IRC log for #asterisk on 20091022

00:00.51beekMan, is my Perl rusty.
00:01.22Kattyleifmadsen: http://www.dietandfitnesstoday.com/rda.php
00:01.32Kattyleifmadsen: those numbers are very acurate.
00:01.43Kattyleifmadsen: http://42ndhealthstreet.blogspot.com/2009/10/nutrient-recommendations-what-we-need.html
00:02.14Kattyleifmadsen: step one of my research is done!
00:02.17*** join/#asterisk geneticx (n=geneticx@adsl-2-215-240.mia.bellsouth.net)
00:02.32Kattyleifmadsen: step two is to keep track of what i eat tomorrow so i can see how much of what i ate.
00:03.09Kattyleifmadsen: and that will be very enlightening
00:03.18leifmadsenKatty: ya, I need to keep a food log again
00:04.04geneticxhiya
00:04.10Kattyleifmadsen: ever see anything called Ascorbic Acid in on an ingredient list?
00:04.38leifmadsenKatty: yes, I have seen that... let me see what that was again
00:04.47Kattyleifmadsen: its a different form or alternative name for vitaminc c
00:04.59leifmadsenah yes, I knew it was like citric acid
00:05.44Kattyleifmadsen: and if you are deficient in vitamin c, you will have bleeding gums, painful joints, slow healing wounds, bruising (i have this), nose bleeds, tooth decay, loss of appetite, muscular weakness, skin hemorrahages, capillary weakness, anemai, and impaired digestion
00:05.58KattyVitaminc C is ascorbic acid.
00:06.15Kattyidk if citric acid is quite the samel.
00:06.16*** join/#asterisk troubled (n=troubled@unaffiliated/troubled)
00:06.20Kattytho citric acid probably contains vitamin c.
00:06.30Kattylet's ask wikipedia!
00:06.38leifmadsenKatty: heh, that's what I did :D
00:07.02Kattyjit's a food additive used for flavoring and preservation
00:07.04Katty:<
00:07.15leifmadsenI need orange juice now
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00:07.48Kattyyes.
00:08.05leifmadsentasty!
00:08.09Kattyyes!
00:08.11kissg_huhello there
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00:08.35Kattyhmm, neat.
00:08.42KattyI kept seeing Riboflavin on packages of pasta.
00:08.47Kattyapparently that's vitamin b2
00:08.53Kattyniacin is b3
00:09.12Kattyi'm gettin smrt!
00:09.23leifmadsenhawt!
00:09.40leifmadsenmaybe you will eventually go to school for chemistry
00:09.52Kattynow this makes me wonder tho
00:09.55Kattycause salt...
00:10.00Kattyyou would say is sodium, right?
00:10.16kissg_huI'd like to ask for your assistance regarding dial rules in Asterisk 1.4
00:10.16Kattyexcept it's sodium chloride :/
00:10.23leifmadsenyes :)
00:10.30kissg_huis anyone there who could help me?
00:10.31leifmadsenkissg_hu: ask away
00:10.32Kattyso i wonder how you figure out the nutritional value of Salt.
00:10.43leifmadsenKatty: there isn't really a nutritional value to it
00:10.51leifmadsenKatty: lower is typically better
00:11.17leifmadsensalt causes dehydration and you to carry excess water (which seems contradictory)
00:11.25Kattyit also raises blood pressure
00:11.35Kattyoooh wait
00:11.36leifmadsenbut kinda makes sense, since your body is trying to retain water since it is dehydrating
00:11.39Kattyit has a * by it
00:11.42kissg_huI've created a dial rule, but I'm not sure it really works the way it should
00:11.51Katty(Note: nutrients with a star indicate Adequate Intake or AI because no RDA can be established)
00:12.21leifmadsenkissg_hu: use a pastebin and show the dialplan you've created, and the console output, and explain what you expect to happen, and what is actually happening
00:12.35Kattyleifmadsen: drink more OJ.
00:12.41Kattyleifmadsen: and eat a tomato
00:12.44leifmadsenKatty: I try to use sea salt when I want salt for tasting
00:12.48kissg_huleifmadsen: thanks, but it's simple and in fact, I have no problem with it :)
00:12.51Kattyleifmadsen: vitamin c decreases the risk of certain cancers by 75%
00:12.56leifmadsenKatty: I love tomatoes, and am drinking OJ now -- yesterday I drank a V8
00:13.04kissg_huI just would like to be sure it really works the way I expect
00:13.14leifmadsen~ask
00:13.14infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
00:13.50Kattyleifmadsen: good news! cayenne pepper is high in vitamin c ^_-
00:13.59Kattyleifmadsen: EAT MOAR HOT SAUCE
00:14.02kissg_huFirst, spoken out: "Remove 06 prefix from number add 0036 instead. the next one or two numbers are local prefixes"
00:14.10leifmadsenKatty: yes, cayenne pepper is very good for you
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00:14.24kissg_hu"Then, 6 or 7 numbers which are subscriber numbers"
00:14.26Kattyoooh and avocado
00:14.29Kattyi had some of that for dinner.
00:14.33leifmadsen0036${EXTEN:2}
00:14.40kissg_huthe rule I created is: "06|0036+[1-99]XXXXXX"
00:14.43leifmadsenKatty: yes, avocado's have the "good" fats in them
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00:15.01leifmadsenKatty: substitute for mayo on sandwiches
00:15.10Kattymayo is pretty icky
00:15.10leifmadsenKatty: mustard is also ok to have as well
00:15.11kissg_huand this one for numbers where 7 digits identifies the subscriber: "06|0036+[1-99]XXXXXXX"
00:15.15leifmadsenKatty: ah, I love mayo :)
00:15.20kissg_huare these okay?
00:15.20Kattyit's okay with chicken salad sammiches
00:15.29Kattybut have you SEEN the ingredient lists?
00:15.30leifmadsenKatty: I use it in tunafish
00:15.36leifmadsenKatty: oh, I know how bad it is :)
00:15.38leifmadsenI just like the taste
00:15.46Kattyhelman's real seems to be Best For You
00:15.51leifmadsenok, off to play some flight simulator on my new 24" monitor :)  reboot to windows time!
00:15.51Kattyconsidering it's a processed food item :/
00:15.56Kattybyebye
00:37.13Knightfalhey guys is dialparties.agi specific to any particular flavor of an asterisk build out there?
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01:01.37Kattyreturns
01:02.46Kattypokes about for life
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01:07.31manxpowerKnightfal: dialedparties.agi is not part of the official Asterisk.
01:07.55manxpowerIt is part of FreePBX GUI
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01:12.37CcRnpanyone used asterisk CEL in production ?
01:12.51CcRnpanyone used asterisk CEL in production server?
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01:54.03knctrnlanyone have any tips on areas to focus on preparing for DCAP?
01:55.06jayteeyeah
01:58.04jayteebe able to setup a network with an Digium analog card with 1 FXO and 1FXS, 1 Polycom phone and X-Lite with dialplan, VM and an IVR in 90 minutes for the lab, for the written test, read "the book" cover to cover and study about protocols, codecs, etc.
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01:58.51jayteeand use CentOS so you're familiar with the lab distro
01:59.03jayteeat least that's what they're using in Huntsville
02:02.51knctrnlIm taking it friday
02:03.08knctrnlcurrently taking the advanced class in huntsville
02:03.59knctrnlDo you know how complex the IVR will be?
02:06.57jayteehad a GotoIfTime in it and a simple Press 1 to go to Bob type of menu. Do a search of sample or example dialplans on voip-info.org and you'll get some stuff that's fairly similar.
02:07.59knctrnlthats not too bad
02:08.01jayteeif they give you the option of using the T1 card or not, pass on it. it'll save you time and let you focus on the dialplan stuff and configs.
02:10.02jayteecuz you'll need to get your 3 sip peers configured so you'll have a DAHDI analog client, a SIP client for the Polycom and a SIP client for the X-Lite softphone on the server with VM boxes and the IVR etc. If you find decent examples from googling cuz you're allowed to use the Web during the lab then you can cut and paste for your configs but you'll still end up with alot of editing.
02:10.24jayteeI ran out of time. passed the written and got 50% on the lab
02:11.25jayteewho's teaching your class? Jared?
02:11.26knctrnlweb access is great
02:11.32knctrnlDavid Duffett
02:11.51knctrnlHe is very good
02:11.53jayteenot sure if I met him. Jared Smith taught my class
02:12.00knctrnlDavid is from UK
02:12.06[TK]D-FenderI've met Jared twice IIRC...
02:12.07knctrnlI think Jared is out this week
02:12.45jayteehe comes to Huntsville for the classes but runs a business in Virginia or D.C. I think the rest of the time
02:13.00knctrnlI came for the Fast Start and so far after 3 days down of Advanced the only new material is PRI config
02:14.21jayteeI took the Advanced back in November. I'd been using * for about a year and a half at that point.
02:15.01knctrnlyeah I have been using for about 8 months.  My main worry is syntax of course
02:16.59jayteeif the lesson plan hasn't changed much then you should already have a fair working copy that's similar to what's going to be required for a config for the lab. I think we went over IVR stuff on thursday.
02:17.06Qwellknctrnl: Did David bring candy with him? :p
02:17.12QwellI might need to go harass him tomorrow..
02:17.20knctrnloh yeah
02:17.23knctrnlloads of candy
02:17.34Qwellsweet.  I'll make a stop down there :p
02:17.38jayteeQwell, have you managed to quit smoking yet?
02:17.42Qwelljaytee: meh
02:17.50jayteesame here
02:18.04knctrnlYou in hsv qwell?
02:18.09Qwellknctrnl: yeah
02:18.22knctrnlim local too
02:18.24knctrnlwhat comp?
02:18.25Qwellknctrnl: I was upstairs yesterday with the group for lunch
02:18.35Qwellknctrnl: 2nd floor :p
02:18.46knctrnlgotcha
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02:23.11knctrnlI was up there too. There were several non class folks up thre
02:23.34Qwell< non class folk
02:24.00QwellI'm difficult to miss
02:24.14knctrnlsitting with JAN?
02:24.21Qwellmmhmm
02:24.58knctrnlI gotcha. im the big bald guy
02:28.35plasmidwhen i add an extension and I have a TDM400 (for my house); do I choose Generic SIP Device or Generic Zaptel Device?
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02:32.47p3nguinhardwire: I don't think so.
02:33.49hardwireI bet you do.
02:34.12p3nguinjblack: Although I acknowledge that I have been unpung, was there something you needed me for?
02:34.35p3nguinhardwire: Personally, I don't like it that much.
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02:36.15plasmidwhat channel is recommended for a TDM400 (This device uses zap technology. (Via DAHDI compatibility mode))  ; I am adding an extension: 100 to test.
02:37.29p3nguinAfter all that fighting with SIP and the conclusion was that I switched the particular client to IAX2, now the audio from said client is a loud buzzing sound.  It worked well for at least one day, now not so much.
02:40.57hardwirebuzzing noises are quite analog.
02:42.12p3nguinNot sure where to look for troubleshooting.  I'm pretty sure the microphone on the computer is still working correctly.
02:42.26jblackp3nguin: I had lost the prompts you made me. I found them
02:42.34jblacksorry to disturb you
02:42.41p3nguinOh, no problem.
02:42.55p3nguinJust got back in and was catching up.
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02:52.36[Outcast]what is that md5 hash created from in sip registration response?
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03:18.34dlynesAnyone happen to know how many 'flavors' there are for the digium t.38 fax engine?
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04:12.04d1ss3ntanyone around to take a PRI question?
04:14.09d1ss3ntchan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1
04:14.11d1ss3nt:(
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04:57.34luckyabaNitro tha CiMien
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05:03.32luckyabaNitro tha CiMien
05:04.16knctrnlanybody know of any vialble solutions to do conference briding wiht G722 wideband?
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05:04.45knctrnlAlso the capability do join nonG722 callers but not hurt the quality for the HD boys
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05:05.58[TK]D-Fenderknctrnl: look at FreeSWITCH
05:09.08knctrnlis it pretty simular to get running?
05:09.12knctrnlconcepts the same?
05:09.42knctrnlwould the confbrige app in 1.6.2 be workth trying?
05:09.43[TK]D-Fenderknit is its own world
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05:10.35[TK]D-Fenderknctrnl: *'s mixing core is SLIN, not WB
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05:11.50knctrnlwb?
05:12.19knctrnlI was speaking with a DIgium engineer and he said that the new conf module can handel it becuse is does not mix on dahdi
05:14.09[TK]D-Fenderknctrnl: Everything involving transcoding always brought it down to SLIN which would kill it
05:14.27[TK]D-Fenderknctrnl: unless they are doing something VERY new..
05:15.00Corydon76-dig[TK]D-Fender: you mean like mixing in slin16?
05:15.39[TK]D-FenderCorydon76-dig: I mean like I'ver reached about a step ahead of my experience :)
05:15.52[TK]D-FenderCorydon76-dig: Plesae do take over ;)
05:16.06Corydon76-digNah, you're doing fine
05:16.14Corydon76-digI was about to go to bed, anyway
05:22.00[TK]D-FenderI'm off myself....
05:22.02[TK]D-Fenderlater all
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05:33.05miteshdigium tdm410, does it support gsm networks?
05:34.58Corydon76-digNope
05:35.10Corydon76-diganalog tdm networks only
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05:36.21miteshCorydon76-dig, so does this mean that call from cellphones cannot be entertained nor routed to cellphones by the asterisk server
05:38.24TJNIINo, it means that card can't serve as the link.
05:39.37miteshbut can the call from and to gsm network be made?
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05:44.29florzno, for that you need radiation proof interface cards (and telephones)
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05:50.20Corydon76-digmitesh: Can you change someone's mind by willing it?  Or do you have to communicate with them?
05:50.24ZenBSDiHey guys .. whats a popular web interface for asterisk?
05:50.59Corydon76-digYou asked about willing it.  We're saying you have to communicate.
05:51.55miteshCorydon76-dig, sorry, didnt get you
05:52.35Corydon76-digThe card does not directly talk to GSM networks, but it certainly can by going through the PSTN backbone
05:52.59Corydon76-digThere IS other hardware that can talk DIRECTLY to the gsm network, but the tdm card is NOT it
05:53.42knctrnlzenbsdi google Asterisk GUI  or asterisk Distribution
05:54.50Corydon76-digZenBSDi: there's only one web interface for Asterisk... AsteriskGUI... other GUIs are for particular applications related to Asterisk
05:55.14Corydon76-digCall centers, call conferencing engines, PBX administration...
05:55.52*** join/#asterisk oej (n=olle@ns.webway.se)
05:55.52Corydon76-digbut as for Asterisk itself, only one
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06:17.24miteshCorydon76-dig, there?
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07:47.46jojo^Is it possible to route a SIP call with asterisk? Thus, instead of creating a new leg with Dial, just route the call like openser, with the same callid.
07:49.09jojo^Maybe "like a SIP proxy" is what I'm looking for.
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07:59.26mchoujojo, use a real sip proxy like Kamailio.  Asterisk is a swiss army knife but individual blades dull quickly
08:00.24kaldemari'd put it like asterisk is a B2BUAS, not a proxy.
08:08.36jojo^mchou, We got Kamailio in front, but due to the very flexible nature of Asterisk AGIs it would be very nice to have all logic there, instead of split logic between Kamailio and our asterisk AGI. But I guess the workable solution will be to pass the call back to Kamailio when the AGI have decided it should be forwarded (but then we need to send it back to our AGI if the forward times out and should be sent to our voicemail.. This is basically what I'm trying to avoid)
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08:09.11d1ss3ntanyone how if its possible to update to a newer firmware on a digium card?
08:09.25d1ss3ntvia asterisk/zaptel, that is
08:11.54ChannelZhmm I don't believe I've ever seen firmware for digium cards
08:13.45kaldemard1ss3nt: ask digium sales for that. years ago you needed to send the card to digium for that.
08:14.33d1ss3ntweak sauce
08:15.55d1ss3ntAnyone ever seen this with a Digium card on a standard US PRI line? --> NOTICE[11150] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 1
08:16.12d1ss3ntw/ intermittent alarms and call drops/resets
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08:51.29mbrevdadoes anything exist like chanspy, but where you can speak to BOTH parties?
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09:06.54*** join/#asterisk Torocatala (n=torocata@80.224.43.54.static.user.ono.com)
09:06.58TorocatalaHi
09:12.01TorocatalaI have a weird issue whit my PBX, I made an IVR, and works fine whit softphones, but, when I use a hardware phone, and I press, 1 (or another number) the IVR do not answer, I try whit to hardware phone whit the same result
09:12.12Torocatalaanyone know about this issue?
09:15.55Torocatalathx...
09:20.51Torocatalais a Matrix?
09:21.49trogssounds like a problem with the dtmf
09:22.13Torocatalathanks!
09:22.24trogsyou'll prob want to check that your phone is set to use rfc2833 mode
09:22.29trogsfor dtmf
09:22.59GuggeAnyone know why Expire would show -1 when i register an realtime iax peer on my asterisk 1.4.21? http://pastebin.com/m7e113018
09:23.16trogsyou'll probably want to check it for dialing in from externally, as there might be trunk settings you need to make sure is configured correctly
09:23.43trogsrfc2833 is generally the best thing to set it to.
09:23.52trogsbut really does depend on your trunk
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09:30.24Torocatalahey, thanks a lot trogs
09:30.56TorocatalaI put the rfc2833 in the phone and now works fine
09:31.02Torocatala:D
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09:39.28devyllwhat is the recommended sollution for uploading queue_log data to mysql real time ?
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10:18.15mattbollhi, do you know some software in linux, that monitor easily blf (who is connected, incall…) and which is not a webpage ?
