IRC log for #asterisk on 20130819

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10:45.07nam3l3zzhi all, have a following, could anyone comment on how bad is it ? http://pastebin.com/Vvzn8spP
10:54.04jkroonnon-issue probably.
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11:21.51nam3l3zzjkroon: thnx
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13:26.21askinhi, i am using queue for a project, i have to dial some numbers after answer(to skip ivr menu). How can i do?
13:27.02jmetrocore show application dial
13:27.11leifmadsenthat ^^
13:27.18leifmadsenthere are flags for sending DTMF
13:27.47askini use elastix, i didnt wrote any code.
13:28.08leifmadsensorry to hear that
13:28.21leifmadsenthis channel can't help you with GUI based projects
13:28.29leifmadsenyou'll need to ask the #elastix people
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13:30.16jmetrothats a shame, it would have been easy
13:31.17askinok, i can write dial plan :) but i dont know how to start? is there any example?
13:31.42mjordan~thebook
13:31.42infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
13:31.52mjordanaskin: ^^^
13:32.01leifmadsenget it while you can, because there won't be a 5th edition :)
13:32.03jmetro^ Thats a good resource, but if you are using a GUI, they most likely mangle the dialplan in such a manner that you will be unable to modify it yourself.
13:32.08mjordanleifmadsen: ???
13:32.16leifmadsenmjordan: too much work :(
13:32.43mjordaninteresante
13:32.43leifmadsenplus, everyone is one 1.4 and 1.8 still... so that book will be valid for 5-6 years :)
13:32.44askinok, thanks
13:33.00mjordanwe should talk about that at some point :-D
13:33.39leifmadsenmjordan: talk about which point? :)
13:33.43leifmadsenmjordan: I'll be at AstriCon!@
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13:50.53jmetroastricon seems fun
13:51.02Qwelljmetro: It is.  You should go.
13:51.08leifmadsenit's super fun
13:51.14leifmadsenI'm excited.
13:51.24leifmadsenQwell: you going this year?
13:51.30Qwellleifmadsen: heck yeah I am
13:51.31leifmadsenQwell: you can be a booth babe!
13:51.36leifmadsen:D
13:51.37jmetroThe engineers here are a cloistered group, i'd never be brought with
13:51.37QwellOH SNAP
13:51.40leifmadsenperfect timing
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13:52.40Qwelljmetro: Make them take you.  Get some dirt on them.  Or their wives.  Or girlfriends.  Or both!
13:53.16jmetroCould work..
13:53.22Qwelloh, it works
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13:56.07raidghostWhen asterisk just start stop doing its job, and not much info is to find in logs. It sounds like crappy hardware takingovah
13:57.17raidghostI reboot the computer. Asterisk works for some hours, and then suddenly without a notice it stop working propperly. I gonna try to set asterisk to debug mode to try figure out whats going on. Annoying when things happends i dont know why.
14:00.45Qwell"it stops working" doesn't really say anythign
14:01.19jmetroset your debug and verbose up and i bet youl lfind something
14:01.22raidghostWell. i dont get tone when i pick up the phone, And inbound and outbound calls does not give other than a "busy" signal
14:01.38QwellWhat version of Asterisk?
14:01.43raidghostLatest one.
14:01.45raidghost11.5.0
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14:07.09Chainsawraidghost: And it's not just the bad bad internet routing calls to Palestine?
14:07.28raidghostChainsaw: I dont do calls outside norway. So yes.
14:11.02raidghostBut, i will reinstall asterisk on a fresh install of debian. And install the version of asterisk that debian have in apt.
14:12.41jmetrojust get the new version, dont use the apt-get install
14:13.10raidghostjmetro: everything did work perfect out of the box. Until i upgraded from 11.3.something to 11.5.0
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14:13.24raidghostafter that every little issue did show up over long time
14:14.28[TK]D-Fenderraidghost: What are youcalling over?
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14:15.23raidghosttry to explain your question and i might understand better.
14:16.47raidghostIAX2 if that was what you wanted of info. Im standing here like a ? trying to understand what your looking for of info from me.
14:17.20[TK]D-Fenderraidghost: You should be showing us debug for the calls then
14:22.23raidghosti thought sip set debug on should ben screaming out info
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14:31.02saxahi, can somebody help me understand why I get this
14:31.04saxahttp://pastebin.com/DYLrMgca
14:31.40saxait seems a nat problem , but it was working. Afaik i have never changed anything
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14:36.01igcewielingYes, it is a problem.    Retransmitting #1 (NAT) to 187.115.169.227:5060:
14:36.10igcewielingthat means you have a networking or NAT issue.
14:37.02saxaok
14:37.34saxathats why it gets disconnected probably, but the thing is how to understand where the issue is ?
