IRC log for #asterisk on 20130108

00:00.50tzangerI did notice on my own phone that it tries for the better part of an hour to contact the boot server (in a loop) until finally giving up and using whatever config it has on hand
00:00.55tzangerI'm running OLD ip501 firmware though
00:01.29tzangersorry ip601 and sip firmware 1.6.7
00:03.02[TK]D-Fender[18:37][TK]D-FenderSo make a new provisioning folder and dump in stock sample configs & firwmware for it to pick up  [18:37]Get_The_FishI cant do that.  [18:38]Get_The_FishBecause I dont have unfettered access to this entire subnet, and since it's looking for a boot server on a 10.x.x.x network, and is now on a 192.168 subnet, it fails.
00:03.22[TK]D-FenderHe's saying up-front that he doesn't have the means of following advice.
00:03.30tzangeryeah, same idea as my phone
00:03.50tzangerit moved from the house to the shop and can only use its internal config until I get off my ass to change things around
00:03.51[TK]D-Fender[18:41]Get_The_FishNot to mention the fact that it is so old I wouldnt even know which firmware to use on it.
00:04.28[TK]D-FenderAnd seems to confirm that he can't read docs on what it even supports.  He simply appears to have no clue what it runs, no means of doing anything
00:04.38[TK]D-FenderAnd *I* get crap from him.
00:06.05tzangeryou're to blame, don't you know? :-)
00:06.15[TK]D-Fendernaturally.
00:07.06[TK]D-Fenderhttp://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
00:07.09[TK]D-Fender^
00:07.21[TK]D-FenderCause yeah ... RAW-CAT SIGH HENCE
00:09.11[TK]D-Fendertzanger: 1.6.7 was the one that fixed the presence issue with IP601's and side-cars,e tc.... something to that effect...
00:09.21[TK]D-FenderI remember that one.
00:10.31tzangeryeah. I have no sidecar but it was one of the first firmwares that did presence support, which I never did get around to using. :-)
00:10.43[TK]D-Fendertzanger: This past week brought my IP600's to SIP 3.1.8 (latest they support) which is from Mar 2012.
00:11.04[TK]D-FenderSo I guess :not supported" depends on a certain point of view...
00:11.25[TK]D-FenderWhci isn't bad for 7 year old phones....
00:12.54tzangerhm, maybe I should update
00:13.13tzangerthe only thing I really wish the 501 had was backlight
00:13.19tzangerit's a pretty nice phone
00:13.24tzangerer 601 dammit I keep saying that
00:15.16[TK]D-FenderThat's the reason I swapped for the IP335 I run now.
00:15.41[TK]D-FenderWhen I change offices in the next 2 months renovations I should end up in a room with good lighting and may switch back
00:19.31rue_workanyone ever used mgcp or am I the only one?
00:19.37rue_workok I have two mgcp gateways that I'm trying to link calls between, but I cant get them both to work at the same time with asterisk
00:19.46rue_workI dont know if mgcp.conf is limited to one gateway
00:19.53rue_workI can get one to work
00:20.02rue_workand it can be either one
00:20.10rue_workbut not both
00:22.56[TK]D-Fenderhttp://www.quickmeme.com/meme/3sh6mz/
00:23.49*** join/#asterisk Sinnerman77 (~Sinnerman@174-21-246-21.tukw.qwest.net)
00:30.32Sinnerman77Hi there.  I'm running Elastix and using Flowroute as my SIP trunk.  This is a new setup and I'm pretty much a newbie.  I have the trunk setup according to instructions from Flowroute, an outbount route setup using the "local 7/10 digit" wizard and an extension.  I have the IP address for the PBX set as allowed on Flowroute's website.  When I try to dial out using X-Lite I'm getting "All circuits are busy now."  Is there something eas
00:32.15jacekowskiryan42: i would use show command to make sure it's all loaded correctly
00:32.27jacekowskibut it looks ok
00:38.28[TK]D-FenderSinnerman77: PASTEBIN the failed call attempt @ * CLI
00:38.30[TK]D-Fender~pb
00:38.30infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
00:38.32[TK]D-Fender^^^
00:38.34[TK]D-Fender^^^
00:40.51*** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano)
00:45.52Sinnerman77[TK]D-Fender, thank you.  I'll do that in a moment if I need to.  I'm on the line with Freeroute support.  He thinks it's my dial pattern that's the problem.
