST TIME.. and find a bug in Outlook due to a patch from Tuesday that makes it almost unusable
23:27.57drmessanoWelcome to Windows
23:28.26drmessanoI want a showstopper bug in 1.6 beta 6
23:28.32drmessanoJust so I feel better
23:29.40drmessanowow
23:29.51drmessanoLes.net just added something called "Virtual PBX"
23:30.05Qwelleh?
23:30.31alrsdrmessano: that fails to shock
23:30.48drmessanoYou dont even know what the feck it is yet..
23:31.01drmessanoIt lets you create a 0 thru 9 single depth IVR.. you record the greeting and prompts, and point each option a different peer
23:31.25drmessanoSo you can point your DID or DIDs to the IVR and route calls to each peer based on the IVR response
23:32.05drmessanoVery cool for say multiple stores with one main number
23:32.27drmessanoSo you dont have one PBX switching the calls
23:32.28znoG_Question: I have a SIP client configured to only use ulaw/alaw. I tried to call someone and Asterisk says "No compatible codecs found". I enabled SIP debug and I see this line: Capabilities: us - 0x0 (nothing), peer - audio=0xc (ulaw|alaw)
23:32.40znoG_why would Asterisk see NO codecs?
23:34.43[TK]D-FenderznoG_, Because maybe you didn't set any in your sip.conf
23:35.35znoG_[TK]D-Fender: i am using res_config_ldap, however, for that user I have set the allowed codecs to alaw and ulaw, and disallow all
23:35.44znoG_but maybe the allow line is not doing its thing
23:36.08[TK]D-FenderznoG_, if your disalloy=all follows it then it will override your allows.  Order counts
23:37.07QwellI don't know if you can order with ldap...
23:37.56znoG_you can't
23:38.06znoG_it's definately reading the disallowed codecs attribute
23:38.12znoG_but the allowed one... doesn't look like it
23:38.54[TK]D-FenderznoG_, It can be reding BOTH, but like I said, order counts.
23:39.08[TK]D-FenderznoG_, If * processes the disallow second then you have no codecs.
23:39.22znoG_[TK]D-Fender: yep, i understand that, however as Qwell said you can't order with LDAP
23:39.57[TK]D-FenderznoG_, This is that unique position they call "SOL"
23:42.56drmessanoRemove the codecs youre not using
23:43.34drmessanoWho is Strom Carlson?
23:44.55Qwelldrmessano: Strom_C
23:45.36drmessanoHe just posted something on twitter about having some Digium shirts to get rid of
23:45.38drmessano:/
23:47.29St1ckm4ndoes anyone here do any realtime monitoring of call center agents with asterisk 1.4?
23:48.01znoG_[TK]D-Fender: SOL?
23:48.11drmessanoShit Outta Luck
23:48.17BobLutzwhoa
23:48.33drmessanoBobLutz: Are you 14?
23:48.59BobLutzdrmessano, are you a doctor?
23:49.09drmessanoAsked you first
23:49.12BobLutzdamn
23:49.13BobLutzno
23:49.30drmessanoThen i'm going with my second guess of 74
23:49.38BobLutzlol
23:49.56jameswf~drmessano
23:49.56jbot[drmessano] the leading cause of censorship in #asterisk, maybe a Doctor, not really a mess um a no..... The Doctor is accepting new patients, <drmessano> I am not a OB/GYN but I'll have a look anyway
23:50.26BobLutzlol, i dont remember the OB/GYN part
23:50.35paci`whats better, sip/iax
23:50.42jameswfIax2
23:50.50jameswf~iax2
23:50.51jbotwell, iax2 is http://www.voip-info.org/wiki-IAX
23:50.52[TK]D-FenderSt1ckm4n, Yes.
23:51.19paci`well,
23:51.23drmessanoLinks that point to the voip-info should be banned
23:51.25paci`ipkall offers sip and iax
23:51.28paci`which would be better?
23:51.34[TK]D-Fenderpaci`, SIP generally
23:52.01paci`what is FWD's sip server?
23:52.02jameswfunless you use nat then sip suxors
23:52.38St1ckm4n[TK]D-Fender: are you using a program that parses the manager api output or are you storing/reading it from a db
23:52.45drmessanofwd.pulver.com
23:53.03jameswf~fwd
23:53.04jbot[~fwd] Free World Dialup, created by Jeff Pulver, is a free SIP server for P2P style that does not involve the PSTN (there is a charged option for this as well though). http://www.freeworlddialup.com/
23:53.26drmessano~ipkall
23:53.28drmessano:(
23:53.43`SauronFWD offers IAX2 as well
23:53.54drmessanoIt doesnt work
23:53.58paci``Sauron, FWD != IPKall
23:54.14`SauronI never said they were the same
23:54.25`SauronIt was an FYI. :p
23:54.28drmessanoFWD IAX2 is so unreliable, they even tell you not to use it
23:54.34paci`ah
23:54.39[TK]D-FenderSt1ckm4n, Yup, I parse it the very dirty way.
23:54.50paci`mmm
23:55.13paci`how exactly
23:55.15`SauronHum, I thought iax was suggested over sip, at least back in the day when I set it up.
23:55.22`Sauronshrug
23:55.30paci`;register => 1234:password@mysipprovider.com
23:55.32drmessanoIts ALWAYS been "experimental" with them
23:55.34paci`ok, so that would be
23:55.48paci`register => NUMBER:NUMBER@fwd.pulver.com
23:55.50paci`?
23:55.50drmessanoFWD has never pushed using IAX2 over SIP
23:56.10drmessanonumber:password
23:56.13St1ckm4n[TK]D-Fender: I've been looking at FOP but it seems buggy, I've written a php page that connects to the manager api and just parses the show agents output but I don't like having to continuously query the server, was it a pain in the ass to do it the dirty way?
23:56.33`Sauronhum
23:56.36`Sauronohwell
23:56.38paci`drmessano, the password to my FWD account?
23:56.44drmessanoyes
23:57.00paci`would the NUMBER be the username
23:57.03paci`or sip #
23:57.08drmessanoNo, it would be the number
23:57.54[TK]D-FenderSt1ckm4n, I *do* continuously query the server... thats the "dirty" part (in addition to the fact its pur text-parsing)
23:57.56[TK]D-Fenderpur*
23:58.12*** join/#asterisk frogonwheels (n=michaelg@203.59.141.93)
23:58.26paci`how can I test asterisk if its conected to my sip server
23:58.32paci`I havn't used it in a long time
23:58.43drmessanosip show registry
23:59.07frogonwheelsI have asterisk running on openwrt:
23:59.24paci`[Mar 14 19:59:14] NOTICE[23880]: chan_sip.c:7425 sip_reg_timeout:    -- Registration for '903316@mysipprovider.com' timed out, trying again (Attempt #3)
23:59.25paci`rawr
23:59.26St1ckm4n[TD]D-Fender: do you think there is any stability issues if you have 5 pc's continually querying the server ever 3-5 seconds
23:59.35frogonwheelsif I start it up - I get no ctl file - but if I use -vv or -d   it does  create one.
23:59.38paci`oh
23:59.38paci`duh
23:59.39paci`rofl
23:59.42paci`i didnt change the host
23:59.43frogonwheelsany clues why?

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