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14:38.05saxaasterisk is behind a NAT and the casasip phone behind another nat at my home
14:43.44igcewielingset nat=yes externip= localnet=
14:43.51igcewielingset directmedia=off
14:44.12igcewielingmake sure the localnet contains a network, not an ip
14:44.21igcewielingyou know, all the standard NAT stuff.
14:45.02igcewielingalso turn off SIP ALG and/or SIP SPI on your home router.
14:45.04saxaigcewieling: yep, thats all set up
14:45.28saxai have to check my home router
14:45.31igcewielingsaxa: put the [general] section of sip.conf on a pasebin
14:46.13[TK]D-Fender[10:22]raidghosti thought sip set debug on should ben screaming out info <- .... you just said your were using IAX2.  What good do you think ***SIP DEBUG*** is going to do about that?
14:47.51[TK]D-Fendersaxa: it is not setup right
14:48.05[TK]D-Fendersaxa: * is giving your phone a LOCAL IP to connect to.
14:48.36[TK]D-Fendersaxa: c=IN IP4 192.168.0.1
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14:50.28saxahttp://pastebin.com/8aCP25q5
14:52.11saxa[TK]D-Fender: but 192.168.0.1 is the * internal ip
14:52.19[TK]D-Fendersaxa: braserv.chickenkiller.com <- ping it from CLI
14:52.57Kattygood morning
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14:53.19jmetrobraserv.chickenkiller.com doesnt seem to work.
14:53.24jmetroelo kati
14:53.36wasanzyafter installing asterisk, where can I copy the sample configs like sip.conf etc from?
14:54.06Kattywaves to jmetro
14:54.36feeshonCan anyone suggest a free VOIP call recording software?
14:54.37[TK]D-Fenderwasanzy: "make sample"
14:54.39newtonrwasanzy, you can run "make samples" which will install them in /etc/asterisk
14:54.54[TK]D-Fender+s
14:55.30newtonrwasanzy, or just copy the ones you want from asterisksource/configs/
14:55.40wasanzyoh ok thank you
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14:57.04saxajmetro: how you mean that ?
14:57.23feeshonThe phone system isn't running asterisk so I would need something else..I know it isn't the prefect place to ask but any help suggestions would be great!
14:58.16Kattygives feeshon weird looks
14:58.35KattySO.
14:58.42Kattyi finally got around to making borscht
14:58.49Kattynot too shabby.
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15:01.53pabelangerIs the minregexpire and maxregexpire settings for chan_iax2 for the farside?  Eg: control incoming registrations?
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15:16.01saxa[TK]D-Fender: http://pastebin.com/hhzMDn7q
15:16.25[TK]D-Fendersaxa: ....
15:16.34[TK]D-Fendersaxa: it resoles... AS YOUR LOCAL IP
15:16.38[TK]D-Fenderresolves*
15:16.43saxayes
15:16.45saxaoh ok
15:16.54saxalet me check my hosts file
15:17.23saxaok i had an entry in there
15:17.32saxathat was probably the issue, let me check
15:18.00[TK]D-Fenders/probably/definitely
15:18.54saxaroot@braserv:~# ping braserv.chickenkiller.com
15:18.54saxaPING braserv.chickenkiller.com (201.50.103.98) 56(84) bytes of data.
15:18.54saxa64 bytes from 201-50-103-98.user.veloxzone.com.br (201.50.103.98): icmp_req=1 ttl=64 time=1.06 ms
15:19.36saxagreat
15:19.48*** join/#asterisk navaismo (~navaismo@189.241.68.101)
15:19.49saxait works now
15:19.59saxathanks [TK]D-Fender
15:20.32saxai have the entry in my hosts file because on that same machine i have also a web server
15:21.01saxaand on my local net when i write braserv.chickenkiller.com it simply refuses to work if i do not have this
15:21.18saxaanyway it seems i forgot to comment it out once
15:21.30saxasorry for bothering , but thanks to all
15:21.56dpeloquinlove the domain
15:26.16saxathats a free one on freedns.org iirc
15:30.34wasanzyanyone has idea about PIAF2?
15:33.54igcewielingwasanzy: Yes.
15:36.46wasanzyam trying to implement google voice which Wolfram provides, the question is do I need PIAF2 before that can be done? or I could go ahead and use my normal asterisk I installed from source?
15:37.31igcewielingThat wasn't my idea.  My idea is that this is not the right channel for PIAF questions, and even if it was, PIAF sucks, I doubt version 2 is much better.
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15:38.14igcewieling~piaf
15:38.38wasanzyigcewieling: am not asking to use PIAF, because I don't want to use it so I was wondering if there is an alternative way
15:38.57igcewielingwasanzy: Do you understand what PBX In A Flash (PIAF) does?
15:39.01*** join/#asterisk jylebleu (~jylebleu@78.208.220.3)
15:39.08igcewielingor core correctly, what it is
15:39.36wasanzyigcewieling: frankly, I don't understand
15:39.55igcewielingIt is a gui distro like freepbx, which uses freepbx and adds lots of other stuff.