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10:43.25trogsmattboll: you might be able to use FOP
10:43.41trogshttp://www.asternic.org/
10:44.00trogsTorocatala: good stuff, was pretty sure that would do it :)
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10:52.33Grofguys
10:52.47Grofi have two asterisk machines
10:53.06Grofwhile there are SIP calls between them
10:53.17Grofone of the machines shuts down
10:53.25Grofand other one just freaks out
10:53.33Grofspamming "exceptionally long queue"
10:53.39Grofis that normal?
10:53.40Grof:D
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11:13.38i-pinkhii
11:13.54i-pinksomeone here?
11:14.36i-pinkhow i can change the user agate in twinkle?
11:14.45*** join/#asterisk Weedle (n=Weedle@ausphreak/Weedle)
11:14.48Weedlehey peeps
11:15.10Weedleim trying to get call manager to work with asterisk, the call come in but comes up with this...
11:15.13Weedle<PROTECTED>
11:15.33i-pink?
11:15.45i-pinkyou know how i can change the user agate in twinkle?
11:16.01Weedlethe call manager basicly sends of the call to asterisk, doesnt register, so what do i have to do to sip.conf to allow it to pick up the call manager and select a context
11:16.42*** join/#asterisk Jankooo (n=jani@dsl5402AECF.pool.t-online.hu)
11:16.52JankoooHi!
11:22.15JankoooIs there anybody who has a little time to help me?
11:22.21JankoooI am just installed the asterisknow under wmvare...
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11:32.35tzafriri-pink, hi
11:32.44i-pinkhii
11:32.59tzafrirwhat do you mean by "user agate"?
11:34.30i-pinki use twinkle for soft phone
11:35.08i-pinkand i want to change the "user agate"
11:35.22kaldemarJankooo: there is #asterisknow for asterisknow related questions.
11:35.43i-pinkis send the name of the program to the server
11:36.12i-pinktzafrir ^
11:36.33kaldemarWeedle: if the call doesn't match any peer, asterisk will look for extensions in the context defined under [general].
11:36.52i-pink?
11:37.05Weedleooh cheers kaldemar
11:37.34tzafriri-pink, "user agent"
11:37.46i-pinkyes
11:38.50Weedlekaldemar nope still get..
11:38.51Weedle[Oct 22 05:38:31] NOTICE[968]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '2222' rejected because extension not found.
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11:41.37i-pinktzafrir,  "user agent " is the name of the program
11:41.39tzafriri-pink, can't find it anywhere. What do you need it for?
11:42.02i-pinkfor me
11:42.24tzafrirIn the worst case, I guess you can always patch the source :-(
11:42.38i-pinki start to compile it :-)
11:43.39i-pinkand i try to make filter to ettercap for temporary change
11:49.29kaldemarWeedle: you don't have extension 2222 under the context asterisk is looking for. pastebin a CLI output for a call with verbosity and sip debug enabled and well take a closer look.
11:51.43Weedlei have the extention in there, ive tryed different contexts, placing it under general.
11:53.30*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
11:53.45Weedlehow do i enable sip debug?
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11:55.46WinZWeedle, core set debug channel all
11:58.18kaldemarWeedle: sip set debug on
11:58.51*** part/#asterisk equality4xy (n=equality@76-219-201-200.lightspeed.irvnca.sbcglobal.net)
12:00.04Weedlehttp://pastebin.com/m528c830d
12:01.50kaldemarstill waiting for the sip debug...
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12:16.03ManxPower-workARGH!  The tech I'm working with simply won'y stop talking.
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12:39.27ariel_Morning
12:39.48WeedleManxPower-work slap him...with your penis
12:42.49beekWeedle: what if the tech guys likes that?
12:43.36ManxPower-workWeedle: He's 800 miles away.  I don't think it would reach.
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12:44.38Weedlebeek then get a chick and slap him with her tits
12:45.09beekWeedle: now *I* would like that.
12:45.35Weedlehehe
12:45.41Weedlewho wouldnt
12:46.36ariel_wow slapping by proxy
12:46.44ariel_just does not work the same way
12:47.04beekJust borrow [TK]D-Fender's Clue-Bat (TM)
12:47.22[TK]D-FenderMY PRECIOUS!!!!!!!!!!!
12:47.53[TK]D-Fenderhordes his ironwood ClueBat (tm) and sneers evilly
12:48.08ariel_argh left my pen at home....need to raid the supply cabinet...
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12:56.15scalex000hey, good morning guys
12:56.39ariel_morning
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13:04.43GuggeI have a box that gives errors when asterisk sendes "Require: timer" in the 180 Ringing packets ... is there any reason to send those ... and is my box broken?
13:05.16Dovidno. from the sound of it you need dahdi (or some timing source) this is a pure guess ;)
13:05.26[TK]D-FenderDovid: Correct.
13:05.28Doviddo u have any telephony hardware in the box ?
13:05.41DovidTK: Thanks. I finally got something right ;)
13:05.44[TK]D-FenderDovid: That clearly doesn't show any sense of know what you're takling about :p
13:05.48DovidPats himself on the bvack
13:06.07[TK]D-FenderDovid: FAIL :D
13:06.13Dovidi am lost.
13:06.24Dovidcorrect was that it was a guess and NOT correct ?
13:06.41[TK]D-FenderGugge: What is the box that gives the error to *'s 180?  And what is the error?
13:06.58Guggeits some fiber box, with an ata
13:06.58Dovidtalking abotu failing..... http://failblog.org/
13:06.59[TK]D-FenderDovid: Correct that it WAS a "pure guess".
13:07.03Guggeand it doesnt return an error to *
13:07.05Dovidhaha
13:07.11Guggethe box just logs "invalid packet"
13:07.19[TK]D-FenderGugge: Where do you see this error?
13:07.28Guggeon the console of the fiber-ata box
13:07.35[TK]D-FenderGugge: Maybe your device doesn't care about progress indication.
13:07.57[TK]D-FenderguBut it sounds more like a nuisance than a "problem
13:07.58Guggewell, it doesnt play ringtones when it get the 180 ringing :)
13:08.09Guggeif u remove the require: timer from chan_sip.c it does :)
13:08.34Guggei just wonder why theres even an require: timer in the ringing packets
13:08.38*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
13:08.38*** mode/#asterisk [+o leifmadsen] by ChanServ
13:09.03Guggei would expect the require header only in INVITE and OK
13:09.43*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:10.30*** join/#asterisk Skeeter- (i=Skeeter-@c216.218.2-65.clta.globetrotter.net)
13:10.44Skeeter-anyone ever worked with Polycom Spectralink wireless phones
13:11.10kaldemarthe timer is an option tag that tells the other end that asterisk requires session timer support.
13:13.58Kattyomnomnomnoms on breakfast burrito
13:14.23beekGood morning Katty
13:14.43ariel_morning Katty hope your doing better this morning. Sends a hug her way....
13:15.00Kattyariel_: yesh :>
13:15.06Kattyariel_: the world is a little brighter this morning
13:15.09Kattyhugs ariel_
13:15.15Kattygood morning beeks :>
13:15.17Kattyhugs on beek
13:15.33beekhugs back
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13:19.32Guggekaldemar: thats what i figured ... but i have it set to "accept" in sip.conf ... and it still sends the require: :)
13:19.56*** join/#asterisk oej (n=olle@static-213-115-251-100.sme.bredbandsbolaget.se)
13:19.57Guggei would expect "accept" to only accept the timer extensions, not require it
13:20.10[TK]D-Fender[09:13]* Kattyomnomnomnoms on breakfast burrito <- so much for eating "healthy" :|
13:21.05*** join/#asterisk galeras (n=galeras@186.80.181.115)
13:21.07Kattyoh it's quite healthy.
13:21.13Kattythankyouverymuch
13:22.15galerasIs possible to send a fax with T38 protocol over a PRI (i mean directly connected to the * box)
13:22.26WinZhttp://pastebin.com/d7bd505ef -- guys, what can this error mean? Sometimes it happens on outgoing call through my trunk. The call is established, but silence only I can hear
13:23.05ManxPower-workgaleras: No.
13:23.27ManxPower-workWinZ: Looks like a provider or network problem
13:24.10WinZok
13:24.14WinZManxPower-work, thank you
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13:34.51tuxx-hey guys. We have a client with an asterisk PBX. Every day they have some calldrops, and the cause seems to be the following error: 'DEBUG[26670] channel.c: Didn't get a frame from channel: Zap/30-1'. I searched google, but everyone having this problem seems to be stuck in finding an answer. Is anyone here who could point me to a solution, or maybe a workaround?
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13:40.16*** join/#asterisk superbeef (n=superbee@74.84.194.4)
13:40.42superbeefdo you think it's more CPU intensive to take calls with a T1 card and software echo cancelationation then from a IAX trunk?
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13:41.33angryusersuperbeef, why dont you test ?
13:42.11*** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek)
13:42.12superbeefangryuser: I prefer to dabble in speculation
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13:48.56[TK]D-Fendersuperbeef: Obviously
13:52.54superbeefi'm actually in a pretty good position to test since i have a 2 PBX's connected with a T1 crossover
13:55.32Kattyblinks a bit
13:55.43Kattywow...there are absolutely no nutrients in that tortilla i ate.
13:55.46ManxPower-worksoftware EC takes CPU cycles
13:55.47Kattyexcept for sodium.
13:55.57Kattyand a smidgen of calcium
13:56.10Kattybut the other stuff was good for me!
13:56.12ManxPower-workthose are nutrients!
13:56.45Kattylooks up whole wheat tortilla
13:57.28Kattywell, it's better.
13:57.31Kattybut still mostly useless.
13:57.46[TK]D-Fendersuperbeef: You shouldn't need EC anyway between them.
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13:59.13DavidR2008does anyone have any information on how accurate timing has to be for iax to work well? Will an average of 99.7+ work? I'm testing dahdi dummy using dahdi_test and that's what I'm getting off of my hardware
14:03.10superbeef[TK]D-Fender: probably not.... I just have the T1 cross over so I can make sure the T1 card is setup right before I deploy a PBX
14:03.32Kattythe day is young, but so far i've had absolutely no Vitamin E, b1, b2, b3, b6, b9, and very little K
14:04.02[TK]D-FenderDavidR2008: Are you using trunk mode to your provider?
14:04.23DavidR2008no, it's between local * servers
14:04.38[TK]D-FenderDavidR2008: then timing should be irrlelvant
14:04.54DavidR2008ok, thanks
14:05.31*** part/#asterisk Boardy (n=chatzill@ip565785d1.direct-adsl.nl)
14:06.40*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
14:06.40*** mode/#asterisk [+o Deeewayne] by ChanServ
14:07.28*** join/#asterisk cusco (n=trilili@213.63.137.210)
14:07.30cuscohi
14:08.59cuscodahdi_scan shows: http://paste.debian.net/49710/
14:09.11cuscothere should be 6 channels with 2 spans
14:09.29cuscoin asterisk's cli dahdi show channels only show 30 channels
14:09.30cuscowhy?
14:10.44kaldemarcusco: you only configured 30 channels in asterisk
14:11.31cuscook /etc/asterisk/dahdi-channels.conf has http://pastebin.com/f62b4b52d
14:11.42cuscowhat am I missing?
14:12.57guaxlunch
14:13.23*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:13.25kaldemarcusco: is that #included in chan_dahdi.conf?
14:14.20[TK]D-Fenderwonders why he doesn't SEE the primary file, or the backup of only 6 channels showing up
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14:18.31tuxx-hey guys. We have a client with an asterisk PBX. Every day they have some calldrops, and the cause seems to be the following error: 'DEBUG[26670] channel.c: Didn't get a frame from channel: Zap/30-1'. I searched google, but everyone having this problem seems to be stuck in finding an answer. Is anyone here who could point me to a solution, or maybe a workaround?
14:18.57cuscokaldemar: you got me
14:18.59cuscothanks
14:19.17ManxPower-worktuxx-: that usually means "far end hungup"
14:19.42Kattyconsiders this lack of vitamin b
14:19.42ManxPower-workyou would not even see the message if you didn't turn on DEBUG
14:19.55Kattyleifmadsen: i'm lacking in B, K, and E so far this morning
14:20.15Kattyleifmadsen: the morning is young, no doubt
14:20.19cuscokaldemar: chan_dahdi.conf has http://pastebin.com/f66bee108
14:20.25leifmadsenKatty: oh snap! Hmmm... I ate my half a banana, so my K should be ok, and I had some greens+ -- I probably need a multivitamin and some fishoil though
14:20.25cuscoso if I modified to look like:
14:20.33ManxPower-workKatty: you realize the eating healthy, getting exersize, not smoking, drinking, drugs, etc doesn't acutally make your live any longer.  It just SEEMS longer.
14:21.03*** join/#asterisk moy (n=moy@74.12.134.3)
14:21.11tuxx-ManxPower-work: we did turn on debug idd, but we called the 'far end' and he/she says the call got disconnected. and this usually happens 3 times a day
14:21.22*** join/#asterisk KavanS (n=KavanS@71.117.242.28)
14:21.30Kattyleifmadsen: judging from items high in vitamin E, i know i will have some by the end of the day.
14:21.38Kattyleifmadsen: nuts, oils, spinach, sunflower seeds, whole grains
14:21.45tuxx-vitamins, eugh.
14:21.53leifmadsenKatty: nice! now I must go take some pills :)
14:21.57Kattyleifmadsen: kk
14:22.01ManxPower-workzttool or dahdi_tool  looked for missed IRQs/Interrupts
14:22.02tuxx-real nerds live on caffeine and pizza!
14:22.02cuscoso if I modified to look like: http://pastebin.com/m10884cc3 it would be OK, right kaldemar ?
14:22.05kaldemarcusco: your dahdi-channels.conf is irrelevant because you don't include it in chan_dahdi.conf
14:22.21kaldemarcusco: yes, that's one way of solving it.
14:22.23cuscokaldemar: i would activate 2nd span that way
14:22.45cuscothats all I need, a new group and setting channel =>
14:22.51*** join/#asterisk Jankooo (n=jani@dsl5402AECF.pool.t-online.hu)
14:22.52kaldemarcusco: but your switchtype is different from dahdi-channels.conf now.
14:23.04DavidR2008do you have to have chan_dahdi to use the dahdi dummy driver?
14:23.12cuscono
14:23.35superbeeftuxx-: you should turn on pri debugging
14:24.02tuxx-hmkay, will do.
14:24.21cuscokaldemar: so now I have to unload and load chan_dahdi.so again
14:24.35cusco(We are having calls right now, I will have to wait)
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14:29.19cuscokaldemar: dahdi-channels.conf is not valid, it was never in use. I just did not realised it before :p
14:29.30cuscoshould have figured from the "context"
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14:31.39CGMChrisIs anyone familiar with how AGI/AMI?  I am trying to place an outbound SIP call and detect when it has been answered, but cant seem to figure it out.  BackgroundDetect() *appears* to be detecting rings as noise.
14:33.06ManxPower-workCGMChris: analog or PRI or SIP?
14:33.35CGMChrisManxPower-work: SIP.
14:33.47*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
14:34.02[TK]D-FenderCGMChris: AMD <-
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14:34.31cuscooops
14:34.32cuscowhere is the changelog at?
14:34.57cuscoCHANGES file has only "Functionality changes from Asterisk 1.6.1.1 to Asterisk 1.6.1.2" and previous
14:35.05cuscowhat are the latest changes until 1.6.1.6
14:35.13[TK]D-Fendercusco: www.asterisk.org
14:35.39cuscoim there
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14:36.24ManxPower-workcusco: ChangeLog
14:36.59cusco"We couldn't find what you were looking for."
14:37.06leifmadsenwhat link?
14:37.16leifmadsenthere are some dead links on the asterisk.org site I'm going to fix up this morning
14:37.21cuscothere is no link, I just typed /Changelog
14:37.31leifmadsenheh
14:37.36leifmadsenit's in the tarball
14:37.44cusco15:34 < cusco> CHANGES file has only "Functionality changes from Asterisk 1.6.1.1 to Asterisk 1.6.1.2" and previous
14:37.45leifmadsensee the -summary files too
14:37.50cuscoah
14:37.55cuscols
14:38.04leifmadsencusco: I know -- that means there haven't been any super-major-freak-out changes since those versions
14:38.18*** join/#asterisk E-bola (i=bola@ip181.rev112.brygge.net)
14:38.21E-bolaHello
14:38.22cuscothere must be some deserving releasing it :p
14:38.28leifmadsenChangeLog is all changes between versions, and the summary is a summary of those changes which closed issues on the bug tracker
14:39.15cuscothanks :)
14:40.42CGMChris[TK]D-Fender: Isnt AMD only useful after the call has in fact been answered?  The problem I'm having is detecting when the call is actually answered, not if it's a machine or person.  I am using AMI to Originate a channel (SIP/provider/external phone #) and an application [AGI(script.php)].  The problem is that script.php begins executing as soon as the SIP channel connects (ringing), but before a person is on the line.
14:40.49E-bolaI got a relatively newbie qeustion: I have a place in my dialplan where i need to call 1 phone for 10 secs, if no answer then call another phone for 10 secs, etc. I would like for the caller to hear continoues music while this system moves from phone to phone, is that possible?
14:41.18[TK]D-FenderCGMChris: if its SIP SIP is supposed to be OOB
14:41.32E-bolaDoes specifying m as an option restart the music on hold for each dial cmd?
14:41.42[TK]D-FenderCGMChris: Which means you're not supposed to do anything
14:41.51ManxPower-workE-bola: yes, possible, no easy.  But if you don't mind the hold music starting over or don't mind a very short gap then it's easy
14:42.30E-bolaManxPower: well idealy there shouldnt be any gaps.