00:48.14[TK]D-Fenderheads out for a while
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03:38.04macroevolvehi, i've been trying to find a softphone that supports Silk @ 8kbps.  I've already tried Ekiga, Jitsi, Vireo, and Microsip, and none of them work well.  Was wondering whether knew of any other softphone I could try?
03:39.29jpsharpThat's not really a standard codec in internet telephony.
03:39.54jpsharpSo you're going to have a PITA time finding something that supports it.
03:40.11macroevolvejpsharp:  yeah - ive alreadd experienced that PITA part unfortunately
03:42.16*** join/#asterisk Docfxit (~Docfxit@netblock-75-79-6-10.dslextreme.com)
03:44.53DocfxitI would like to record a voice prompt.  I added extension 130 to extensions.conf. The lines I added are at pastebin.com/JJPYZwRx  I have restarted Asterisk.  It isn't working.  Could someone please suggest what I need to do to get it working?
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03:47.05pabelangerDocfxit: you recorded asterisk-Recording, but playback asterisk-recording
03:47.43DocfxitGreat catch
03:47.49DocfxitI'll test it.
03:49.00asr33shakes fist at uppercase
03:56.12*** join/#asterisk FireAndIce (~FireAndIc@123.201.83.153)
03:56.28leifmadsenthere's something to be said about using a channel variable there....
03:56.59leifmadsenat line 4 I'd have placed a Set(thisRecordedFile=/tmp/asterisk-recording) or something
03:57.11leifmadsenthen used ${thisRecordedFile}.gsm on the next line
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04:18.14DocfxitWhen trying to restart asterisk:
04:18.14Docfxitroot@UbuntuAsterisk:~# asterisk -r
04:18.14DocfxitI get:
04:18.14DocfxitUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
04:18.14DocfxitIf I input this:
04:18.14Docfxitroot@UbuntuAsterisk:~# asterisk -vvvc
04:18.16DocfxitThe errors I get are:
04:18.18DocfxitUnable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
04:18.20DocfxitUnable to bind socket to /var/run/asterisk/asterisk.ctl: No such file or directory
04:18.23Docfxit[Jan  7 20:33:07] NOTICE[7261]: http.c:611 http_server_start: Unable to bind http server to 0.0.0.0:8088: Address already in use
04:18.25Docfxit[Jan  7 20:33:07] WARNING[7261]: manager.c:3159 init_manager: Unable to bind socket: Address already in use
04:19.29DocfxitAny idea what I can do to fix these errors?
04:21.15ChannelZdoes /var/run exist?
04:21.48DocfxitYes.
04:23.52[TK]D-FenderDo you see an Asterisk process actually running?
04:23.57[TK]D-FenderCan you find the PID somewhere else?
04:24.12[TK]D-FenderDid you look where whatever you used to init it chooses to put it?
04:27.25DocfxitAsterisk does run.  It answers the phone, plays the prompts, transfers calls.
04:28.39DocfxitWhat command would I use to find asterisk.pid?
04:29.41dpilondo this: ls /var/run/asterisk/*.pid
04:30.39DocfxitNo such file.
04:31.38DocfxitI found someone else with the same problem and they said SELINUX=disabled did the trick.
04:31.58DocfxitWhat is SELINUX=disabled?
04:33.26[TK]D-FenderDocfxit: look in var/run
04:35.10DocfxitIn var/run/asterisk there is a file asterisk.ctl with zero bytes
04:36.14DocfxitIn var/run there are a number of .pid but no asterisk.pid
04:36.15[TK]D-FenderI mean the base of it
04:36.48[TK]D-Fenderwho owns the PID's in /var/run/asterisk ?