15:40.00jmetroyou can make anything work in vanilla asterisk that works in a packaged asterisk
15:40.14jmetropackaged asterisk = elasticx, freepbx, pbiaf.
15:40.32igcewielingjmetro: don't forget trixbox!
15:40.50jmetrotrixbox is the most fun of them
15:41.31wasanzyjmetro: I have asterisk installed from source (core) and I don't want to use any of those that is why am asking
15:41.48igcewielingwasanzy: no you do not need PIAF.   Your question is answered.
15:42.09igcewielingyou also do not need FreePBX, Trixbox, Elastix, or a camel.
15:42.09wasanzyright
15:42.11wasanzythank you
15:42.29igcewielingthough the camel would be the most useful of the 4
15:42.52wasanzyso working with my sip.conf and extension.conf will be ok right?
15:43.01jmetroand possibly asterisk.conf
15:43.24jmetroi think they have you download modules to add to it
15:43.31jmetrobut no pakages / guis / things that make it hard to use
15:43.47wasanzyjmetro: yea
15:43.51igcewielingwasanzy: unlikely
15:44.09igcewielingI believe google talk uses XMPP
15:44.14boom^timeit does
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15:44.22igcewielingthen it is unlikely sip.conf will help.
15:44.41igcewielingI keep forgetting wasanzy doesn't read docs
15:44.45*** join/#asterisk leedm777 (~leedm777@nat/digium/x-tcsrqdlxgrasnhwv)
15:45.00wasanzyigcewieling: do you have a link that can show me how to setup just a simple VOIP ?
15:45.12igcewieling~book
15:45.12infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
15:45.40igcewielingI believe that talks about setting up google voice,  If not a simple google search should turn up thousands of hits
15:45.49wasanzyigcewieling: I do read docs and when I don't understand the book is when I ask here
15:45.50boom^timeI still have the first edition of that book somewhere that was giving out at the first astricon.
15:45.53[TK]D-Fenderwasanzy: Google Voice is going to die shortly.
15:46.10[TK]D-Fenderwasanzy: Google is moving it to Google Hangouts and dropping XMPP support
15:46.26[TK]D-Fenderwasanzy: is is pretty much a waste of time
15:46.36wasanzythe main thing am working on is wolfram
15:46.37igcewieling[TK]D-Fender: google voice has been dying shortly for months.
15:46.48igcewielingwasanzy: what the hell is wolfram?
15:47.05igcewielingIs that the evil law firm from Buffy?
15:47.24wasanzyigcewieling: http://nerdvittles.com/?p=798
15:47.49igcewielingah.  we can't really help with that
15:48.47igcewieling[TK]D-Fender: and honestly, based on his questions over the past few weeks I'm perfectly happy for him to waste says of time
15:48.56igcewielings/say/days
15:50.05[TK]D-Fenderigcewieling: Wolfram & Hart
15:50.16igcewieling[TK]D-Fender: ah, silly me.
15:50.28[TK]D-Fenderigcewieling: And the was from Angel (the Buffy spin-off)
15:50.55igcewielingyup.  Amy Acker is AWESOME
15:50.57*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
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15:51.17[TK]D-Fenderigcewieling: Agreed
15:51.18saxaok, it does not work properly yet, i can hear the other side, but it seems the other side does not hear me.
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15:51.37saxaany ideas on where to check this ?
15:51.45igcewielingsaxa: I must have missed the pastebin of your sip.conf [general]
15:51.59saxaprobably i should set a lower resource consuming codec ?
15:52.19[TK]D-Fenderigcewieling: She was in Dollhouse and still in Once Upon A Time (unless something has happened (NO SPOILERS!)
15:52.43[TK]D-Fendersaxa: has nothing to do with codec.  has everything to do with NETWORKING
15:52.58saxaoh ok
15:53.02igcewieling[TK]D-Fender: I watch neither, though if I'd known I might have watched them
15:53.12saxaso then i still have to check that thing
15:53.27igcewielingthere is twice I asked.
15:53.32*** join/#asterisk Gaiax (~Mabel@unaffiliated/gaiax)
15:54.23saxaigcewieling: http://pastebin.com/8aCP25q5 <- here it is.
15:55.13*** part/#asterisk Gaiax (~Mabel@unaffiliated/gaiax)
15:55.26igcewielingsaxa: ther external ip address of your asterisk server is 201.50.103.98 ?
15:56.09igcewielingsaxa: how many g729 licenses do you have?
15:56.24saxayes, right now is that one 201.50.103.98
15:56.46saxaigcewieling: do i have some ?
15:57.07igcewielingsaxa: that is my question.  how many g729 licenses did you purchase?
15:57.11saxaigcewieling: i never bought an g729 licence, unless it cames with the phone
15:57.22*** join/#asterisk glambert (~glambert@37.157.50.80)
15:57.26igcewielingthen you should not have g729 allowed in your codec list.