14:42.54E-bolaThey should just hear streaming music without interuptions untill a dial commands reaches an extension that picks up
14:43.11ManxPower-workE-bola: that is non-trivial unless you are good with dialplans
14:43.17E-bola:(
14:43.29E-bolaIt woudl have thought it was a relatively common type of behavior
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14:43.33E-bolaI even
14:43.40ManxPower-workE-bola: what are the two extensions?
14:43.52E-bolaManxPower: 81 and 83
14:44.04ManxPower-workwhich one needs to have a 10 second delay?
14:44.26E-bolaWell both, i mean it should try 81 for 10 secs, then 83 for 10 secs
14:45.00CGMChris[TK]D-Fender: My goal is to call the owner of the phone system (AMI originate) and allow them to record custom greetings for their phone system (AGI).  Am I going about this the wrong way?  My test call is SIP to SIP, rfc2833 for dtmfmode.
14:45.04DavidR2008can anyone point me to instructions on how to install dahdi_dummy?
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14:46.15ManxPower-workDial(Local/8183-delay@context,20,m) then exten => 8183-delay,1,Dial(SIP/81,10) 8183-delay,n,SIP/83,10)
14:46.23ManxPower-workThat will make hold music stay consistant.
14:46.46*** join/#asterisk puzzled (n=foobar@83.163.53.136)
14:47.22ManxPower-workSince the Local/ Dial will provide the hold music
14:47.58ManxPower-workRemember most users are so slow it takes them 10 seconds just to lift the handset.
14:47.59E-bolahmmm
14:48.36E-bolaye i see how it must work, ile try it
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14:48.55E-bolaIm not sure i understand the Local/ part?
14:49.26ManxPower-workLocal/ allows Dial to "dial" extensions instead of devices.
14:49.35ManxPower-worklocalchannel.txt in /path/to/src/asterisk/doc
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14:50.32E-bolaohh cool, thank you
14:50.38E-bolanever stumbled upon that
14:50.50ManxPower-workIts one of the most useful Asterisk features
14:51.21superbeeffinally came up with a fix for IAX dropped calls between new Asterisk 1.4 boxes and old asterisk 1.2 boxes
14:51.59superbeefI ditched speex and switch to G711
14:52.05superbeefbandwidth should be interesting
14:52.08CGMChris[TK]D-Fender: Figured it out.  The system I was calling into was calling Answer() as soon as calls were received, rather than when a real person answered.  My fault!
14:52.53ManxPower-workNEVER Answer() unless you have to.
14:53.35p3nguinIf you're just doing a Dial(), let the phone perform the answering.
14:53.55ManxPower-workDon't use "r" option to dial either
14:53.57CGMChrisYes, I am letting it fall through to Dial now and that seems to work just fine.
14:54.14ManxPower-work"r" stands for "retard", as in "only a retard would use the 'r' option to Dial"
14:55.19p3nguinIf m isn't an option, won't it ring anyway?
14:55.25p3nguinwithout r
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14:55.40ManxPower-workp3nguin: correct
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14:56.26ManxPower-workWell, it will send whatever is correct.  ringback, your call cannot be completed as dialed, number disconnected, the cellular subscriber you are trying to reach is out of the service area, etc
14:57.46*** join/#asterisk ghento (n=ghento@user146-1.wireless.utoronto.ca)
14:57.54p3nguinI need to do some testing on that, now.
15:01.52*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
15:02.36ghentoHi all.  I'm just curious if anyone can recommend a method to detect voicemail?  For example, if a call is answered, I'd like to use call flow A, but if I detect a voicemail, would use call flow B.
15:03.04[TK]D-Fenderghento: "core show application amd"
15:03.08p3nguinDial(SIP/202|40|ktT)  produces a ringing sound in my phone, so there's no real good reason to need to add an r in there.
15:03.45ManxPower-workp3nguin: you want BOTH the call-er and the call-ee to be able to do DTMF transfers?
15:03.57ManxPower-workwhat did you say the number is?
15:04.07p3nguinmanxpower-work: Yeah, it's an internal to internal Dial.
15:04.23ghentoD-Fender: Many thanks!
15:04.38ManxPower-workp3nguin: just making sure.  There is a MASSIVE increase in using unsecured PBXs to make calls for scammers
15:05.12ManxPower-workOnce a person receives their first phone bill 100x what it normally is, they usually secure their PBXs
15:05.31p3nguinmanxpower-work: Ooooooh... I'll go check the dialplan for Ts in the wrong spot and remove them.
15:06.09ManxPower-workJust don't use T when a call comes from an untrusted source like the outside.
15:06.20p3nguinyeah
15:06.36ManxPower-workOr *gasp* Don't use T or t at all and use the transfer button on your phone.
15:07.09p3nguinmanxpower-work: That's fine for real phones, but most of the free softphones have the xfer button blocked out.
15:07.45*** join/#asterisk pirulo (n=andres@65.102.99.5)
15:07.57ManxPower-worksoftphone == cheap pastard
15:07.58p3nguinI'm still going to review and revise based on what you just said, though.
15:08.11p3nguinBetter safe than sorry.
15:09.12*** join/#asterisk gardo (n=gardo@110.55.240.37)
15:09.28p3nguinAlso, do you normally "hop" into the internal context when a call comes in from outside, or do you have the Dial() commands right in the inbound context?
15:09.54*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
15:10.24p3nguinSay someone calls in from outside and enters ext202 in the IVR prompts.  Seems like a perfect opportunity to send that call to internal 202.
15:10.41[TK]D-FenderO>o
15:10.53ManxPower-workI normally have an inbound context to do any pre-Dial processing (change callerID, set the ring cadence, etc), then I route the call via Goto to the correct internal extension in the dialplan for the actual Dial
15:11.18ManxPower-workif it needs to go into an IVR, I sent the call into an IVR context.
15:12.16p3nguinThat's how I'm doing it, too, but I figured I would use this time to also analyze that as potentially being the wrong way.
15:13.46p3nguinexten => _2XX,n,Goto(internal|${EXTEN}|1)
15:14.25p3nguin[tk]d-fender talked me into hard-coding the internal extensions rather than having the same matching scheme in internal.
15:14.47ManxPower-workp3nguin: don't use |  they will go away very soon/
15:14.49ManxPower-workuse ,
15:15.07p3nguinIt won't go away if I don't upgrade to 1.6.  :)
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15:15.45superbeefso what is more useful than show codecs to actually show what codecs i have
15:16.01*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
15:16.50Qwellsuperbeef: core show translations
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15:18.07p3nguinmanxpower-work: I did recently consider converting my pipes to commas for the sake of helping others who do have 1.6 and my syntax not being compatible.
15:18.22p3nguinI might do that today.
15:18.27*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)
15:18.48p3nguinShould only take couple minutes to run the replacement and then visually inspect the result.
15:19.29ManxPower-workJoy.  We have one of our techs, a Broadview tech, and a verizon tech all in the phone closet all doing different things
15:20.14superbeefinteresting
15:20.15p3nguinQuick!  Lock the door!
15:20.26superbeefi wonder why i can't get g726 to work.. maybe because its between asteirsk 1.4 and 1.2
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15:23.07Corydon76-digsuperbeef: some endpoints use the AAL2 alternative encoding of G726 and call it G726
15:23.55superbeefCorydon76-dig: my 1.4 box has g726 and g726aal2
15:24.50Qwellsuperbeef: You're trying to use g726 between Asterisk boxes?
15:25.41superbeefQwell: yep
15:25.45*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
15:25.46Qwellwhy?
15:26.18superbeefbecause speex is causing dropped calls, and alaw has a big bandwidth footprint
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15:28.56superbeefQwell: I upgraded my main server to 1.4.current, but it still sends calls via IAX to some really old 1.2 boxes... I started getting dropped calls after a bit of utilization, and after a week of madness, it finally stopped when I quit using speex
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15:32.42[TK]D-Fender[11:13]<p3nguin>exten => _2XX,n,Goto(internal|${EXTEN}|1) <--- INCLUDE
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15:33.46p3nguinWhat's the reason to include it when it can easily just Goto the internal context?
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15:34.37vaflenHi. I have a question regarding Skype for Asterisk and support for Arm based systems, or rather the lack thereof.
15:34.45Assidheya
15:34.46[TK]D-Fenderp3nguin: Your goto may not have a match to land on.  The contexts of "internal may change.  Syntax on the Goto has already changed.  Whats the point of contexts when you don't use includes?
15:35.02[TK]D-Fenderp3nguin: What if you have multiple patterns there?
15:35.24Assidi have a strange issue.. recently users have been noticing that the calls arent being picked up correctly.. like if i call an extension an they pick up the phone.. it continues to ring
15:35.49Assidi have a diall all policy where i am dialling the sip users all atonce..
15:36.08p3nguin[tk]d-fender: The contents of the internal context will be the same whether I include it to Goto it.
15:36.10Assidso i am guessing the sip signalling is getting messed up.. this is also happening between individual extensions
15:36.13vaflenIs there any chance there might be support for Arm based systems coming to the Skype for Asterisk addon?
15:36.36[TK]D-Fenderp3nguin: It is the equivalent of bad programming, adding complexity and risk for nothing.
15:36.39Assidcan someone suggest where i can look.. this has recently started and was fine till about a month back.. and now its just happening more often
15:36.47[TK]D-Fenderp3nguin: if they dial a # that is not valid you call can drop like a rock
15:37.06Qwellvaflen: You'd have to talk to Digium sales.
15:37.23vaflenQwell: Thanks.
15:37.25Qwellvaflen: If there's enough interest, it's possible something we'd consider.
15:38.10*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
15:38.38Assidi hae the latest bootrom and sip
15:38.49vaflenQwell: I have a small home office where I am currently running Asterisk on a Sheevaplug. I'd like to keep the power consumption to a minimum and the plug is great for that. Asterisk runs very well on it but it would be nice to extend the functionality to include Skype as well.
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15:42.16grandpapadotHi all.  Is there an effective limit to how many peers can be dialed at the same time with Dial(SIP/100&SIP/200...), etc?
15:42.36*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
15:44.17grandpapadot.. rather, a recommended limit?
15:44.31leifmadsengrandpapadot: pretty much the limit is the number of characters, or CPU, whichever you hit first
15:44.47grandpapadotThanks, leif!
15:45.00[TK]D-Fendergrandpapadot: Yes.. you'll encounter a hard limit in the raw # of CHARS that any app can accept
15:45.16[TK]D-Fenderleifmadsen: Char limi hits first by far :)
15:45.32Chainsaw*G* the char limi
15:46.55leifmadsen[TK]D-Fender: assuming you're not running on an embedded device like a Linksys or something, in which case the CPU will die first
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15:51.12Kattyis getting a 19" Lcd!
15:52.43Kattyand i have chicken baking in the oven :>
15:55.16beekKatty: You're at work and are baking chicken?
15:56.12Qwellbeek: where else would you bake chicken?
15:56.23ariel_would love to have 2 23's widescreens on his desk...
15:56.46beekQwell: If I did that here the vultures would start circling...
15:56.49*** join/#asterisk asterwiki (n=asterwik@69.77.169.14)
15:57.19scalex000help,  what is mean this "Spawn extension (from-zaptel, s, 2) exited non-zero on 'SIP/5229-b7c0e870<ZOMBIE>'"
15:57.55p3nguinIt means that the s extension in the from-zaptel context exited at priorty 2.
15:58.06scalex000ok
15:58.08scalex000thanks
15:58.35Kattybeek: we have a full kitchen at work
15:58.48p3nguinThat's awesome!
15:58.49Kattybeek: and i'm going to make some asparagus to go with it in just a little bit
15:58.55*** join/#asterisk knctrnl (n=aembrey@nat/digium/x-qgxwizyrdvnsbmdm)
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15:59.00p3nguinNow you can make me some food.
15:59.08Kattymhmm
15:59.10beekKatty -- I'll hop on a plane and be right out.
15:59.13Kattyk
15:59.28knctrnlj
15:59.30p3nguinI can be there in an hour if I leave.... now.
15:59.44Kattyit will be out of the oven at 11:45
15:59.46Kattyso hurry it up
16:00.16p3nguinI can't do it in 45 minutes, no matter how fast I drive.
16:00.48Katty:<
16:00.53Kattywell i can always give you the recipe
16:00.55Kattywill that help?
16:01.13p3nguinprobably
16:01.15Kattyk
16:02.19diatonic1[TK]D-Fender: I think you were right about the Redfone - This thing has been a PITA and the support is not very responsive. Shipping it back and going Sangoma
16:07.10beekWhen bridging Telco PRI through Asterisk to a legacy PBX via PRI do you usually find that the rx/txgain=0.0 settings adequate?  I'm getting echo and wondering where to start troubleshooting.
16:07.11*** join/#asterisk knctrnl (n=aembrey@nat/digium/x-pmzzenjdqtezkxxb)
16:07.15beekThe echo isn't consistent.
16:07.42ManxPower-workbeek: echo comes from the FAR end analog loop.
16:08.08ManxPower-workSo you won't get echo if the destination is VoIP, PRI, T-1, or Cell phone.
16:08.52beekManxPower-work: That's what I always thought.   I have some complaints from employees that they're hearing themselves.  I thought that the far end is responsible (and I just told someone that).
16:09.22ManxPower-workbeek: ALL calls that terminate in analog have echo.  But without VoIP the echo happens so FAST you can't actually perceive it.
16:09.23*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)
16:09.39ManxPower-workOnce you put VoIP in place latency is high enough that you hear the echo that was there all along.
16:10.07ManxPower-worksort of like in a small room your voice still echos off the wall, but you can't perceive it because it happens so fast.  In a very large room the delay is long enough you could herart.
16:10.18ManxPower-work.... could hear it.
16:10.20*** join/#asterisk voipmonk (n=voipmonk@67.204.45.155)
16:10.20diatonic1ManxPower-work: Is echo cancellation in a T1 card necessary if it's used to connect to PRIs or analog T1s?.. calling from IP phones internally?.. Is there anything to produce echo there?
16:10.34beekHmmm... I have a legacy PBX connected to * via PRI to the Telco  via PRI.
16:10.36ManxPower-workdiatonic1: you have not been reading have you.
16:11.09beekI haven't had a time where I have had that echo problem but some of the employees have.
16:11.10ManxPower-workdiatonic1: echo comes from the far end, so you can't eliminate it unless you put echo canceler on every telephone line you call.  obviously not practical
16:11.31ManxPower-workbeek: don't test it by calling a cell phone or a large company
16:12.16beekManxPower-work: I told the person to let me know when she gets a call like that and I'll snoop around a bit.
16:12.23beekI'm not sure that there's anything I can do about it.
16:12.24diatonic1ManxPower-work Thanks. It seems like I always get stuff with echo cancellation to play it safe, but I'm wondering if it's really necessary
16:12.27ManxPower-worktry calling the same number that called them
16:12.33ManxPower-workdiatonic1: yes it is required.
16:13.09beekManxPower-work: I'll do that this evening.. the employee said that she called in and had echo.   I'll call her this evening.
16:13.29ManxPower-workbeek: the far end had echo (PSTN side)??
16:13.46beekShe claimed that she heard herself and that the inside person did not hear it.
16:16.02*** join/#asterisk chazzm (n=chazz@173-24-217-9.client.mchsi.com)
16:16.08ManxPower-workand she was using an IP phone?
16:16.25beekManxPower-work: I just double-checked with her.   She called the office, our side her the echo.  Thus, it's on her end.
16:16.35beekI told her to call me this evening when she gets home.
16:17.47superbeefYou guys ever end up with asterisk getting an extension stuck adn thinking its busy when its not?    I always end up boucning asterisk to clear it, but there's gotta be a better way
16:18.28asterwikibeek: you can also try enabling echo training 'echotraining=yes' when testing to see if that helps (or disable it its already enabled);
16:18.54*** join/#asterisk JoeMoretti (n=jmoretti@76.164.171.81)
16:19.16beekasterwiki: This is HWEC... I thought that parameter was for SWEC
16:20.03*** join/#asterisk cesar_CR (n=cesar@201.192.86.30)
16:20.25asterwikibeek: that parameter is normally in your zapata.conf
16:21.05beekasterwiki: chan_dahdi.conf in this case.   The docs state that this is for SWEC, which I'm not using.
16:21.11beekor am I reading that wrong?
16:22.45ManxPower-workwith HWEC on digium cards I believe the echocancel= option is ignored.  You should test.
16:23.04ManxPower-workDigium does a terrible, terrible, job of informing the user which EC is in use.
16:23.08bpgoldsbIn voicemail.conf, if I don't configure an email address for a mailbox, will Asterisk just not send an email notification?  Or will it do something bad?
16:23.23beekbpgoldsb: those are optional
16:23.42bpgoldsbSo just not send?
16:24.12chazzmto use HWEC you still have to have echocancel=yes
16:24.21beekbpgoldsb: correct.
16:24.22chazzmor some number, which will just be handled as a yes for HWEC
16:25.07beekchazzm: I have echocancel=yes, but 'echotraining' is ignored if you're using HWEC according to the docs.
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16:25.36beekI'm not sure about 'echocancelwhenbridged' though...   Any ideas there?
16:25.49chazzmcorrect, but sometimes it is better to comment out echotraining when using HWEC. I think that depends on what version of zaptel you are using though
16:26.06beekchazzm: DAHDI here
16:26.08chazzmI believe echocancelwhenbridged still applies, but it is normally recommended to set it to no anyway
16:26.20beekchazzm: which is what I have set
16:26.25ManxPower-workbeek: you almost never need to EC when TDM is bridged.  The latency is low enough nobody should hear echo
16:27.03beekManxPower-work: It's off, so we're good there.
16:27.24beekAhh... the lunch bell has been rung.   I'm heading out for chinese today.   Thanks guys!  Back later.
16:29.25*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
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16:33.15ariel_#bitos
16:33.20ariel_argh
16:34.06ariel_Anyone heard of BitOS's asterisk based pbx?
16:43.30*** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright)
16:45.00Gido-Eariel_ url orso?
16:46.01angryuseris there any way to force PAP2T send external ip in Contact field instead of local ? thanks
16:47.16*** join/#asterisk telnettech (n=telnette@71.5.32.196)
16:47.51hardwireurl or it never happened
16:47.58*** join/#asterisk friartuck (n=pmccary@66.162.90.57)
16:48.01ariel_Gido-E: it's actually from a co. called bitrail.net
16:49.08ManxPower-workangryuser: why would you do that?
16:49.47*** join/#asterisk andres833 (n=andres83@201.244.125.6)
16:51.19angryuserManxPower-work,  i got    client <>nat <nat> server   ports routed to server, client registers, but when invite comes audio is sent to local clients ip, i have specifyed external ip to pap2, but still not working, maybe you know adress of a free stun server ?
16:51.35*** part/#asterisk pietro (n=pietro@nat.xsec.it)
16:51.48ManxPower-workangryuser: why not put nat=yes in the sip.conf for that peer?
16:52.30angryuserManxPower-work, it is set
16:52.56ManxPower-workangryuser: then Asterisk will ignore the IP info inside the packet and use the info from the packet header.
16:54.01angryuserManxPower-work, look like not all the way : Peer audio RTP is at port 192.168.1.5:16480  (if server replaces 1.5 by ext it will work)
16:56.56ManxPower-workangryuser: then you have some other problem
16:57.00Gido-Eariel_ anny clue about SVN server or track system or technical docu?
16:57.08ManxPower-workmaybe the incoming call is not matching the peer you think it matches
16:57.35ManxPower-workIs your server behind nat?
16:58.41angryuserManxPower-work, basicly he gets the invite from pap2t bu audio is sent locall (remote) ip my server is behind nat, the sip mach is good 100%
16:59.11ariel_Gido-E: no idea, just trying to get more info about them or it, I have a customer that wants that.
16:59.20ManxPower-workso you have localnet= and externip= set?
16:59.53*** join/#asterisk pta200 (n=paolo@goose.specialai.com)
17:00.49*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
17:00.57*** part/#asterisk Joel (n=jjshoe@wsip-70-183-82-162.sd.sd.cox.net)
17:01.01*** join/#asterisk ryduh (n=ryduh@204.16.143.186)
17:01.03angryuserManxPower-work, yes, i am sending correct external ip when trying to reach that peer, but to incorrect peers ip
17:01.28ManxPower-workmaybe SIP ALG on your router is screwing it up.
17:01.38angryuserManxPower-work, hm, maybe
17:03.26ManxPower-work"Yahoo Offered Lap Dances At Hack Event"  I guess that's one way to make an...impression.
17:04.29pta200Why is that if you pass the meetme application the wrong conf room/pin number and it jumps to the 'h' extension after telling playing the conf-invalid file, if you try to playback any other audio file in the 'h' extension  Asterisk will throw a file.c:747 ast_readaudio_callback: Failed to write frame and not play the file? I've seen this in 1.4.24-1.4.26.2
17:06.33ManxPower-workpta200: do you have a priority after the broken meetme line?
17:06.58ManxPower-workif not, that would explain the channel hanging up
17:08.42pta200I have a hangup priority, but the cli show asterisk going to the 'h' extension and running through priorities sequentially, I'll pull ot the hang up priority in the meetme call and try that
17:09.49ManxPower-workno, you don't have a "hangup priority"  You have a hangup extension
17:10.12ManxPower-workput a line after the meetme to handle when meetme fails
17:13.19pta200That doesn't seem to make a different app_meetme skips to the hang up extension and fails trying to play an audio file back regardless of there being a priority after the failed called to the meetme application
17:13.25ChainsawQwell: I've got an easy one for you, kernel 2.6.32 API change wrt driver_data: https://issues.asterisk.org/view.php?id=16114
17:13.32ChainsawQwell: Patch, license OK, etc.
17:14.09pta200This the CLI output
17:14.50pta200DEBUG[10196]: app_meetme.c:2587 find_conf: 2222 isn't a valid conference
17:14.50pta200-- <SIP/0004f211f05c-0a326d80> Playing 'conf-invalid' (language 'en')
17:14.50pta200== Spawn extension (meetme-conf, login, 2) exited non-zero on 'SIP/0004f211f05c-0a326d80'
17:14.51pta200-- Executing [h@meetme-conf:1] Playback("SIP/0004f211f05c-0a326d80", "sorry") in new stack
17:14.51pta200WARNING[10196]: file.c:747 ast_readaudio_callback: Failed to write frame
17:14.51pta200-- <SIP/0004f211f05c-0a326d80> Playing 'sorry' (language 'en')
17:14.53pta200WARNING[10196]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/0004f211f05c-0a326d80 for sorry
17:15.08Chainsaw~pb
17:15.09infobot[~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://bin.cakephp.org/ , or apt-get install pastebinit
17:15.56Kattyso far i am still severely lacking in vitramin e, b1, b2, b6, b9, copper, and manganese
17:16.28telnettechkatty: need to start taking your daily geritol it sounds like
17:16.28ManxPower-workn00bs flooding the channel, this looks like a Monday
17:16.34*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
17:16.40Kattytelnettech: well we'll see where i'ma t after dinner.
17:16.40Nuggettelnet is eeeeeeevil!
17:17.02telnettechgood day (turd) nugget
17:17.30ManxPower-workNugget: so is microsoft, but lots of poor sods still use their products.
17:18.13Kattynugget has a telnet script.
17:18.19Kattyi doubt if he's really there
17:19.43*** join/#asterisk Jankooo (n=jani@dsl5402AECF.pool.t-online.hu)
17:20.00Kattyi've hit, or gone over, on the recommendation of b3, k, phosphorus, sodium, and selenium
17:20.31Kattynot a surprise on the sodium there.
17:21.00telnettecheverything seems to have sodium
17:21.08Kattyyeah :<
17:21.18[TK]D-FenderKatty: Don't forget the arsenic, lithium, and lead!
17:21.31Katty:P
17:21.45Kattynow this is interesting
17:21.54Kattya deficiency of selenium can cause dandruff
17:22.29Kattyand loose skin.
17:22.33Kattylooks at arm.
17:22.49Kattywhat does loose skin look like ^_-
17:24.26[TK]D-FenderKatty: http://farm2.static.flickr.com/1326/671376073_712b66aaaf_o.jpg
17:24.39[TK]D-FenderKatty: Oh.... and sorry :)
17:24.56*** join/#asterisk jantypas (n=jantypas@42.sub-75-208-27.myvzw.com)
17:25.12Kattyohh
17:25.13Kattymy.
17:25.16*** join/#asterisk mchou (n=quassel@unaffiliated/mchou)
17:25.24Kattycloses tab.
17:25.34Kattywell i certainly don't have that problem.
17:25.59[TK]D-FenderKatty: Now... what to take to deal with your neurosis.... hmmmm
17:26.00*** part/#asterisk pta200 (n=paolo@goose.specialai.com)
17:26.17Kattywell, if i had to guess.
17:26.21Kattybased on current information
17:26.24Kattyb1, b2, and b6
17:27.24[TK]D-Fenderfacepalms
17:27.37Katty*hee*
17:28.36*** join/#asterisk Blackvel (n=blackvel@84.57.75.0)
17:30.02ManxPower-workKatty: you are alive so things can't be too bad.
17:30.09Blackvelhi all. anyone with an isdn pbx (e.g euracom) AND patton smartnode media gateway 4634? I have both connected to (same) isdn ntba. now I get a problem with "busy on busy" and telco as long as patton s0 cable is plugged in (patton steals d-info information)
17:30.31ChainsawBlackvel: I do have a Patton Smartnode 4634, but mine is attached to two regular ISDN BRIs.
17:31.11BlackvelChainsaw: what could be the reason that patton "steals" from d-channel?
17:31.30ChainsawBlackvel: Configuration issue, most likely.
17:31.43Blackveli really thought both would be independent
17:31.58ChainsawBlackvel: A configuration issue *on* the Patton.
17:32.01Blackvelwhen i concentrate on msdn
17:32.04Blackvelmsn
17:32.18Blackvelsomehow my patton sees my whole ntba bus :(
17:32.48Blackvelyou do not have some other isdn equipment connected in parallel to ntba?
17:33.14ChainsawBlackvel: No, I have two ISDN BRIs straight from British Telecom plugged into the 4634.
17:34.31ChainsawBlackvel: I must warn you for the Patton. It likely has R4 firmware on it now. If you upgrade to R5, *nothing* will work anymore.
17:35.05ChainsawBlackvel: Well, everything you care about like SIP will be thoroughly shafted. It seems the ISDN side remains working.
17:35.29Blackvelright. i invested ONE week in 2008 to get R5.2 build running
17:35.43Blackvelworks quite well..besides "busy on busy" stealing and wrong telco message
17:35.55Chainsaw*nod* A week sounds right. I'm on R5.4
17:36.24p3nguinHow do I specify which conference number to use when using Page()?
17:36.31Blackvelmy telco says on busy on busy: this dailed number is right now not assigned
17:36.44Blackveland it is the patton which lets telco talk in that way
17:36.52Blackvelif i unplug patton everything is fine
17:37.01Blackvelwell probably a little bug in config :)
17:37.17Blackvelis R5.3/R5.4 quite different?
17:37.24Blackvelis it save to upgrade?
17:37.31ChainsawLikely. To rule out a bus termination problem you could connect only one device at a time.
17:37.33*** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com)
17:37.48ChainsawCompared to R4->R5 it is effortless, yes.
17:38.10Blackveldo you have an echo problem from time to e.g (e.g 5-10% in all cases)?
17:38.19ChainsawNo echo, no.
17:38.26Blackveli know its always the other party (i hear me in echo locally)
17:38.35[TK]D-Fenderp3nguin: You don't.  Its dynamic
17:38.36Blackvelbut patton / snoms can not compensate the inbound problem
17:38.36ChainsawNeither on the Cisco handsets nor on the analog handsets (on a SmartNode 4118).
17:38.51Blackveli really hate that...i thought even inbound can be compensated
17:39.31Blackvelhow would you separate patton / euracom isdn on a nbta bus?
17:39.42Blackveltey share d-channel?
17:39.44Blackveltey
17:39.46Blackvelthey
17:39.57*** join/#asterisk thansen (n=thansen@76.27.110.194)
17:40.01ChainsawNormally your NT1 has two outputs.
17:40.10Blackvelright
17:40.25Blackveland they are separated?
17:40.35ChainsawIt could be one of those annoying faulty ones, in which case you'll want to use a 1-to-2 splitter on the port you were already using.
17:40.52ChainsawBut as I said, the simple test would be to disconnect the other device, connect only the Patton and then make some test calls.
17:41.07Chainsaw(I realise you may have to do that out of hours when the PBX isn't needed for other things)
17:41.48*** part/#asterisk ManxPower-work (n=ewieling@24.42.221.26)
17:44.07ryduhhello everybody!
17:44.22ChainsawHi Dr. Nic.. eh, ryduh.
17:44.51Blackvelchainsaw: pardon? nt1 could be faulty?
17:45.04*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
17:45.18Blackvelwhat are those splitters doing you can use for one port?
17:45.20ChainsawBlackvel: The secondary port on it could be. I've had that. The engineer who installed it stuck a T-splitter on the working port and left.
17:45.26*** join/#asterisk oej (n=olle@ns.webway.se)
17:45.36Blackveli guess they all share d-channel information. that is the problem in my case (i think)
17:45.38loubenSo Asterisk isn't supporting h323 and its newer versions anymore... Maybe are plans for the future?
17:45.41ChainsawBlackvel: But still, I'd test *just* the Patton *without* the PBX.
17:45.54Blackvelwell it works
17:46.05Blackvelit pickups my 2 business numbers (on ivr / snom)
17:46.16Blackvelalso my isdn pbx works (euracom)
17:46.31Blackvelbut not for busy on busy (opposite to call waiting)
17:47.00Blackvelit makes tuttuttuttuttut when msn is busy on euracom (without patton)
17:47.45Blackvelwith my patton connected its no busy signal anymore but that weird telco announcement that this number is not assigned. i really dont get it ;)
17:47.51DavidR2008I'm trying to fix this error:
17:47.53DavidR2008Unable to support trunking on user 'asteriskm' without a timing interface
17:47.55DavidR2008I have dahdi_dummy installed and the pseudo channel shows up when I issue a dahdi show channels, what do I need to do?
17:48.12BlackvelChainsaw: how do you use the routing table? route <msn> ...?
17:48.21Blackveli used that for my both msn's
17:48.39Blackveleven tried "route default none" this evening
17:48.42ChainsawBlackvel: I'd have to look it up. I think you'll be best off contacting Patton about it.
17:49.08Blackveli even forgot how to damn login into my patton to enable isdn tracing
17:49.17Blackvelforgot root / shell password
17:49.18Blackvelhaha
17:49.25ChainsawNo configuration file backup?
17:49.28Blackvelwell yes, probably
17:49.32Blackvelyes i have
17:49.33ChainsawJust do what I did.
17:49.37Blackveli can even login into gui
17:49.42ChainsawClear it out with the reset button, set a password you like.
17:49.43Qwelllouben: Where did you get this information from?
17:49.50ChainsawThen string the two configuration files together.
17:50.03ChainsawBlackvel: If you can log into the GUI you can log into the telnet side of it.
17:50.19*** join/#asterisk slinksh0t (n=slinksh0@98.64.206.62)
17:50.22*** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net)
17:50.28loubenQwell, we had a discussion with Corydon76-dig about that issue
17:51.13Kattyblinks.
17:51.27Kattyleifmadsen: holy crap, have you seen all the vitamins in Total?
17:52.24Blackvelchainsaw: what was the shell login user? Administrator (like gui)?
17:52.31ChainsawBlackvel: Yes, administrator.
17:52.40ChainsawBlackvel: It may require you to write it as admin there though.
17:52.46Blackvelah
17:52.50Blackvelcould be
17:52.59Blackvelhow does your msn routing table look like?
17:54.23Blackvelyou just put your business msn(n's) in which you like to pickup for incoming call?
17:55.41ChainsawBlackvel: I do not have a degree in ISDN terminology.
17:55.41leifmadsenKatty: yes, but there is also a lot of sugar last I saw
17:55.54leifmadsenKatty: or maybe I'm thinking of Vector
17:56.48ChainsawBlackvel: "MSN"
17:57.29Blackvelexternal number
17:57.33Blackvelcalled number
17:57.42Kattyleifmadsen: i don't know. but it seems to contain all of the vitamins i'm lacking in.
17:57.46Blackvelcalled e164
17:57.55*** join/#asterisk d1ss3nt (n=nebula@s0up.digitalkharma.org)
17:58.19d1ss3ntcan someone recommend a reliable single span T1/PRI card for asterisk that is not a digium?
17:58.27leifmadsenKatty: I'd check the sugar count on it to make sure it isn't too high -- sometimes they add a lot of that in order to make it "taste good"
17:58.44ChainsawBlackvel: Ah, right. I check the end digits.
17:59.11ChainsawBlackvel: And depending on whether it's a locally attached analog device, remotely attached analog device (other office) or a regular extension, send it on its way.
17:59.39Kattyleifmadsen: 4.5g of sugar in 3/4c
18:00.04kaldemard1ss3nt: sangoma ones are widely used
18:00.06*** join/#asterisk knctrnl (n=aembrey@nat/digium/x-haolasdondnmkbpi)
18:00.48leifmadsenKatty: that doesn't sound like too much
18:01.10Kattyleifmadsen: no, it doesn't...but the real question is do i really want to stuff myself with all the additives and preservatives listed on the ingredient list.
18:01.55leifmadsenKatty: that's the key question
18:02.10Kattyleifmadsen: i should probably just eat more lentils, bananas, and tuna.
18:02.21leifmadsenideally you'd just make your own cereal -- oatmeal is good too, especially if you add some cinnamon, nuts, seeds, and berries
18:02.30leifmadsenKatty: berries too
18:02.40Kattymmmmmmmmm, berries.
18:02.45leifmadsenblueberries are amazingly good for you
18:02.49Kattyyes, yes they are.
18:02.52leifmadsenanti-oxitant
18:02.56theharmmmmm blueberries
18:03.20Kattysalmon also has quite a bit of b12
18:03.28Kattymore so than tuna, actually
18:03.54Kattysounds like Salmon patties might be on the menu soon
18:04.10ryduhleifmadsen: antioxidant?
18:04.16Corydon76-diglouben: NO.  What I told you is that we don't have experts on the code, so if you have trouble with compiling, you're best to ask a person who has worked on that code.
18:04.31Corydon76-diglouben: specifically, I told you to contact the person who wrote the most recent patch.
18:04.44loubenCorydon76-dig, I did
18:05.23Kattyryduh: oxidation of cells leads to Free Radicals
18:05.34Corydon76-diglouben: did that work?
18:05.36Kattyryduh: bits of damaged molecules.
18:05.47loubenNot really...
18:05.53Kattyryduh: specifically sells that are lacking an electron
18:06.00Kattyryduh: so they run around stealing electrons from other cells.
18:06.11QwellNot the electrons!
18:06.16Kattyyes.
18:06.19ryduhKatty: Sorry, no I was correcting his spelling/hypenation
18:06.58Kattyryduh: protect your electrons, eat a blueberry.
18:07.13Jumpiehmm im tryin to think of a really annoying wav file to play on blacklist match
18:07.20loubenh323 should be mandandory for Large scale implementations
18:07.32Corydon76-digJumpie: tt-chickens
18:07.35ChainsawJumpie: Unannounced echo test.
18:07.47JumpieCorydon76-dig im scared to ask...