15:57.32saxaok
15:57.32igcewielingnot your current issue, but it will become one.
15:57.41igcewielingnow do a new pastebin of a failed call with sip debug
15:57.43saxalet me disable it
15:57.51saxaok a moment
15:58.28glambertHi, we have an issue where the ptime the phone and server return in the invite is 20ms yet the RTP ptime from server is reaching 84ms and phone reaching max of 24ms, how do I work out why the server is no where near the negotiated ptime?
15:58.33glambertcodec is g722
16:00.08igcewielingglambert: what version of Asterisk
16:00.26glambert1.4.44
16:02.39*** join/#asterisk petris (~petris@ramnode-vps.srv.petrisdns.com)
16:03.53[TK]D-Fender1.4 is not supported, nor is 1.6.0, nor 1.6.1, nor 1.6.2
16:04.06igcewielingI wish you the best of luck with that old version of Asterisk which is no longer supported and so few people use.
16:04.15saxahttp://pastebin.com/b0tS7g1w
16:04.34saxaigcewieling: this is the new sip set debug peer casasip
16:04.37boom^timeigcewieling, Hey! I have a lot of fond memories of that version of Asterisk.
16:05.01igcewielingboom^time: I have a lot of fond memories of my ex-wife, but sometimes it is time to move on.
16:05.45glambertwe've used it for years and if it ain't broke and all that
16:05.59[TK]D-Fenderglambert: It's broke now.
16:05.59glambertnewer model of phone released and seem to have issues going through IVRs which is frustrating
16:07.09[TK]D-Fender1.4 doesn't actually even transcode G.733... only passthrough
16:07.13coppicenothing seems to last like things used to :-)
16:07.14[TK]D-FenderG7.22*
16:07.27[TK]D-Fendergah
16:07.38coppicethere's G.722 in 1.4
16:07.42glambertthousands of calls going through it per day with no issues and MOS of >=3 on over 95% of calls doesn't indicate broken really, just an issue with this particular model going through IVRs, very strange
16:07.55glambertwe've enabled g722, I believe we patched it
16:08.01boom^timeigcewieling, I was speaking facetiously. I have a lot of fond memories of your ex-wife also.
16:08.05glambertbut not me who would've done that
16:08.28glambertall of our calls use g722, internal, outbound or inbound
16:08.36*** join/#asterisk Chotaire (chotaire@chotaire-home.vipri.net)
16:08.54*** join/#asterisk amizraa (~amizraa@gateway/tor-sasl/amizraa)
16:09.22igcewielingglambert: so no only are you using an old version of Asterisk, you are actually using an old version of Asterisk you applied custom patches to.
16:09.30[TK]D-Fenderglambert: you are running a non-standard patched version of an Asterisk branch which has been off support for some time.  Best of luck
16:11.54glambertThere is a backport for 1.4 for g722 listed on voip-info.org which I can only assume we're using
16:12.07boom^timeI don't understand how difficult could it really be to upgrade to 11.5? It's not like it would take _that_ long to handle the syntaxual changes
16:13.10boom^timewithin the configuration files
16:13.42igcewielingboom^time: doesn't matter.  he won't upgrade to a supported version, most people here won't help with unsupported versions.  end of story.
16:14.44boom^timeBut it's a terrible story. The premise is crap and the ending is depressing.
16:15.43*** join/#asterisk hehol (~hehol@217.9.101.222)
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16:48.13dpeloquinboom^time: it insists upon itself!
16:54.29nam3l3zzwhat could be the problem, x-lite /w username/pass/domain registers easy, but asterisk doesn't, exchanges /w OPTIONS packets, but no REGISTER ?
16:54.37*** join/#asterisk petris (~petris@ramnode-vps.srv.petrisdns.com)
16:55.20[TK]D-Fendernam3l3zz: huh?
16:55.40nam3l3zz[TK]D-Fender:  can i pm ? pastebins
16:55.44[TK]D-Fendernam3l3zz: you say registers easy then "no register".. Please be clear
16:55.57[TK]D-Fenderjust pastebin in public and mask ONLY passwords
16:56.53nam3l3zz[TK]D-Fender: x-lite a win app, sends a register request, but asterisk sends options request and register request doesn't follow, althou register field is defined in sip.conf
16:57.37[TK]D-Fender"register" in sip.conf is for ASTERISk to register to antoher service.  this is NOT for setting up your x-lite\
16:57.48nam3l3zzi know
16:57.49nam3l3zz:)
16:57.51[TK]D-FenderX-Lite registers to YOUR server, not the other way around
16:58.23[TK]D-Fenderyour are not being clear, and not showing configs an debug.