18:07.47loubenThat is what I was told
18:07.57ryduhwe-dont-have-tech-support
18:08.02ChainsawJumpie: I take it is some anti-telemarketing measure.
18:08.09JumpieChainsaw yeah
18:08.13Jumpiei have the same 5 people that call me daily
18:08.18Jumpieone is also about a studen loan of my ex wife
18:08.18Corydon76-diglouben: That's an interesting opinion
18:08.20Jumpiethat is particular nasty
18:08.27ChainsawI take unannounced echo test will freak them out more then a .wav
18:08.33kaldemarJumpie: a simple "hello" with 20 seconds of silence and then a hangup. works like a charm every time.
18:08.37Corydon76-diglouben: it's contrary to virtually the entire voip industry
18:08.39Jumpiekaldemar lol
18:08.49Jumpiei cant find tt-chickens
18:08.53Chainsawkaldemar: I approve! A .wav with hello and *then* the echo test.
18:09.02Chainsawkaldemar: It'll feel like a bad line.
18:09.15Corydon76-digJumpie: look in the sounds-extras
18:09.18Jumpieim an idiot..how do i test what that echo test is?
18:09.36kaldemarChainsaw: pure awesomeness
18:09.37ryduhJumpie: you could record one saying they have been permanently blacklisted and to not bother calling back
18:10.08Chainsawkaldemar: exten => s,n,Echo
18:10.16Chainsawkaldemar: If they hit a # it'll continue.
18:10.20Chainsawkaldemar: So you can stick more behind it.
18:10.33JumpieChainsaw....as long as it wont tie up the channel indefniitely
18:10.37Jumpieindefinitely
18:10.38Corydon76-digJumpie: tt-hangup is another good one
18:10.38ryduhChainsaw: does that just echo everything they say until they hit a #
18:10.42loubenSo is asterisk going to stop it?
18:10.42Naikrovekit'll hang up when theyd o
18:10.48Chainsawryduh: It does.
18:10.52Jumpierofl ryduh
18:10.55Chainsawryduh: Or until they hang up.
18:11.00Corydon76-digOh, and tt-chickens was renamed to nobody-but-chickens
18:11.19Corydon76-diglouben: stop what?
18:11.38ryduhJumpie: what about a hello, Wait(10) then an Echo
18:11.43loubenCorydon76-dig, supporting h323
18:11.46Jumpiei like that ryduh
18:11.55Chainsawryduh: I'd start echoing immediately. Ever been on a bad speakerphone?
18:12.01Corydon76-diglouben: that depends upon what you mean by "support"
18:12.08ryduhChainsaw: maybe?
18:12.23Chainsawryduh: It'll sound just the same to them. They'll be talking for up to a minute.
18:12.25ryduhbrb- testing Echo
18:12.37Chainsawryduh: It's like a TCP tarpit for telemarketers.
18:13.03loubenCorydon76-dig, I mean functionality of * with the latest features of h323
18:13.37loubenCorydon76-dig, ...that are written for openh323
18:13.41Corydon76-diglouben: There are no plans to enhance H323, no
18:13.42Jumpiei love it
18:13.49Jumpieim tryin to think if i wanna do echo immediately or not
18:13.58*** join/#asterisk coppice (n=chatzill@host86-132-26-86.range86-132.btcentralplus.com)
18:14.02*** part/#asterisk diatonic1 (n=chillman@208.186.73.35)
18:14.03Corydon76-digThat could change in the next 5 minutes, 3 months, or never
18:14.46loubenI see. Damn we would like to have this functionality
18:15.06Corydon76-digThen again, this is open source.  If YOU want to contribute that, then we're happy to take those contributions, as long as they fit within our architecture
18:16.00ryduhChainsaw: Is there a way where I could use something like EAGI to autotune the caller?
18:16.05ryduhChainsaw: I think that would be quite fun.
18:16.07Corydon76-digThat is, as a multi-protocol back-to-back-user-agent
18:16.37Chainsawryduh: I'm not sure, I've not used that. I like your creativity though :)
18:17.04loubenCorydon76-dig, maybe. Now I'm looking forward for this
18:17.14Jumpieman..this is awesome
18:21.02*** join/#asterisk bluOxigen (n=asad@119.73.68.102)
18:21.22ryduhI'm going to create a quick poll. How many of you have a dedicated server in house for asterisk? How many of you are using a VPS or dedicated host somewhere else? Obviously this is more for VoIP.
18:21.41Chainsawryduh: Dedicated, in-house.
18:22.13*** join/#asterisk i-pink (n=my-pink@bzq-79-177-64-90.red.bezeqint.net)
18:22.55i-pinkhii all
18:23.09ChainsawHi there.
18:23.16beekryduh: ditto Chainsaw
18:23.39i-pinki need help with make conference room base on sip
18:25.13ChainsawDo you mean a dial-in conference bridge?
18:25.45*** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802)
18:26.11ryduhFailrar: where does the inspiration for your nick come from?
18:26.24*** join/#asterisk aleshus (n=aleshus@81.0.207.104)
18:27.21Failrarno idea
18:27.32aleshusHi to all.. Can anybody help me with asterisk? I need to configure it with authenticate users from LDAP..
18:28.12[TK]D-Fenderryduh: Vast majority = inhouse
18:28.18ryduhFailrar: inspirationfail
18:28.28*** join/#asterisk QaDeS (n=mklaus@p4FC72A2F.dip0.t-ipconnect.de)
18:28.37ChainsawMine's based on my favourite Doom II weapon.
18:28.43ChainsawComputer games moved on but the name stuck :)
18:28.55[TK]D-Fenderrider : hukt on fonix fale
18:28.58asterwikiryduh:dedicated in-house
18:30.09loubenis away: I'm busy
18:30.38Chainsawwatches to channel grind to halt as everyone waits for louben to return from his errand
18:30.51aleshusnobody have an experience?
18:32.39loubenChainsaw, sorry I'm also chatting in other networks
18:32.52Qwelllouben: turn off your public away message
18:33.27Blackvelchainsaw: my experience from patton debug: it gets the call on the NT1 and disconnects it again:  Cause  NoRouteToDestaination
18:33.31loubensure
18:33.42BlackvelChainsaw: to me the debug is the same (for #1 new call or #2 call where 'busy on busy' should follow
18:34.14*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:34.37Blackveli am debugging layer2/layer3
18:35.07*** join/#asterisk adam0563 (n=damagele@70.103.115.194)
18:35.07Blackveli can not find any information from euracom about "busy on busy" and an special release cause on patton :(
18:35.59Blackvelso when patton does not pickup the #2 call - euracom should send busy. still figuring out where the telco gets the returncode about a different release cause message
18:36.08Chainsaw"Busy on busy" has no meaning to me in the first place.
18:36.35*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
18:37.14adam0563are certain channels assigned numbers?
18:37.25adam0563or a group of channels?
18:38.54p3nguinWhat am I doing wrong here?  http://pastebin.ca/1638014
18:39.42BlackvelChainsaw: "busy on busy" means when the dailed number is busy (because phone1 picked up) it should send busy instead let phone 2/3 ring
18:39.53Chainsawp3nguin: That "unable to open pseudo device" suggests you have no DAHDI timing.
18:40.27Blackvelor let phone1 handle call waiting (knock knock)
18:41.42p3nguinchainsaw: I'm running purely SIP and IAX2, and the call to Page() is SIP to SIP.  What does dahdi have to do with anything in this case?
18:41.58Chainsawp3nguin: In order to do conferencing, a timing source is required.
18:42.17Chainsawp3nguin: You can do that without further hardware, but you'll have to install the "dummy" driver of DAHDI (or Zaptel, if your Asterisk is that old).
18:42.17p3nguinOkay, so I should probably look into ztdummy, right?
18:42.26ChainsawIndeed.
18:42.53p3nguinAsterisk 1.4.24.1
18:43.07Chainsaw*waggles hand* Either one would work.
18:43.41*** join/#asterisk madsara (i=madsara@2001:328:2002:f159:0:0:0:1)
18:43.50kaldemarapp Page needs dahdi or zaptel timing. go for dahdi since 1.4.24.1 supports it.
18:44.25madsaraHey, what's the easiest way to prepend a wav file to a voicemail, so when it's checked a prefacing audio snippit of "This is a so-and-so-customer" plays.
18:45.02ryduhAnother poll. For your dedicated inhouse server, What do you run as your OS? Centos? Or something packaged like trixbox?
18:45.30madsararyduh: Centos
18:46.08p3nguinryduh: FreeBSD
18:46.11Qwell~polls
18:46.12infobot"Does anyone have X or use Y?" is taking a poll, not asking a valid question. Don't do it or our army of rabid weasels will hurt you. Usually, people other than those with the exact same set up can help you and those who have sometimes will not be able to help you. Also see <ask>
18:46.30asterwikiryduh: Several inhouse, distributed several servers (CentOS, Ubuntu, FreeBSD)
18:46.36Chainsawryduh: Gentoo (non-multilib AMD64 hardened)
18:47.23ryduhQwell: I'm not asking who runs what to then ask them a question. I'm genuinely interested in what people are running. Is that forbidden?
18:47.40BlackvelChainsaw: is layer2 d-channel and layer3 b-channel or quite different than that?
18:47.52Qwellryduh: If all 254 people in the channel answered, that would be bad.
18:48.00ChainsawBlackvel: D-channel has your call setup information.
18:48.03ChainsawBlackvel: B-channel carries voice.
18:48.04[TK]D-FenderMultiplied byt he number of questions...
18:48.09*** join/#asterisk clyons (n=clyons@unaffiliated/clyons)
18:48.36p3nguincarry the 4...
18:48.36ryduhQwell: good point. Is there a better way to ask people what they are running? Maybe linking to a poll on a website?
18:48.44p3nguindivide by Pi...
18:49.14BlackvelChainsaw: i can debug isdn layer2 and layer3. are they both d-channel?
18:49.16Qwellryduh: how about researching it using information that's widely available?
18:50.06ryduhQwell: fine. I'll stop the polls.
18:50.09ChainsawBlackvel: As I said before, I do not have a degree in ISDN. I just send calls over it to BT. And sometimes they send me some back, if they're in a good mood. That's it.
18:50.57*** join/#asterisk ManxPower-work (n=ewieling@24.42.221.26)
18:51.12BlackvelChainsaw: our philosohpy looks like to be the same :)
18:52.09Blackvelhow long did it take you to get the patton setup up and running?
18:52.15ChainsawQwell: Did you see my bug? Or is DAHDI not your area?
18:52.29*** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net)
18:52.29ChainsawBlackvel: I inherited this setup, working, on R4.
18:52.41ChainsawBlackvel: It took me about a week to convert it all over to R5.2 and have it working properly.
18:53.59Blackvellooks like 0->100 is not too bad in one week :)
18:54.56Blackvelhow many people use pattons even?
18:55.16ChainsawI think most just pop in a Sangoma card to be honest.
18:55.43*** join/#asterisk TimToady_ (n=moi@adsl320-234.kln.forthnet.gr)
18:55.51ChainsawIn my case it made sense from a failover point of view. There's 2x ISDN BRI and 8x analog FXO.
18:56.07adam0563how do I change the number that DAHDI uses?
18:56.08ChainsawIn both offices. If the connection to Asterisk fails, the calls go straight from ISDN BRI to the analog handsets.
18:57.12[TK]D-Fenderadam0563: Uses for what?  What kind of channel?
18:58.00adam0563we have a few numbers assigned to us and I need to change the number that goes out on our asterisk box
18:58.03BlackvelChainsaw: where do you connect your analog phones? fxo box? fxo connected to asterisk?
18:58.17ChainsawBlackvel: Patton 4118 gateway.
18:58.29ChainsawBlackvel: Both of the gateways connect to Asterisk in normal operation.
18:58.30[TK]D-Fenderadam0563: what kind of channels?
18:58.38ChainsawBlackvel: It's just the fallback path that they talk directly.
18:59.02*** join/#asterisk titter (n=titter@c-98-208-156-75.hsd1.fl.comcast.net)
18:59.04BlackvelChainsaw: i checked sangoma before and noted that the drivers didnt always enable HWEC. just wanted to get around problems with my little epia via nemiah 1gig asterisk server (maybe too small for sw echo cancel)
18:59.19ChainsawBlackvel: ...whoa?
18:59.56Blackveli guess patton is not a bad thing....if you have time to or support to configure. when its done it works :)
19:00.10*** join/#asterisk lewellyn (n=lewellyn@greenviolet/lewellyn)
19:00.25ChainsawBlackvel: Once you're done configuring it, it's definitely one of those "Nobody $#@* touch that. Ever!" devices, true.
19:00.56p3nguin[tk]d-fender: I still can't see any valid reason to include internal if there is a goto() that sends to call to the internal context.  It just doesn't make any sense.  I tried it the way you wanted me to do it and the include never gets touched because the goto() sends the call to a completely different context.
19:01.01adam0563D-Fender: not sure what you mean by type of channel. We are connected through a T1 line so we use DAHDI
19:01.28[TK]D-Fenderp3nguin: it doesn't get touched because you put an overlapping pattering in the cotext you're doing your include in <-
19:01.35[TK]D-Fenderp3nguin: You are creating the conflict
19:01.50[TK]D-Fenderadam0563: what SIGNALING?:
19:02.10Chainsawhands Fender the megaphone
19:02.25[TK]D-Fenderreaches for his ClueBat (tm)
19:03.15adam0563pri_cpe
19:03.30p3nguin[tk]d-fender: If I have a Goto(), there's no logical reason to use an include.  If I use an include, there's no logical reason to use Goto().
19:03.38Blackvelso you have 4118 connected to asterisk and another patton isdn bri gw connected to NT1, asterisk and for backup directly to 4118 (over network routing)?
19:04.01[TK]D-Fenderp3nguin: Are you ever going to put something in [internal] that the menu won't have access to?
19:04.11[TK]D-Fenderadam0563: then jsut set the CALLErID()
19:04.20p3nguin[tk]d-fender: I doubt it.
19:04.57p3nguin[tk]d-fender: The internal context is purely for "local extensions" to be able to be dialed and reach another phone in the system.
19:05.06[TK]D-Fenderp3nguin: then all you'll end up doing is running the risk of dropping calls due to lack of a pattern match on your range, and missing things that don't fit into the goto you have.
19:05.31[TK]D-Fenderp3nguin: p3nguin and slowing down dialplan execution but wassting time on processing an app
19:05.58p3nguin[tk]d-fender: So are you advising that I can include internal and remove the Goto(), right?
19:05.59[TK]D-Fenderp3nguin: the point of includes is so you don't run into the myriad kinds of stupidity you can create with Goto.
19:06.26ryduhWhy would this context not recognize when I press an extension? http://pastebin.com/d54d1b7c8 I can press 8 and it will not recognize it
19:06.29[TK]D-Fender"Oh shit, I need extens in the 300 range!  Now I haev to modify 5 IVR's!"
19:06.57[TK]D-FenderRybad DTMFMODE on the incoming channel
19:07.02[TK]D-Fenderryduh: bad DTMFMODE on the incoming channel
19:07.46*** part/#asterisk JoeMoretti (n=jmoretti@76.164.171.81)
19:07.53ryduh[TK]D-Fender: I set dtmfmode=rfc2833 in sip.conf. I'm using a call file to originate this call
19:08.15[TK]D-Fenderryduh: I don't see a call to look at with SIP DEBUG...
19:08.17[TK]D-Fender~pb
19:08.18infobot[~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://bin.cakephp.org/ , or apt-get install pastebinit
19:08.20[TK]D-Fender^^^^^^
19:08.33ryduhk
19:08.34p3nguinSee, for that scenario using my current configuration with the Goto() in place, I would just change  exten => _[1-2]XX,n,Goto(internal,${EXTEN},1)  to  exten => _[1-3]XX,n,Goto(internal,${EXTEN},1)
19:08.47p3nguinNow I can dial 100s, 200s, and 300s.
19:08.52[TK]D-Fenderp3nguin: and update 3-4 menus that may reference it.
19:09.07p3nguinponders
19:09.09[TK]D-Fenderp3nguin: Your way asks for trouble, wastes processing and adds work.
19:09.28[TK]D-Fenderp3nguin: there isn't a single stated advantage to using a Goto, and tons of bad.
19:09.45[TK]D-Fenderp3nguin: the entire point of contexts is to use includes to build a heirarchy
19:09.47p3nguinI'm trying to weigh that factor.
19:09.54[TK]D-Fenderp3nguin: so that you avoid duplication.
19:10.14[TK]D-Fenderp3nguin: And you're doing your best to think backwards
19:11.34adam0563Fender: how do I set the caller ID on an outbound DAHDI call?
19:11.46p3nguinI'm doing my best to logically rationalize everything.
19:11.48[TK]D-Fenderadam0563: "core show function CALLERID" <-
19:12.06[TK]D-Fenderp3nguin: its the "rational" part you're having trouble with ;)
19:12.45p3nguinAt least my logic is still good.
19:13.44ManxPower-workI think I may have figured out an explanation for how FreePBX works.  It was written by drunken college students after they dropped some really good LSD.
19:13.46[TK]D-Fenderp3nguin: or more like a well formulated bad idea ;)
19:14.24[TK]D-FenderManxPower-work: Lick the frog instead of clicking it ;)
19:14.28DavidR2008how reliable is res_timing_pthread?
19:15.09ryduhis it normal for AMI to take 10-15 seconds before originating a call?
19:15.44*** join/#asterisk denon (i=denon@sassinak.net)
19:15.44*** mode/#asterisk [+o denon] by ChanServ
19:15.59adam0563D-Fender: exten => _801NXXXXXX,3,Set(CALLERID(num)=xxxxxxxxx) <- am I doing something wrong?