16:59.04nam3l3zzin the scenario, x-lite registers to a remote account , to which /w same account settings asterisk doesn't want to register
16:59.17nam3l3zzi will try to be clear
16:59.34[TK]D-Fendernam3l3zz: most services won't let you register MULTIPLE things to the same acount like that
16:59.39[TK]D-Fendernam3l3zz: SHOW US
17:00.02nam3l3zzi close x-lite and wait , then start asterisk
17:00.07nam3l3zzto avoid conflicts
17:01.00nam3l3zzbut x-lite always registers, and wireshark i can see, that it actually starts from sending a register request packet, but asterisk starts from sending options request & never sends register after
17:01.12[TK]D-FenderStill not seeing configs....
17:01.18igcewielingnor sip debug
17:01.30[TK]D-Fenderand OPTIONS has NOTHING to do with registration to an outside service
17:01.43[TK]D-Fenderthat is due to "qualify" being set in a peer.
17:01.57[TK]D-Fenderand your peer definitions have nothing to do with asterisk registering to anything
17:03.44nam3l3zzi get that, i'll try to be more specific, can asterisk be against me having quotes in the username ?
17:03.44*** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)
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17:03.51nam3l3zz*dot
17:04.08nam3l3zznot quotes
17:04.53[TK]D-Fenderawaits getting the configs
17:05.04igcewielingnam3l3zz: less talk, more do.
17:05.37nam3l3zzsent stuff with passwords to pastebin.....
17:11.06nam3l3zzhttp://pastebin.com/rPefaWAw
17:11.07[TK]D-Fender...
17:11.38[TK]D-Fendernam3l3zz: your second register is DEAD
17:11.52nam3l3zzbecause of the quotes
17:11.53nam3l3zz?
17:11.59[TK]D-Fendernam3l3zz: ALL registers have to appear after everything else in [general] and BEFORE the first entry
17:12.10nam3l3zzaffirmative
17:12.11nam3l3zzthanx
17:12.35*** part/#asterisk navaismo (~navaismo@189.241.68.101)
17:12.43*** join/#asterisk navaismo (~navaismo@189.241.68.101)
17:12.49nam3l3zzfound plenty shit examples /w register allover, thnx again
17:13.10navaismoIs there a way to ignore and no transmit dtmf when the call is bridged?
17:13.11igcewielingnam3l3zz: I bet the example in sip.conf.sample is correct.
17:13.26igcewielingdoesn't ANYONE read the .conf.sample files??????
17:13.30[TK]D-Fenderyour provider entries should all be "nat=no", canreinvite was changed to "directmedia" Wway back in 1.6
17:13.31nam3l3zzmost likely, 2 much trust 2 google almighty
17:13.59[TK]D-FenderGoogle doesn't write Asterisk guides
17:14.03[TK]D-Fenderdon't say "google"
17:14.16nam3l3zzpatent breach ?
17:14.19nam3l3zz:)
17:15.56nam3l3zz[TK]D-Fender: although I have nat=yes in general, i need to put nat=no in every peer ?
17:16.10igcewielingnam3l3zz: just the ones not behind nat
17:16.43nam3l3zzfound myself being a noob in this one
17:17.08nam3l3zzpeers are remote, /w global ips, so not behind nat... nice
17:21.43*** join/#asterisk yano (yano@freenode/staff/yano)
17:24.04nam3l3zzigcewieling: u may know, i can see in debug of sip, sip server reply 401 for dialing a number via it, but asterisk keep retransmitting dial request
17:25.06navaismoanyone? some hints?
17:25.21nam3l3zzigcewieling: http://pastebin.com/0ZmyjVrJ
17:26.51igcewielingyou still have a NAT issue
17:27.09igcewielingdid you read https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions ?
17:28.05nam3l3zzhttp://pastebin.com/uMVMY3W1
17:28.10nam3l3zzit seems like a nat issue
17:28.15nam3l3zzwill read, thanx
17:30.23igcewielinghas been working on an ASCII over DTMF protocol. Why? Because I can. 8-)
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17:34.14nam3l3zzigcewieling: got different reply http://pastebin.com/snbz7M3X , could nat be sorted ?
17:34.30coppiceI used to build systems that allowed ASCII and Chinese entry by DTMF
17:34.42igcewielingnam3l3zz: unlikely
17:34.59nam3l3zzgot it , caller id doesn't match server's requirement
17:35.26igcewielingcoppice: I'm converting the text to base-16 encoding, replace all "f" with * and all "e" with #, wrap it in a packet with crc and retransmits and send it
17:35.53coppiceoh, automated entry. I used to let people enter messages by hand
17:36.04nam3l3zzno, actually it is a registered number :\
17:36.16igcewielingcoppice: *nod*  this is mainly for machine-machine communication.
17:36.33igcewielingI think I get something like 5cps
17:37.08igcewielingnam3l3zz: forbidden is a config issue
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17:38.29saxaigcewieling: had a chance to take a look at the second debug output ?