19:16.08*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:16.29adam0563D-Fender, I replaced the number with the small x's
19:16.44[TK]D-Fenderadam0563: Not based on that line alone.  enable PRI debug and see what's getting passed <-
19:16.53p3nguinIs this for an incoming call or outgoing?
19:17.00[TK]D-Fenderadam0563: And show us GOOD debug not masking ANYTHING
19:17.24p3nguinadam0563: [tk]d-fender won't call you.  He promises.
19:17.33adam0563ha ha
19:17.45adam0563exten => _801NXXXXXX,3,Set(CALLERID(num)=18019911000)
19:17.48adam0563that is what I have
19:17.53[TK]D-Fenderadam0563: Not based on that line alone.  enable PRI debug and see what's getting passed <-
19:18.31p3nguinIf that's for an incoming call, you're going to be overwriting whatever the real caller ID is with that number.  Was that what you wanted to do?
19:19.43p3nguinAnd most callerID data seems to have a 10-digit number, so drop the 1 off the front of your number.
19:20.50*** join/#asterisk CcRnp (n=shishir@208.179.165.18)
19:20.58adam0563it's for outgoing
19:21.22CcRnpwhat is the maximum sample rate asterisk can support for MOH ?
19:21.37ManxPower-workCaller*ID in NANPA (USA, Canada, a couple of other countries) is 10 digits with no quotes, dashes, or leading 1
19:21.46ManxPower-workno spaces either
19:22.07p3nguinOkay, so that line says that any number you are calling in the 801 area code you want to set your CID to 8019911000.  Is that your goal?
19:23.20adam0563yes
19:23.20adam0563I want to do it to every number but that's in different dial plans so I'll be modifying them later
19:23.20p3nguinccrnp: As far as I know, it must be 8000 Hz.
19:23.41ManxPower-workThere is no max, no min.  It's 8000 Hz, MONO or it won't work
19:23.47adam0563however, I think the signal is actually em_w
19:23.55adam0563not pri
19:24.02ManxPower-workadam0563: you can't set outgoing callerid on E&M
19:24.02CcRnpThank you guys !! appriciated
19:24.09adam0563because I don't have pri debugging available in the asterisk cli
19:24.30p3nguinexten => _801NXXXXXX,3,Set(CALLERID(num)=8019911000)
19:24.31p3nguinexten => _801NXXXXXX,4,Dial(IAX2/account@context/${EXTEN})
19:24.36ManxPower-workadam0563: did you install libpri before you installed Asterisk?
19:25.24ManxPower-workadam0563: You need to find out EXACTLY what kind of line you have before you do anything else.
19:25.29adam0563I wasn't the one that installed the Asterisk machine which is why I'm having this problem. The person who set it up put like 3 asterisk installs on here so I have a tough time because everything is disorganized
19:25.48ManxPower-workadam0563: what is the signaling set to?
19:27.47[TK]D-Fenderadam0563: .... where is my PRI debug?  Obviously have have libpri, etc installed otherwise you wouldn't be placing calls
19:28.02adam0563okay, correct me if I'm wrong, but my dahdi-channels.conf says  pri and uses the from-pstn context, so it should be going through pri
19:28.16ManxPower-workEXACTLY what says "pri"?
19:28.22adam0563the thing I am confused about is that there is no pri debugging in there, is that a seperate module?
19:28.44p3nguinAssuming dahdi is like SIP, the dialplan determines where calls go out.
19:28.55adam0563http://pastebin.com/m15fef56a
19:28.56ManxPower-workadam0563: yes.  It is part of the PRI support for Asterisk.  Looks like you are not running a PRI.
19:29.01p3nguinThe dahdi conf would just determine what happens to calls that come in.
19:29.25[TK]D-Fenderadam0563: Please ignro adam0563 ENABLE IT
19:29.32[TK]D-Fendergah
19:29.34[TK]D-Fenderaasdhasasjhdaslyuioweynr
19:29.51ManxPower-workadam0563: that is /etc/asterisk/chan_dahdi.conf?
19:29.51[TK]D-Fenderadam0563: Ok, "pri debug span 1"
19:29.52*** part/#asterisk E-bola (i=bola@ip181.rev112.brygge.net)
19:29.52adam0563yes
19:29.55[TK]D-Fenderadam0563: go to CLI.  Place a call.  Pastebin it.
19:30.53adam0563http://pastebin.com/m3a0d287e
19:31.11titterif you enabled dahdi, and edited the show_dahdi.conf you will need to stop the asterisk service, and restart it
19:32.14ManxPower-work[TK]D-Fender: Best of luck helping adam0563.  You'll need it.
19:32.59[TK]D-Fenderadam0563: enable PRI DEBUG like I dtold you and include proof of its activation in your pastebin <-
19:33.51*** join/#asterisk Hexbomber (n=Louis@S0106004854811dff.wp.shawcable.net)
19:34.23adam0563in the asterisk CLI, I type in help and it comes up with the commands, there is nothing even related to PRI
19:34.33adam0563therefore, when I type in PRI DEBUG it says it can't do anything
19:35.07ManxPower-workthis is just too painful to watch
19:35.09*** part/#asterisk ManxPower-work (n=ewieling@24.42.221.26)
19:35.14p3nguinhahahahaha
19:35.23[TK]D-Fenderadam0563: pastebin your system.conf
19:35.48[TK]D-Fenderadam0563: and show me the attempt to do the command I told you to do
19:35.49adam0563ha ha......
19:35.55adam0563no system.conf
19:35.59[TK]D-FenderWTF
19:36.07[TK]D-Fenderadam0563: DAHDI REQUIRES IT.
19:36.16[TK]D-Fendercalls BULLSHIT
19:36.17adam0563wait a sec
19:36.18adam0563sorry
19:36.26[TK]D-Fenderreaches for his ClueBat (tm)
19:36.34beekducks
19:36.36adam0563I'm working in like 20 different directories here
19:37.14adam0563http://pastebin.com/m2f35bbc7
19:37.37[TK]D-Fenderadam0563: that is not a &#@^$ing PRI
19:37.47[TK]D-Fendere&m=1-6
19:38.08[TK]D-Fenderadam0563: So FORGET the CALLERID() function.
19:38.24adam0563I told you earlier, there are duplicates of these files and its disorganized, yell at the guy who did this stupid system before me, it says pri in a different config file
19:38.33HexbomberI'm going to admit, I'm a total asterisk noob. I've downloaded / installed asterisk + freepbx on my linux server, yet now I'm not sure what to do haha... I've been looking for DID's, but there are none in my area code.
19:38.39[TK]D-Fenderadam0563: there is no explicit mechanism for setting CID.  Check with your vendor to see if they support "wink + DTMF", etc for setting it
19:39.01HexbomberIn theory I'd like to setup a system where somone calls my DID, and is prompted to press a button, and it transfers them to various numbers based on what they enter.
19:39.04[TK]D-Fenderadam0563: And system.conf begs to differ and says whats what instantly.
19:39.09[TK]D-Fenderadam0563: So pay attention.
19:39.38[TK]D-FenderHexbomber: very doable.  Now go get yourself that DID
19:40.32HexbomberD-Fender: Is it possible that a DID does not exist in my area.. I mean, I live in a fairly large city, with a decent sized area code.. but I can't find providers anywhere.
19:41.00*** join/#asterisk doolittlework (n=d@196.211.34.2)
19:41.11superbeefcheck out this madness
19:41.14superbeefhttp://pastebin.ca/1638165
19:41.36[TK]D-FenderheWhere have you looked?
19:41.41[TK]D-FenderHexbomber: Where have you looked?
19:42.19Hexbomberflowroute, as well as a bunch of other providers that I've googled online.
19:42.22doolittleworksomething strange is cooking, had a power faulure system boots no problem, but for some reason the command restart now in the cli is missing, am i going bonkers i am sure there is a command "restart now"?
19:42.23HexbomberI'm in the 204 area code.
19:44.12titterdoolittlework: service asterisk stop
19:44.18titterthen start it again
19:44.31[TK]D-FenderHexbomber: Waht city
19:44.54HexbomberWinnipeg, MB, Canada.
19:45.06doolittleworktitter: did that no luck the command is still missing
19:45.14titterdon't do that in the cli
19:45.38[TK]D-FenderHexbomber: voip.ms
19:45.55adam0563D-Fender, how can I get the number changed that it calls from? It needs to be from all numbers anyways, I'm assuming I do it somewhere in the DAHDI config files, I just need that number changed.
19:46.07[TK]D-Fendernexles.net
19:46.13doolittleworkwhat does the syatem setting netplug do in services?
19:46.27[TK]D-Fender[15:38]<[TK]D-Fender>adam0563: there is no explicit mechanism for setting CID. Check with your vendor to see if they support "wink + DTMF", etc for setting it
19:46.41[TK]D-Fenderadam0563: PAY ATTENTION
19:47.08adam0563I got all that, but there has to be a way to just change the number. Are you telling me that there is no way to do that?
19:47.19[TK]D-FenderHexbomber: les.net
19:47.39adam0563when I called the provider they told me that the number on outgoing calls was through Asterisk.
19:47.54*** join/#asterisk xpot-mobile (n=xpot@173.8.94.1)
19:48.16titteradam0563: if your co is allowing out bound caller id numbers, that would tell me you are doing something wrong.
19:48.30[TK]D-Fenderadam0563: Feel fre to pick up some communications skills for your next call with them.
19:48.43[TK]D-Fendertitter: Generic and non-applicable.
19:49.47doolittleworkjust loaded * 1.6 did they remove the command " restart now" in cli
19:49.57titterworks for me on 1.6.10
19:50.15doolittleworkstrange
19:50.29leifmadsendoolittlework: core restart now
19:50.30doolittleworki sure i used before i had a power failure
19:51.02adam0563okay. Thanks D-Fender. I'm sorry.
19:51.04*** part/#asterisk adam0563 (n=damagele@70.103.115.194)
19:51.04doolittlework@leifmadsen: ta
19:53.00doolittleworki have 2 * boxes linking with iax, was making test calls between them before power failure on one server, iax show peers os Ok on both but for some reason i can not dial of the other server
19:56.30HexbomberOkay, so now I just fubar'd my database setup...
19:56.45HexbomberI changed my sql password for freepbx, and now it won't let me connect.
19:56.59[TK]D-Fender~freepbx
19:57.00infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
19:57.01[TK]D-Fender^^^^^6
19:57.15Hexbomberthx :)
19:57.24asterwikiis on a break and back in the forum
20:00.23doolittleworkis there a way to soft hangup all busy channels at once
20:00.42*** join/#asterisk andres833 (n=andres83@201.244.125.6)
20:00.43Qwelldoolittlework: restart
20:01.16doolittleworklol
20:01.33doolittleworkpulling the power also works
20:03.53doolittleworki was meaning a command like softhangp *
20:05.50asterwikidoolittlework: google this 'soft hangup <channel> '
20:06.17DavidR2008I'm seeing DTMF begin .... and DTMF end ... messages on my console, but I have verbose set to zero, where are these messages coming from? * 1.6.1
20:08.56[TK]D-Fendercheckout time, later all
20:09.38*** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com)
20:10.11coppiceCheckout time? You work in a supermarket? :-\
20:10.30brunnerI need to hire some asterisk developers.  I don't suppose there would be a job site that would be better than average for finding asterisk devs, right?
20:11.56DavidR2008what kind of work?
20:12.29brunnerdialplan development, mostly, but basic knowledge of PHP and MySQL would be a huge plus
20:13.33ryduhbrunner: what kind of dialplan work are you looking to have done for you? and what are you expecting to pay?
20:13.43mchoulol
20:13.51mchounegotiating already :)
20:13.55*** join/#asterisk errotan (n=errotan@81.0.115.122)
20:14.25brunnerthe pay would be negotiated one-on-one... it's just some interactive stuff with chanspy that would allow the user to control things that are going on in a meetme
20:14.33doolittleworkwhat is php &mysql can one eat it?
20:14.48DavidR2008no, but it can eat you ;-)
20:14.57doolittleworklol
20:15.48doolittleworkanyone here dcap cert
20:15.50doolittlework?
20:16.09ryduhbrunner: I'm a PHP/MySQL pro and just getting into asterisk. I haven't used chanspy or meetme yet.
20:16.44doolittleworkfree pbx has a meetme module, to get your feet wet
20:16.47brunnerryduh: I still might be able to hire you for other things, but I need an asterisk pro more urgently
20:18.15*** join/#asterisk lesouvage (n=lesouvag@82.73.69.76)
20:21.07ryduhdoolittlework: I haven't messed around with freepbx. From reading the ~freepbx notice I'm not sure if I want to get into it.
20:22.11ryduhIf I'm comfortable developing my own dialplans, is there any need to use freepbx?
20:22.38*** join/#asterisk knobo (n=user@90.149.4.182)
20:22.48Qwellryduh: nope
20:23.23*** join/#asterisk eppigy (n=Dave@snugglenets.com)
20:23.29eppigy8[]
20:24.33ryduhwhat is that infobot msg from? i love 8[]
20:24.44eppigyI am dave
20:25.51*** join/#asterisk gardo (n=gardo@110.55.240.37)
20:27.19ryduhbrunner: what are you looking to do with php aside from *?
20:27.47*** join/#asterisk Katty (n=asterisk@mail.copi-rite.com)
20:27.54Kattyscowls.
20:28.00Kattypouts at isp.
20:28.07p3nguinscowls too
20:28.17Kattygives p3nguin a hug :>
20:28.36p3nguin:D
20:28.58p3nguinHow'd the chicken turn out?
20:29.25Kattyjuicy! :>
20:29.36theharnom
20:29.40Kattyand perhaps a bit too salty
20:30.28p3nguinAww.  What do you do when you put too much salt in?
20:30.42Kattyyeah the recipe listed too much seasoned salt
20:30.43eppigyhttp://www.youtube.com/watch?v=-M49SdszkH4
20:30.49eppigyspeaking of penguins
20:30.54Kattyi think i'll make my own Seasoned Salt, minus the salt.
20:31.33jblackNot a bad idea. You can get it pretty cheap in the store too
20:31.59p3nguinsalt-free seasoned salt?
20:32.23jblackUsually it's called seasoning.
20:32.24p3nguinI think they call that "seasoning"
20:32.34eppigyceasoning
20:32.36Katty:>>>>>>
20:32.51jblackNo need to be rude, p3nguin. :)
20:32.58Kattyi'm glad mister penguin got away from the killer whales.
20:32.59*** join/#asterisk denon (i=denon@69.165.165.115)
20:32.59*** mode/#asterisk [+o denon] by ChanServ
20:33.01p3nguinshakes fist at jblack
20:33.32Kattytho i am very sure that pod of whales was sad about the whole situation and dinner getting away.
20:33.34jblackflips p3nguin the bird
20:33.48jblackGet it? Penguin? bird?
20:33.54ryduhlol
20:34.30*** join/#asterisk guywith2names (n=chatzill@69.70.201.246)
20:34.43ryduhguywith2names: what's your other name?
20:35.04guywith2nameslol i dont think you want to know
20:35.12guywith2namesits redundant
20:35.52eppigyKatty: yeah the penguin looks pretty thrilled
20:35.56guywith2namescan anyone here help a noobie out?
20:35.59eppigyand ppossibly trying to blend in
20:36.02*** join/#asterisk bluOxigen (n=asad@static-host119-73-66-63.link.net.pk)
20:36.10p3nguinguywith2names: You'll have to actually address a specific topic.
20:36.19ryduhguywith2names: and ask a question
20:36.21jblackguywith2names: Yes. Anybody can.
20:36.30*** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-146.cablep.bezeqint.net)
20:36.58guywith2namesSure. I have access to my asterisk server from my provider through ssh. i need to be able to edit manager.conf file but i don't know how to access it
20:37.21jblackYou have to ask your provider for instructions.
20:37.21*** join/#asterisk jicksta (n=jicksta@c-98-210-108-28.hsd1.ca.comcast.net)
20:37.43p3nguinOr just use a text editor of your choice.
20:37.52ryduhguywith2names: is it in /etc/asterisk/ ?
20:37.56guywith2namesyes but how do i get to it lol
20:38.01Kattyeppigy: he did an excellent job.
20:38.04guywith2namesi beleive so but i don't kno how to get there
20:38.10Kattyeppigy: now he can take snacks home to the kids.
20:38.11p3nguinvim /etc/asterisk/manager.conf
20:38.12jblackp3nguin: Do you know where manager.conf is on that system? Whether or not he's going to require su, if he needs to run a special binary to restart *? etc etc etc
20:38.30jblackguywith2names: Ask the provider for help. Really.
20:38.40jblackp3nguin: sorry for being rude.
20:38.44guywith2namesim sorry i really suck at linux... all i know is ls
20:38.46guywith2nameslol
20:38.53p3nguinshakes fist at jblack again
20:38.58guywith2nameslet me try what penguin sugested
20:39.13ryduhguywith2names: are you on windows? If yes, have you downloaded PuTTY? If not, do you know how to open a terminal
20:39.24p3nguinI made like four assumptions when I typed that command, just so you know.
20:39.29guywith2namesyes i am using putty to ssh into the server
20:39.31ryduhlol @ telling a noobie to use vim
20:39.38jblackReally bad idea guys. He should be asking for help.
20:39.43kaldemarvi is probably the last editor you want to open if you have absolutely no idea how to use it.
20:39.44p3nguinI don't know why that's funny to you.
20:39.53Qwellp3nguin: ed would have been the better choice.
20:40.23jblacksed -i the best of all
20:40.26ryduhp3nguin: just because it has somewhat of a learning curve. I guess it's not too terribly funny
20:40.30guywith2namesi have some experience with linux
20:40.49guywith2namesvim is an editor if im not mistakend
20:40.54guywith2namesanyhow lemme try that
20:40.56p3nguinIt's not that hard to press i to insert some stuff, delete or x to delete it, and ZZ to write and quit.