17:40.04igcewielingsaxa: still have a NAT issue.  since the source port is 5060 I can only assume SIP SPI or SIP ALG on the router
17:40.25igcewielingthough it could easily be something like a broken softphone too.
17:48.44coppiceigcewieling: I built a paging for the deaf system. deaf chinese were actually prepared to learn what are termed the chinese telegraph codes - numeric codes for each chinese character - and use those codes with a DTMF phone to send Chinese pages to their friends
17:49.32igcewielingcoppice: nice!  Users willing to learn!  I didn't know they existed.
17:50.06*** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell)
17:50.17igcewielingWe need an Allison sound file "Watson come here, I want to see you."
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17:53.50bananapieMy SIP provider gave me a backup IP, I have the primary IP in the "host" field. How do I enter the backup IP into the SIP peer without modifying the dialplan? Thanks.
17:54.57igcewielingbananapie: you can't.
17:55.01coppiceigcewieling: deaf people are willing to do quite a lot
17:55.14igcewielingcreate another peer, use your dialplan to failover
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17:56.29bananapieigcewieling : I read that solution already, I was hoping to not use it. Thanks!
17:56.51bananapieI think if I use a dialplan macro, I could do it in such a way that would be easy to use
17:57.24igcewielingbetter to use a gosub
18:01.39saxaigcewieling: ok, will check that thing, SIP SPI or SIP ALG on my router. But its not a softphone
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18:46.40igcewieling"Thank you for your unsolicited contacting of me about a job.  I'm not interested, but as a courtesy I've added your contact information to our Direct Marketing List"
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18:47.24jmetrooh yes
18:47.26jmetrooh yes please.
18:47.32jmetroI have signed you up for Cat Facts.
18:47.50igcewielingjmetro: I'm being nice, only signeing them for OUR DMA list.
18:55.47jmetrohttp://tinyurl.com/84laak3
18:56.41SuperNulljmetro someone on reddit was doing that to me for a week..
18:58.55jmetroTo cancel, reply "ÐØѱ" within 30 seconds
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20:57.31ChannelZBORED
21:02.37saxaigcewieling: this is a third debug i'm sending , i doublechecked that no SIP SPI or other thing is enabled on the phone side
21:02.41saxahttp://pastebin.com/ZcmesZji
21:03.15saxaigcewieling: on the * side I'm 99% sure there is no one filter or anything related to SIP enabled.
21:03.52saxaigcewieling: i will double check it again, but i'm nearly 100% sure i have disabled all those things
21:04.15saxaanother thing is that on the phone side a even disabled the firewall.
21:10.53*** join/#asterisk iTrojan (~Adium@41.233.28.50)
21:13.21iTrojanplease help
21:13.27iTrojani keep getting this error ;
21:13.28iTrojan<PROTECTED>
21:13.34iTrojanhttp://pastebin.com/YzFu9Bd1
21:13.42iTrojanthis si my configurations
21:14.02[TK]D-Fenderand just like it says... no match
21:14.03iTrojanit is driving me nutes
21:14.06iTrojannuts
21:14.10[TK]D-Fender"nutes" indeed
21:14.26[TK]D-Fenderbecause it isn't spelled EXTERN either....
21:14.45iTrojanwhat is it then!!
21:14.52[TK]D-Fenderand [new] should not "include" .... ITSELF
21:14.59[TK]D-FenderEXTEN <-----------
21:15.06iTrojanoh lol
21:15.08[TK]D-Fenderho do you not know this?
21:15.12[TK]D-Fenderhow*
21:15.18[TK]D-Fender~book
21:15.18infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
21:15.19[TK]D-Fender^^^^^^^^
21:15.25iTrojanhmm maybe because I just learned about this 10 minutes ago?
21:15.45[TK]D-FenderI'd be interested to see who else wrote it that way
21:15.47iTrojanI already have a book
21:15.51iTrojanbut I want to do this hello world
21:15.56iTrojanto get a feeling of pprogreessing
21:16.06iTrojanI googled alot
21:16.10iTrojannothing works
21:16.24[TK]D-Fender"nothing works"?
21:16.33[TK]D-Fenderwhat kind of "nothing"?
21:16.59iTrojancan you leave your saracsm ?
21:17.07iTrojansarcasm !?
21:17.07igcewielingedges away and gets out the garlic
21:17.16[TK]D-FenderI'm just not sure what you mean by "nothing works"
21:17.31[TK]D-Fenderso far I saw adialplan error I already gave you an answer for
21:17.38[TK]D-FenderHave you already applied the changes and retested?
21:17.49iTrojanhttp://pastebin.com/YzFu9Bd1
21:17.51iTrojanoh not this
21:18.01iTrojanallowexternaldomains=yes
21:18.05iTrojanlike setting this option
21:18.17[TK]D-Fenderthat's the same pastebin
21:18.27iTrojanyeah, I said not this..