20:41.04jblackguywith2names: ask your provider for help. Really. You'll thank me later.
20:41.30jblackI'm not shoving you off to be mean. I really think it's appropriate for a variety of reasons.
20:42.04guywith2namesjblack i made sure he made a backup before i mess around with it
20:42.08ryduhp3nguin: wow, I can't believe I forgot about ZZ. gotta start using that again
20:42.23ryduhguywith2names: who is  your prodiver?
20:42.23guywith2namesim just doing a little modifications
20:42.42guywith2nameswell its actually a friend of mine lol
20:42.45kaldemarp3nguin: it's easy, if you know what buttons to press.
20:42.51jblackDo you have control over the account asterisk is running as, either directly, or indirectly as root?
20:43.04guywith2namesyes
20:43.13guywith2namesi think...
20:43.19*** join/#asterisk esaym153 (n=esaym153@cpe-24-174-176-203.satx.res.rr.com)
20:43.30jblackgives p3nguin a look
20:44.11guywith2namesthanks for the advice, i should come here more often :D, can i get some paid private support here too?
20:44.14jblacklike p3nguin  said, the default directory for configuration files for asterisk is /etc/asterisk . Don't forget to tell asterisk to reload hte config file in the ami
20:44.34esaym153howdy, does asterisk have any options to manage echo in a full voip environment (ie, no pstn or psi).  I get my service from an upstream sip/iax termination..
20:44.36jblackDefinitely. There's several people that'll probably hit you up in about 5 minutes.
20:45.12guywith2namesSure i'd love for whoever offers some paid support to PM me
20:45.16jblackesaym153: in "full voip" It's handled by the endpoints. i.e. the phones.
20:45.40esaym153hmm, then I guess my phone sucks
20:45.49eppigyKatty: yesh
20:45.52jblackThat can definitely be. Is your phone made by "grandstrea" ?
20:46.15jblackbtw, try playing with the volumes. Often you can mitigate or eliminate echo that way.
20:46.54jblackI meant "grandstream", btw
20:47.18esaym153jblack: you talking to me? yes it is a grandsteam ata
20:47.39jblackesaym153: grandstream has a reputation of making the worst phones. Ever.
20:47.52esaym153jblack: what do you recommend for an ata then?
20:48.08p3nguin~grandsteam
20:48.22jblackDid you consider getting away from ATAs, and getting an IP phone?
20:48.27p3nguin~gs
20:48.28infobotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
20:48.49jblackpolycom makes phones for about $120 that are SIP devices. Actual physical phones that do tcp/ip
20:49.05esaym153jblack: well I could get a phone, but I like my cordless handset
20:49.13*** join/#asterisk RobH (n=RobH@216.38.133.254)
20:49.32p3nguinGet a cordless/wifi SIP phone.
20:49.36ryduhYou can get a cordless handset for VoIP phones
20:49.43jblackOk. You can probably get a cheap ata from cisco (aka linksys). They can be a little funky at times, but generally perform the core features well as long as you don't screw with 'em.
20:50.14jblackryduh: Hrmm. I wonder if he meant "my cordless handset" or 'a cordless handset'
20:50.40ryduhjblack: good point. There are probably adapters?
20:50.43jblackesaym153: For $250, I picked up a linksys SPA-8000 that does 8 lines.
20:50.47p3nguinCisco ATA 186 or Linksys SPA-3012 might be reasonable devices.
20:51.20p3nguinSorry, SPA-3102.
20:51.24esaym153my gs cost $25 :)
20:51.31p3nguinAnd look what you got!
20:51.31jblackI hear a lot of line nose on the spa-8k, but I also suspect that I have a poor ground.
20:51.38jblackesaym153: And how's it working for you?
20:51.48ryduhI wish there was a way to wirelessly use asterisk and ditch my cell phone.
20:51.57*** join/#asterisk nightrid3r (n=kvirc@78-20-232-172.access.telenet.be)
20:52.10p3nguinryduh: You could rely on wireless networking everywhere you go.
20:52.17esaym153jblack: well it works, some people get echo though
20:52.22jblackif you can tolerate the echo on that GS you have, then all the more power to you. I was under the impression, though, that you had a problem you were trying to solve.
20:52.29ryduhp3nguin: what about areas without wifi?
20:52.36p3nguinYou'll be withouta phone.
20:52.42jblackwhich, when you get down to it, is "your shit hardware is aqcting like shit"
20:52.54esaym153yea
20:54.24jblackgah. I'm acting like such an asshole today. what's my problem
20:54.40p3nguinMy guess is that yer an asshole.
20:54.40esaym153jblack: you need Jesus ;)
20:54.58jblackp3nguin: Good point.
20:55.08p3nguinAt least that's often why I act that way.  :)
20:55.29p3nguinjblack: Something got you uptight today?
20:55.46p3nguindifferent from the day-to-day crap, I mean.
20:55.56jblackwell, my sleep is off because I'm at war with my doctors.. maybe that's it
20:56.17p3nguinAnything preventing you from having a nap for a while?
20:56.38jblackyeah. I ate 2 dozen bbq hotwings, and need to work out before I spontaneously combust.
20:57.05esaym153I disabled the echo cancellation on the ata I think that helped fix it.  The cordless phone should have its own echo canceller I would think
20:57.10p3nguinWorking out could also help relieve some tension in addition to saving your life.
20:57.58jblackyeah.
20:58.06jblackI hate this diabetes stuff. it sucks bad.
20:58.47p3nguinThis may be too personal of a question, but is your body still able to produce those feel-good chemicals when you work out?
20:59.01jblackIt's starting to get there.
20:59.20p3nguinStarting to not produce them or starting to get too personal?
20:59.36jblackI'm starting to get mild endorphins on occasion.
20:59.51p3nguinThat's should be a good thing.
21:00.09theharrussellb: !
21:00.12jblackyeah. finally a palpable payoff to the treadmill. :)
21:01.01*** join/#asterisk thansen (n=thansen@76.27.110.194)
21:02.10*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
21:02.25guywith2namesanyone know if its possible to inject a .wav file into a call?
21:02.34jblacksigh.
21:02.38guywith2namesplay a recording back to someone?
21:02.49ryduhPlayback(filename)
21:02.50jblackLook at the Background() app, guywith2names . Also, consider getting the asterisk book if you want to do this yourself.
21:03.38jblackand if no one took you up on your money offer yet, hit me up after I finish my gerbil impression
21:03.41guywith2namesheh i don't plan on doing anything, still waiting for someone to pm me so i can talk and maybe get some customizations done
21:07.39*** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net)
21:09.11Kattyyou can also use a call file to call someone and play an audio file
21:10.52*** join/#asterisk WinZ (n=winz@82.146.61.218)
21:20.50*** join/#asterisk mike-ekim (n=digiport@204.13.2.30)
21:20.58mike-ekimI am getting Unable to open /dev/dahdi/ctl: No such file or directory  when i try to run dahdi_tool
21:21.01mike-ekimany ideas why?
21:21.50*** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net)
21:23.26drmessano^dahdi not installed or running?
21:23.35mike-ekimi just did a reinstall of dahdi
21:24.09mike-ekimany specific way to check if dahdi is running?
21:24.28Kattydecides to go home.
21:25.31beekKatty -- how did the chicken turn out?
21:26.38*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
21:26.38*** mode/#asterisk [+o leifmadsen] by ChanServ
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21:33.33*** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net)
21:34.35Chainsawmike-ekim: lsmod will show you.
21:35.46*** join/#asterisk Number5 (n=El@j97231.upc-j.chello.nl)
21:36.10ryduhlol @ evil_gordita's nick
21:36.30evil_gorditaI get that a lot.
21:36.38Number5hello guys, does anyone know a channel about programming own voip/sip software?
21:38.03ryduhHas anyone originated calls by transferring call files via scp ?
21:38.34*** join/#asterisk riddlebox (n=james@75.132.225.75)
21:46.20jblackoh that felt good.
21:47.09*** join/#asterisk tzafrir__laptop (n=tzafrir@bzq-218-155-147.cablep.bezeqint.net)
21:47.46ryduhjblack: did you exercise?
21:48.09jblackI did. 26 minutes and 300 calories. Started feeling so good, I had to stop. :(
21:49.07jblackI can get right back to it though, as it turns out.
21:51.40ryduhI don't know much about diabetes and forgive me if it's too personal, how does diabetes affect exercise?
21:51.42*** join/#asterisk djdefi (n=rtrauntv@64.136.179.53)
21:52.36jblackYour muscles use a type of sugar called glucose, which is in the blood. So, when you exercise, your blood sugar changes around a lot. Diabetics have problems controlling blood sugar.
21:54.04jblackIn the case of type 2 diabetics, cells have trouble absorbing sugar, so the body compensates by making blood sugar higher.
21:54.22jblackso diabetics take something called insulin, which helps with absorption.
21:54.55jblackboth insulin and exercise cause blood sugar to go down. See how they could interract?
21:55.17ryduhgot it
21:56.58coppicediabetes is booming, with fantastic new opportunities for insulin and blood sugar meter sales in emerging markets
21:57.53ryduhehh OSX is shitting it self
21:57.53jblackYay for the widespread adoption of american food. ;)
21:58.00ryduhmust restart
21:58.03*** part/#asterisk ryduh (n=ryduh@204.16.143.186)
21:58.40*** join/#asterisk RobH (n=RobH@216.38.133.254)
21:59.46coppiceUS haute cuisine, brought to you by McDonalds
22:00.44jblacksigh. Be back in another 30.
22:00.51drmessano^That's not chicken
22:01.11jblackIt's the bbq sauce that's giving me the problem.
22:02.28*** join/#asterisk RobH (n=RobH@216.38.133.254)
22:05.33*** join/#asterisk clive- (i=ident@dsl-242-144-214.telkomadsl.co.za)
22:07.00clive-hi, can someone please point me in the right direction on how to do a dialplan so that I can replace $EXTEN with my own value
22:07.15clive-the point is so that I can do pattern matching
22:07.19*** join/#asterisk atis_home (n=atis@193.238.213.215)
22:07.46jblackclive-: reread the section on extensions, particularly _ for pattern matching
22:07.57*** join/#asterisk ryduh (n=ryduh@204.16.143.186)
22:08.05clive-I tried set(EXTEN=${diallednumber})  but that didnt really do the trick
22:08.24clive-jblack,,, in the wiki?
22:09.52leifmadsenyou can't set EXTEN
22:09.56leifmadsenit is automatically generated
22:10.21ryduhWhen I first visited voip-info I thought it was a fake wiki generated for advertising
22:10.30leifmadsenUse something like Set(xtn=${EXTEN}modified)
22:11.38clive-leif and then something like xtn => _1X.,1,noop etc
22:11.41clive-?
22:11.50leifmadsenno....
22:11.54leifmadsenwhat are you trying to do?
22:12.36clive-i have an agi scrip that is asking for the required number, now I want to dial that from the dialplan, with matching ability
22:13.00leifmadsenok... so you just pattern match and use ${EXTEN}
22:14.15*** join/#asterisk dkirker (n=dkirker@129.65.204.135)
22:14.34clive-problem is that the ${EXTEN} value is the access number into the system, not the number generated from the IVR (agi)
22:16.12*** join/#asterisk coder2000 (n=coder200@beigetower/coder2000)
22:16.58*** join/#asterisk JKac3BEq (n=JKac3BEq@173-20-71-143.client.mchsi.com)
22:17.21clive-I guess I could easily just calculate the complete dial string in the agi script... that may be easier ... hmm
22:17.52coder2000I am trying to install switchvox in virtualbox 3.0.8 and found out I needed to disable apci but because I have to pass these options the root partition option is lost and I can't figure it out. Can anyone help?
22:22.17Katty:>
22:27.26*** join/#asterisk voipmonk (n=voipmonk@67.204.45.155)
22:30.53clive-ok, I just did it all in the agi script... thanks for your help.
22:32.03*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
22:32.26Kattymmm, dinner was good.
22:32.33Kattyrice, chicken, celery, onion, cheese.
22:32.57Kattyand some edamame beans (=
22:33.08coppicethose are components, not a product
22:33.19Kattywell chicken rice and veggie casserole
22:33.24Kattyand edamame beans on the side.
22:33.28Qwellcoppice: You clearly are not a bachelor.
22:33.41Qwellcheese is a product.  as are rice and chicken.
22:33.51clive-gnight guys..coppice !
22:34.09Kattyi often make the mistake of listing ingredients asuming people can visualize the end result in their head.
22:34.21Kattynot everyone cooks as much as i do--sorry )=
22:34.26[TK]D-FenderKatty: But... I just went to the bathroom!
22:34.43Kattythat's...nice?
22:34.48Kattydid you wash your hands?
22:34.59[TK]D-Fenderthinks of other "end results"
22:35.08Katty<PROTECTED>
22:35.21[TK]D-FenderKatty: pwned
22:35.25Kattyyou so did.
22:38.08Kattyshower time (=
22:38.14Kattyleafs.
22:40.08coppiceQwell: are you the sort who eats chicken sashimi simply to save the time and effort of cooking it?
22:40.33QwellI don't know what sashimi is, so...yes.
22:40.51[TK]D-Fendercoppice: sashimi?  Nah.. that takes too long to prepare
22:41.03[TK]D-Fenderchases down a live chicken and eats it live....
22:41.31[TK]D-Fenderpicks the excess feathers out with a toochpick...
22:42.52coppiceQwell: you need to get past McDonalds more often
22:43.39[TK]D-FenderQwell: You're still young enough to save yourself from ending up "tragically white"!
22:43.53Kobazsashimi is the raw sushi
22:44.09[TK]D-FenderKobaz: .... not quite....
22:44.19mchouno joke, domino's has a large 1 topping for $6 special going on now
22:44.48mchouhad one last night
22:44.57Kobazs a Japanese delicacy primarily consisting of very fresh raw seafood,
22:45.04Kobazraw fish
22:45.11mchoutasted like the cardboard box it came in
22:45.15[TK]D-Fendermchou: thats the price of a full meal at my favourite indian restaurant, only incredibly less healthy & appealing :)
22:45.40coppicemchou: that's not a cardboard box. its the 2 for 1 deal
22:45.54Kobaz[TK]D-Fender: my cousins wedding had an all you can eat sushi bar
22:45.59mchou[TK]D-Fender: really?  you get a full Indian meal for $6?
22:46.01Kobazalong with all kinds of other foods
22:46.34mchou[TK]D-Fender: around here that's $10
22:47.11[TK]D-Fendermchou: Yup....
22:47.30Kobazgood prepared foods are expensive these days... i paid like 15 bucks at this pakistani place... the food was good, but it was like half a serving... maybe a third of one
22:47.54Kobazand in order to get filled up on sushi, you need to order like 5 dishes
22:47.55drmessano^I loved going to the pakistani restaurant.. Not so much being fired upon during the meal
22:47.56Weedlepakastani is a type of food ?
22:47.59coppiceI've had excellent indian meals for <$6. mostly in india
22:48.22mchoudrmessano^: see any Talibans over at the other table? :)
22:48.23Weedleyeah we get massive indian here for 8 bucks, so much that ylu can barely finish
22:48.24Kobazno wait.. it was an afghani resturant
22:48.26[TK]D-FenderKobaz: for $10 I'd be stuffed silly there... the food is awesome... I'd gladly be there paying more if the prices had to go up...
22:48.46[TK]D-FenderKobaz: as it is its cheaper than most McDonalds trios
22:48.50Kobazwell yeah
22:49.10Kobazmcdonalds gives you an 8000 calorie burger, with fries and a drink for like 7 bucks
22:49.17drmessano^We get two meals out of a $15 portion of Indian food.. that includes a full order of Naan, and a Samosa per person
22:49.27mchouKobaz: that's lame
22:49.34drmessano^My wife and I eat for two days for $30
22:49.39mchouKobaz: incredibly BAD deal
22:49.54Kobazmchou: if you need the calories from fat.. it's not bad
22:49.59mchoulol
22:50.18Kobazalthough getting some $1.50 cans of corn beef would get you the same intake
22:50.26drmessano^wants some Saag Paneer now
22:50.32drmessano^bastards
22:50.35*** join/#asterisk knctrnl (n=aembrey@nat/digium/x-awntosopzfodanov)
22:51.10Kobazfor 7 bucks, at the corner store... i get a foot long sub, a drink, and a bowl of soup
22:51.31mchoustill a bad deal
22:51.51Kobazwell, it's obviously cheaper to make it yourself
22:52.02Kobazbut for across the street, it's not bad
22:52.07drmessano^For $1.12 I get a bowl of soup in a paper bag and a badass hunk of 10 day old bread
22:52.11drmessano^Beat that
22:52.14Kobazhaha
22:52.23[TK]D-Fenderdrmessano^: You know... I can't say that I've every had saag paneer....
22:52.29[TK]D-Fenderever*
22:52.32drmessano^[TK]D-Fender: O.O
22:52.37drmessano^[TK]D-Fender: WHAT?
22:52.42[TK]D-Fenderdrmessano^: I've had jsut about everything else...
22:52.47Kobazwell for 7 bucks i can get a sack of bagels and a pound of deli meat and cheese... and make a week's worth of lunch
22:52.50mchou[TK]D-Fender: what?? That's a fake Indian place you're going to
22:52.55[TK]D-Fenderdrmessano^: drmessano^ what is "saag"?
22:53.07Kobazmchou: is that a good deal?!
22:53.09[TK]D-Fendermchou: No, THEY have it... I've simply never gotten to ordering it.
22:53.11drmessano^Spinach
22:53.37[TK]D-Fenderdrmessano^: that sounds like palak paneer... that is the soupy spinach & cheese
22:53.38mchouKobaz: no.  bagels are not such a great deal.  Nor are cold cuts
22:53.39drmessano^Depending where you get it, it MAY or MAY not be creamed with Coconut milk.. Where we get it, it is
22:53.49Kobazmchou: what the hell do you eat then?
22:53.50drmessano^Palak Paneer is the same
22:53.52[TK]D-Fenderdrmessano^: Perhaps a slightly diggerent blend
22:54.06mchouKobaz: thai food
22:54.09Kobazhehe
22:54.19mchouKobaz: India food
22:54.23mchouIndian*
22:54.26[TK]D-Fenderdrmessano^: I've heard someone order saag there, but I've ordered palak myself... wonder if its a dialect thing
22:54.26nix8n82is this like rachel ray hour?
22:54.32[TK]D-Fenderdifferent*
22:54.33Kobazand how much is it... versus how much you get filled up?
22:54.34mchouKobaz: Mexican Food
22:54.37drmessano^I DONT think there is a difference between palak paneer and saag paneer, though I have wondered if thats the distinction in the creaming
22:55.03Kobaznix8n82: no soup for you
22:55.05mchouKobaz: I get a BBQ Pork Burrito for $4, good for 2 meals
22:55.16drmessano^All the store brand foods are "Palak Paneer" and they all taste like shit unless I heat them with coconut milk
22:55.32mchouKobaz: and definitely healthier than McDs and the like
22:55.38[TK]D-Fendernix8n82: http://www.youtube.com/watch?v=dpVdkHZsOW8
22:55.48Qwellhttp://en.wikipedia.org/wiki/Saag
22:55.50Kobazmchou: heh... a bbq pork burito usually is one meal for me
22:55.50Qwelllrn2wikipedia
22:56.17Kobazmchou: and the place i get them from... they are huge... like 1.5 pounds probably alltogether
22:56.19drmessano^I discovered that little trick after getting 10 boxes of "Palak Paneer" for $1 a box (marked down from $4) in the discount bin at the grocery store in redneckville
22:56.45Kobazi can eat like... two footlong subs
22:56.51Kobazand be hungry an hour later
22:56.55drmessano^I see
22:56.58drmessano^So both are correct
22:56.59nix8n82lol..thanks
22:57.06mchouKobaz: that's why the corner store aint a good deal
22:57.07drmessano^I just never bothered to wiki it
22:57.11drmessano^heh
22:57.11Kobazmchou: yeah
22:57.21Kobazmchou: or i just need to lower my metabolism
22:57.21*** join/#asterisk neurosys (n=vinix@adsl-072-151-208-134.sip.mia.bellsouth.net)
22:57.35Kobazmchou: with the soup though... i get filled up
22:57.35mchouKobaz: you still a teenager?
22:57.39Kobaz26
22:57.48mchouKobaz: close enough
22:58.12Kobazhaha
22:58.12mchouKobaz: whait till you hit 30
22:58.12mchouwait*
22:58.13Kobazi dunno
22:58.14[TK]D-FenderQwell: Per that link palak paner has always been fiercely green, not yellow.
22:58.23Kobazwith the amount of excersize i get... i'm usually losing weight
22:58.37mchouKobaz: what you do for exercise?
22:58.43*** join/#asterisk ecrane (n=ecrane@o1-69-19-166-10.static.o1.com)
22:58.43[TK]D-FenderQwell: So does look a fair bit different
22:58.47Kobazrun, mountain bike, climb, ski
22:59.03mchouKobaz: wher you live? CO?
22:59.07Kobazcentral pa
22:59.11Kobazi did live in co for 6 months though
22:59.21Kobazi wanna move back at some point
22:59.30[TK]D-Fendernix8n82: You're welcome ;)
22:59.41ecraneSorry, any rules on asterisk questions? I am having trouble finding out which version of asterisk is the official 'Stable' version.
22:59.42*** join/#asterisk wonderworld (n=w@62.143.22.226)
22:59.55Kobazasterisk... what's asterisk?
23:00.03neurosysshift + 8
23:00.12[TK]D-Fender.. what is this "stable" of which you speak?
23:00.13*** join/#asterisk tzafrir__laptop (n=tzafrir@bzq-218-155-170.cablep.bezeqint.net)
23:00.15ecraneyeah; it's the shift + 8
23:00.25Kobazthere is no *
23:00.26[TK]D-Fenderfetches some more spackle to hold his together
23:00.30mchoustable of horses
23:00.49neurosysstable of hookers?
23:00.58[TK]D-Fenderkills mchou's equine and begins flogging.....
23:00.58Kobazisn't it a gaggle of hookers
23:01.11neurosysgeese. Gaggle of geese
23:01.33[TK]D-FenderKobaz: Only if they're into auto-erotic asphyxiation
23:01.34drmessano^ecrane: 1.4, 1.6.0, 1.6.1, are all "Stable" branches
23:01.37ecranefistful of hookers?
23:01.50mchounaw, fistful of $
23:01.56mchouwrong key
23:02.03drmessano^Trunkful of dead hookers?
23:02.08ecranethanks drmessano^.
23:02.11Kobazsounds about right
23:02.18neurosysbingo
23:02.22ecraneThis seems like a pretty cool channel.
23:02.25[TK]D-FenderNothing quite as stable as a "dead stop"
23:02.33drmessano^It's all fun and games til you get stuck figuring out where to bury the hooker
23:02.53drmessano^Easy way to ruin a hangover
23:03.44neurosysMust always have lime in the trunk
23:03.48jblackHaving sex. For a job.
23:04.02Kobazthat's the idea
23:04.22jblacksounds better than "fry cook at mcdonald's" to me
23:04.38mchouyo wanna some fries with that?
23:04.46ecranesmells worse though.
23:04.47jblackNo, but I'll take some shake. =)
23:05.07jblackrimshots
23:05.17neurosysBut what if its a woman who's 300+ and smells like rotten egg? Still sound like the ideal job
23:05.40Kobazrimjob?
23:05.47jblackNot everybody is well suited for any particular job.
23:05.49drmessano^I always find throwing a few shredded kittens in the trunk will kill the dead hooker smell
23:05.51mchouneurosys: is that years or #?
23:06.13neurosyslbs. but you get the idea ;)
23:06.17jblackdrmessano^: So... your trunk smells like dead pussy, right?
23:06.25drmessano^Rimshot?
23:06.31mchoudead pussy detail!
23:07.02jblackrimshot.. that's the drum sound that you hear after a joke.
23:07.22drmessano^jblack: I was cueing it
23:07.33drmessano^BA DUMP CHING!
23:07.34*** join/#asterisk RobH (n=RobH@216.38.133.254)
23:07.39drmessano^But you failed me
23:07.41drmessano^:(
23:07.55jblackDid not
23:08.08drmessano^The thing about hooking is, you wont ever have to worry about women over 50 propositioning you.. Thats like expecting a kid to beg for brussell sprouts
23:08.12jblackWhy don't you go tweet that there's chickin in your chicken nuggets, or something
23:08.31drmessano^jblack: That was the 17th of August
23:08.57jblackwhat was
23:09.02drmessano^jblack: You've been tweetfeated
23:09.12drmessano^The chickin the chicking nuggets
23:09.19drmessano^Or was that on Facebook?
23:09.20jblackoh.
23:09.37CurtmanIs it possible to have my fxo port monitor a line for rings, and ring my fxs extensions, but not pick up the line until/if an extension picks up?
23:09.37jblacksame difference.
23:09.37drmessano^Maybe I blogged it
23:09.37drmessano^or Laconica
23:09.54drmessano^or Windows Live Tweet Blog Thingo
23:09.58jblackcurtman: Sure. Just do a dial without an answer.
23:09.59KobazCurtman: that's kinda how it works by default
23:10.06KobazCurtman: unless you Answer()
23:10.23neurosysDoes Freenode/#asterisk have a facebook fanpage? seriously?
23:10.34drmessano^Microsoft will eventually add a twitter like service to Windows Live... they will probably call it "Windows Live Net Send"
23:10.42drmessano^or "Windows Live Notepad"
23:10.47Kobazwe should make an asterisk facebook game
23:11.17neurosyssetup the most convoluted dialplan? What kinda game would we make :P
23:11.19drmessano^"Kick the newb in the nuts?"
23:11.31drmessano^Thats kinda boring
23:11.34Kobazcomplete with voip packet loss, crappy telco tech support, acts of nature, and construction crews cutting your fiber
23:11.51jblackkobaz: Oh, they have that. It's caled "myspace"
23:11.53CurtmanKobaz, I've just started to play with the GUI's..  I don't know of another "default".  I'm still doing a lot of reading, just wanting to know if there is light at the end of the tunnel. There's only so much free time to spend.   ;)
23:12.03drmessano^High Availability Scenarios with Grandstream ATAs?
23:12.07neurosysdont forget the homeless men pissing on your punch block in the meter rooms
23:12.07drmessano^Now that would be a cool game
23:12.10Kobazdrmessano^: yes!
23:12.29jblackTLC could do a series called "I shouldn't be calling"
23:12.52drmessano^"Grandstream ATA clustering solutions 2: Full Frontal Dialplan"
23:12.55drmessano^I can see it now
23:12.58KobazCurtman: the gui's will only get you so far
23:12.59drmessano^In stores next month
23:13.16drmessano^Astercraft!
23:13.21CurtmanKobaz, I realize that.  I'm just using them as a starting point to learn with.
23:13.23drmessano^World of Astercraft!
23:13.30*** join/#asterisk jadl_ (n=jadl@89.130.82.210)
23:13.41Kobazgrandstream makes the worst shit imaginable
23:13.53Kobazit's amazing people buy their stuff
23:13.57drmessano^spends all day in the woods sniping newbs, then heads off to slay an Avaya installer
23:14.09neurosysI finally listened to [TK]D-Fender  and started replacing my ciscos with polycom
23:14.16Kobazpolycrum!
23:14.21Kobazyeah polycom rocks
23:14.24drmessano^Asterisk Interactive Fiction!
23:14.26Kobazthey need more buttons though
23:14.41Kobazif polycom had like, even one more extra programmable button
23:14.53Kobazwell the 320's anyway
23:14.54drmessano^"It is dark.  You hear the ringing of a Cisco phone in the distance"
23:14.57drmessano^> Look
23:15.07drmessano^There is a punchblock and a grandstream ATA here
23:15.10drmessano^> Get ATA
23:15.19Kobazattach punchblock
23:15.28drmessano^"Why the hell would you do that?"
23:15.31Kobaz> the puchblock awakens
23:15.32ryduhI originate a call to a PTSN # over SIP. I have my dtmfmode=rfc2833. I'd like to dial extension 102 at the destination. I try SendDTMF(1w) SendDTMF(0w) SendDTMF(2w) but it only sounds like 1 digit is being sent. is that normal? is there a better way to do that?
23:15.34drmessano^> Kill ATA
23:15.43drmessano^"You slay the ATA and it explodes"
23:15.44Kobaz> the pucnblock hits you with 90 volts
23:15.57neurosysuch
23:15.57drmessano^> North
23:16.17drmessano^"You head north and approach a scary AT&T tech in the network closet"
23:16.27drmessano^> Kill AT&T tech
23:16.49neurosysman i miss the telnet MUDs :P
23:16.49drmessano^"The AT&T tech smashed your head in with a buttset.  You are now dead.  Score 0"
23:16.49Kobaz"the AT&T network arrives"
23:16.59drmessano^:(
23:17.32jadl_hello, I have a problem, I have a sip voip provider Yacom and x-lite but do not work, I configured asterisk and everything is fine but I think something went wrong, when I call, the operator says the prefix is not valid
23:17.58*** join/#asterisk coppice (n=chatzill@host86-132-26-86.range86-132.btcentralplus.com)
23:18.02drmessano^"It is pitch black.  You are likely to have your day eaten fighting echo on old, rotted pots lines."
23:18.04Kobaz> Look
23:18.05Kobaz"You are in the main control room, you are likely to be eaten by a grue"
23:18.06drmessano^^^^ thats IT!
23:18.12drmessano^ha
23:18.15jblackSometimes, when my head hurts really bad, I turn off the lights and wait for the grue that should surely eat me.
23:18.16jadl_<PROTECTED>
23:18.16jadl_can someone help
23:18.24[TK]D-Fender|<--Deeewayne has left irc.freenode.net ("Herbivores ate well cause their food didn't never run") <-- Double negative FAIL
23:18.26jblackYou stole my grue line!
23:18.34Kobazhaha
23:18.39drmessano^You stole my grue line
23:18.47drmessano^Though i didnt use grue..
23:19.08drmessano^But mine was a blatant ripoff
23:19.24neurosystelenet (NUAA Attacker).......
23:19.28neurosysc sprint
23:19.31neurosys:P
23:20.51jblacklooks for a t-shirt with the grue line
23:21.07jadl_I have read about Asterisk, I've seen videos and I downloaded some instructions and VozTelecom asterisk and nothing ... please help
23:21.07drmessano^"You wake up.  it is 4:30 in the afternoon.  The smell of rotten pizza permeates the air.  You're not wearing any pants, and there's poorly written dialplan glaring at you through from your 24 inch LED monitor."
23:21.10drmessano^> whoami
23:21.15*** join/#asterisk natlonehat (n=natloneh@202.170.42.67)
23:21.16drmessano^"You are [TK]D-Fender"
23:21.34drmessano^\o/
23:21.44neurosyslol
23:22.03[TK]D-Fenderdrmessano^: http://xkcd.com/386/
23:22.15drmessano^One of my FAVS
23:22.20cuscohi
23:22.28drmessano^ho
23:22.34cuscohow can I logoff a manager user
23:22.51cuscoif possible trough cli
23:22.57cuscoasterisk's cli
23:23.03jblackhey, infocom lets you download zork 1-3 at not cost
23:23.42[TK]D-Fendercusco: iptables -..... err ;)
23:23.42Jumpieis it possible to do kinda an inverse blacklist?
23:23.42Jumpiei wanna rject all but a few i wanna specifically allow
23:23.42drmessano^has the infocom pack with 20 games in it
23:23.42cuscolol, just logim off once
23:23.42cuscoisn't it possible?
23:23.42drmessano^HHGG ftw
23:23.43[TK]D-FenderJumpie: You can do whatever you want.  Its your dialplan.
23:23.43drmessano^"get not tea"
23:23.43neurosyscusco:core stop now
23:23.44[TK]D-FenderJumpie: Filter based on WEATHER if you feel like it
23:23.45drmessano^"You pick up the not tea"
23:23.47cusconeurosys: hmm maybe later
23:23.48Jumpiei guess i meant simplistically heh
23:23.54Jumpieah
23:24.19drmessano^Whitelist
23:24.21[TK]D-FenderOk, martial arts then stage time... later all
23:24.23drmessano^then Blacklist
23:24.28neurosysl8r
23:24.38drmessano^Whitelist has a GOTO to jump past the blacklist
23:24.38ryduhanyone for some DTMF help?
23:24.58*** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1)
23:25.00Jumpieah..right
23:25.04jblack[TK]D-Fender: already leave yet?
23:25.08Jumpielike network access lists somewhat
23:25.11Jumpieorder is important
23:25.30jadl_what can I do?
23:26.00drmessano^jadl_: Anything you want
23:26.22jadl_I don't speak english
23:26.48jadl_I use google
23:27.13jadl_sometimes
23:27.39jadl_hello, I have a problem, I have a sip voip provider Yacom and x-lite but do not work, I configured asterisk and everything is fine but I think something went wrong, when I call, the operator says the prefix is not valid
23:27.50jadl_(repeat)
23:28.14jblackI have drmessano^'s picture. Who wants to see it?
23:28.30jblackhttp://i258.photobucket.com/albums/hh260/lawngnome8273/untitled40.jpg
23:28.54drmessanoIS that a BATMAN lunchbox?
23:28.59drmessanoOh SNAP
23:30.03jblackWhen 'catty' (mispelled to avoided console beeping)  comes back later, I'm going to offer her "my" picture, and see what she says. :)
23:30.54jadl_drmessano ^: you can help, or language is a problem ...
23:31.52jblackjadl_: set debug 9, and verbose 9, and take thedebugging from there.
23:31.54jblackhttp://media.photobucket.com/image/fat%20people/staunchusa/2ontruck.jpg
23:32.18ryduhCan I run multiple SendDTMF commands in a row?
23:33.45*** join/#asterisk manxpower (n=ewieling@24.42.221.26)
23:36.26*** join/#asterisk stew (i=1413@freenode/staff/stew)
23:36.32*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
23:36.39jadl_I know that there is a verbose and debug mode and I can not remember how to do what you say?
23:37.38Chainsawjadl_: core set debug 9
23:37.46Chainsawjadl_: core set verbose 9
23:38.23jadl_thanks you
23:38.53jadl_
23:38.54jadl_Well, now what?
23:39.42*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
23:40.23*** join/#asterisk RobH (n=RobH@216.38.133.254)
23:41.57ecraneGuys my compile process for asterisk is acting really weird. I did a './configure', did a 'make menuselect', 'make', 'make install' and now the program runs fine without any problems. I didn't have to chase down any dependencies or build errors... this doesn't feel right at all :<.
23:42.29ChainsawDon't worry, it'll feel more real when you get to DAHDI.
23:42.30*** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68)
23:43.21*** join/#asterisk bn-7bc (n=bjarne-i@mac.lan.noare-1.holmedal.net)
23:44.43*** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202)
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23:47.45*** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk)
23:49.13jadl_bye
23:50.27jadl_
23:50.27jadl_back tomorrow
23:50.52jadl_xd
23:54.32*** join/#asterisk voipmonk (n=voipmonk@CPE002191f85581-CM00186830a2e0.cpe.net.cable.rogers.com)
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