21:18.34iTrojanI meant to paste the option
21:18.36[TK]D-FenderSo go fix it and see if it works
21:18.42iTrojanwhat to fix?
21:18.46[TK]D-Fenderyour dialplan
21:18.52[TK]D-FenderI already told you what your typo was
21:18.55iTrojanto change what?
21:18.58[TK]D-FenderDidn't you go and fix it?
21:18.59iTrojanah that is it?
21:19.01iTrojanexten ?
21:19.03[TK]D-FenderEXTERN is not right
21:19.06[TK]D-FenderEXTEN
21:19.08[TK]D-Fenderis
21:19.10[TK]D-Fenderno "R"
21:19.14iTrojanok ok let me do it
21:19.17[TK]D-Fenderall those lines were wrong
21:19.40iTrojanaleast extern has a meaning
21:19.47[TK]D-FenderEXTENsion <-
21:19.54iTrojanEXTERNal
21:19.57[TK]D-Fenderhas a meaning that isn't related
21:20.01[TK]D-Fenderthat is an EXTENsion
21:20.10[TK]D-Fenderas in EXTENSIONS.conf
21:20.29*** join/#asterisk asteriskmonkey (~darlek@server.xcel.on.ca)
21:20.30iTrojanhm that makes sense
21:20.41asteriskmonkeydoes callerid routing not work in realtime?
21:20.51iTrojanwow that works
21:20.57asteriskmonkeyi have exxten => did/callierd .... dosnt seem to work
21:21.12iTrojanalthough you're not so 'friendly' , but thank you [TK]D-Fender
21:21.55[TK]D-FenderiTrojan: do remove that "include" as well
21:22.03*** join/#asterisk SGjunior (~sgjunior@96.127.222.20)
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21:24.28zozeerI have a random question about call forwarding.  if 20 calls 21 and 21 is forwarded to 22 and 22 is forwarded to 23 and 23 does not answer will it drop me in 23's mailbox, or 20's mailbox?
21:25.13[TK]D-Fenderzozeer: it goes wherever your dialplan tells it to go.
21:25.22[TK]D-Fenderzozeer: what did you tell it it do for that?
21:25.33zozeerso I can nest as far deep as I want?
21:25.54[TK]D-Fenderthese are phones bouncing calls....
21:26.10[TK]D-Fenderit bounces as far as your devices keep pointing elsewhere
21:26.21zozeerhttp://www.voip-info.org/wiki/view/Asterisk+call+forwarding  example 5
21:26.25[TK]D-Fenderor until where you bounce them to doesn't try to dial another
21:26.37zozeerI was having a debate with our old nortel guy.
21:27.03[TK]D-Fender* call processing has nothing to do with the "Nortel Experience"
21:27.17[TK]D-FenderYou can an infinite loop any time you want
21:27.25[TK]D-FenderDialplan = programming = whatever you tell it to do
21:27.41[TK]D-FenderSo YOU are the limit of what you want to do
21:27.47*** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch)
21:28.01[TK]D-FenderSo if you copied that code verbatim.. trace it.  Where do you see it going?
21:28.22zozeer[TK]D-Fender, would that form a loop?  as in it just keeps bouncing down the line?  (I am a router guy so some of this is a little counter to what I know)
21:28.33[TK]D-Fendertrace the code...
21:30.32[TK]D-FenderThat sample is also pretty broken
21:30.51*** part/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch)
21:31.27[TK]D-FenderAnd this is a sample for "dialplan controlled callforwarding" as a concept
21:31.33[TK]D-FenderAs opposed to "device-based
21:33.31asteriskmonkeydoes pattern matching not work in realtime?
21:34.38jmetroeverything works in realtime
21:34.43navaismoasteriskmonkey i have used realtime static and that works
21:35.03navaismoyou should show us your cli output
21:35.05asteriskmonkeyi have in my realtie db did/callerid and it dosnt seem to get hit
21:35.30asteriskmonkeycallerid based routing with the / method dosnt seem to get hit in my realtime :/
21:35.33asteriskmonkeyis that a bug?
21:35.40[TK]D-FenderasSHOW US
21:35.54[TK]D-Fenderand be specific on versions
21:35.57[TK]D-Fenderyou know better.
21:36.03asteriskmonkeylol
21:36.20*** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:98bc:4a61:946:6c89)
21:37.09asteriskmonkeyversion 10.3.0, dialplan (switch statement then a catchall, the realtime i have an entry based on callerid which isnt getting hit, the catch all in teh flat file is taking it)
21:37.56asteriskmonkeyexmple in my realtime would be 1234/555 Playback ttweasels..
21:38.15asteriskmonkeythe call iv nooped i can see correct criteria.. so i know its just not catching in realtime
21:38.24[TK]D-Fenderreal DB dump, real call with core debug
21:38.33[TK]D-Fendernone of this pseudo nonsense...
21:39.09[TK]D-Fenderbecause that first answers comes back as "your DB values are wrong and has more errors than syntax" :p
21:39.18asteriskmonkeythanks, ill do further beating and dump, though id just do a quick poke to see if the exten/callerid was buster like it was in 2006 :)
21:40.13*** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:98bc:4a61:946:6c89)
21:40.21navaismoIm going to ask for the last time..today: Is there a way to ignore & dont transmit dtmf to the called party on a bridged call?
21:41.15jmetrohm
21:41.22jmetroyou mean dont let the called party hear dtmf
21:41.23*** join/#asterisk imcdona (~Thunderbi@2001:470:e8f1:2:98bc:4a61:946:6c89)
21:41.31jmetrobut also asterisk doesnt hear dtmf?
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21:46.36*** join/#asterisk navaismo (~navaismo@189.241.68.101)
21:46.58navaismojmetro yes
21:47.21navaismodont let the called party hear dtmf and also asterisk ignore them
21:47.43*** join/#asterisk kresp0 (~kresp0@109.Red-79-144-71.dynamicIP.rima-tde.net)
21:48.43jmetroI would expect it to be in core show app dial
21:48.45jmetrobut i dont see anything
21:48.59navaismonope there is nothing about that
21:49.01imcdonaI'd like to remind everyone that freenode supports connecting via SSL. If you have trouble connecting via SSL on the default port, you can try these: on ports 6697 (SSL only), 7000 (SSL only), 7070 (SSL only)
21:50.00jmetronavaismo: sorry nav, dont see anything in my google-fu
21:50.18navaismono problem,  neither do i
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21:58.37saxaok, i understand that in some way my dtmf tones do not get transmitted
21:58.46saxaor at least they do not reach asterisk
21:58.59saxadtmfmode=auto
21:59.26saxabut when i try to access my mailbox it doesnt recognise the password
22:01.25saxaany ideas what should i look at ?
22:05.19newtonrsaxa, You should figure out what style of DTMF your endpoint is using when talking to Asterisk. You should also look at https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information. With that information you'll be headed down the right path
22:06.41saxa[Aug 19 19:55:40] WARNING[21114]: app_voicemail.c:9834 vm_authenticate: Couldn't read username
22:06.59saxanewtonr: thanks
22:14.21saxahttp://pastebin.com/4z7v5FVH
22:14.46*** join/#asterisk mardok45 (~mardok@74-140-49-109.dhcp.insightbb.com)
22:15.00saxaThis is a sip debug on the casasip peer who seems doesnt get recognised the dtmf tones
22:16.06[TK]D-Fenderdon't assume "auto" will work
22:16.17[TK]D-FenderSet the proper mode yourself
22:18.58mardok45I'm using Asterisk for my Google Voice phone.  Is there a way to also is it for SMS?  I'm using FreePBX.
22:19.08mardok45to also use it for SMS?*
22:20.38jmetroyou should check #freepbx
22:20.46jmetroas we cannot accurately support freepbx config files here
22:20.55tm1000jmetro: +1
22:21.02tm1000a plesant response! :-)
22:21.42jmetroWell who says anyone has to be nasty anyway :3
22:22.29mardok45Linus Torvalds?
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22:23.33tm1000mardok45: ha!
22:24.36[TK]D-FenderNo, with linus it's simply bundled with the package
22:24.56[TK]D-FenderAnd no, FreePBX does not support SMS
22:25.04saxa[TK]D-Fender: thx
22:27.54newtonrsaxa, get a pcap of SIP and RTP traffic, if your endpoint is doing rfc2833 then read up on that and look at your traffic to see if its hitting Asterisk.
22:28.31newtonrsaxa, also, if it is hitting Asterisk, turning on DTMF debug (see logger.conf) may give you additional info.
22:31.07saxahttp://pastebin.com/L13rqUbf this one is done with dtmfmode=rcf2833
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22:31.57saxanewtonr: ok many thanks, will try also with the dtmf debug
22:32.41saxagoes to eat a pizza, l8r i'm back here, thx for now.
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23:03.13igcewielingNothing wrong with gently pointing people to #FreePBX, but if they don't listen we get out the garlic and wooden stakes and take care of the problem.
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23:10.54PreludeZzzzhello everyone
23:11.12PreludeZzzza quick question if someone can guide me in the right direction
23:11.29PreludeZzzzi want to send an audio tone to a particular live asterisk channel ( one side i guess ) .. how do i do that ?
23:11.50PreludeZzzzsay someone is on a call and its been more then say 1/2 hour.. maybe i wnat to send a warning beep that they have exceed a particular time window on a call
23:11.56PreludeZzzzhow does one send a tone to a live channel that way ?
23:15.58igcewielingfor that SPECIFIC thing use the options to Dial()
23:16.03ChannelZDial() can do time limits with warnings
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