04:37.24[TK]D-Fendersrwxr-xr-x  1 asterisk asterisk    0 Dec 10 20:39 asterisk.ctl
04:37.26[TK]D-Fender-rw-r--r--  1 asterisk asterisk    5 Dec 10 20:39 asterisk.pid
04:37.36[TK]D-Fenderthat's mine
04:37.37DocfxitThere are no pid's in /var/run/asterisk only one file asterisk.ctl
04:38.04[TK]D-Fenderperhaps you should kill * and restart it
04:38.45Docfxit/var/run/asterisk/asterisk.ctl -rw-rw-rw-
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04:41.52DocfxitIs it normal to get 5 lines after the kill *
04:42.52Docfxitroot@UbuntuAsterisk:~# kill *
04:42.52Docfxit-bash: kill: callfile: arguments must be process or job IDs
04:42.52Docfxit-bash: kill: configackup.tar.gz: arguments must be process or job IDs
04:42.52Docfxit-bash: kill: hpec-9.00.007-prescott.tar.gz: arguments must be process or job IDs
04:42.52Docfxit-bash: kill: register: arguments must be process or job IDs
04:42.52Docfxit-bash: kill: zaptel-backup.tgz: arguments must be process or job IDs
04:43.03Penguin~pb
04:43.03infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
04:43.35DocfxitSorry.  I will do that.
04:43.45dpilonouch...the meant... kill the actual process  not a *
04:44.07dpilonps axf |grep asterisk    get the id
04:44.22dpilonthen: kill -9 <id>
04:44.36Penguinpkill asterisk   <------
04:44.49dpilonthat works too
04:45.00PenguinIt's a lot less bother.
04:45.43PenguinAlso, pgrep, not ps|grep
04:45.55Penguinlrn2sysadmin
04:47.45*** join/#asterisk radic (~radic@dslb-088-065-148-032.pools.arcor-ip.net)
04:48.28DocfxitAfter pkill asterisk  I tried asterisk -r  I'm getting the same error
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04:48.55DocfxitUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?
04:49.09DocfxitThat file is the only file in that folder.
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04:50.03rue_houseanyone have an answer for my mgcp problem?
05:02.39DocfxitWhere is a good place to share a jpg for you?
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05:04.39Scryehas anyone ever experienced a 7960 that will not attempt dhcp and will not factory reset?
05:04.49Penguindocfxit: imagebin.org
05:07.53DocfxitPlease see imagebin.org/242086
05:08.10ryan42[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~[6~
05:08.20ryan42err, oops
05:08.24DocfxitThat is the only file in /var/run/asterisk
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05:34.45asteriskmonkeyim having a bit of difficulty with cut, can anyone lend a ahnd?
05:36.52rue_houseI have an mgcp problem, I cant get two gateways to work
05:37.29*** part/#asterisk asteriskmonkey (~philip@206.51.27.151)
05:39.02[TK]D-Fenderpatience[-1]
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05:45.45rue_house[TK]D-Fender, your awake! sweeet
05:46.00rue_houseI couldn't hang by the keyboard at work
05:46.18rue_houseafter the first time I asked some guy needed the computer and I had to make an appointment
05:46.37rue_housethe second time I came back and was told the room was being locked up
05:46.41rue_houseso now I'm at home
05:46.51rue_housenice laaaaaaaaaaaaaaaaaarge scroll buffer
05:47.06rue_houseand a bot I can get kicked for dragging in
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07:21.26schmidtsgood morning
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07:23.13bombevgood morning guys :)
07:25.22ChannelZWell, average.
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07:50.31gavimobileI cannot register one of my sip peers, when I try to debug this peers it says Unable to get IP address of peer '0000FFFF0000'
07:50.38gavimobilefrom my pc, I can ping this device
07:54.48kaldemardon't try to enable debug for the device only
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07:56.39gavimobilekaldemar: that what im doing now
07:57.10gavimobilekaldemar: I still don't see anything though
07:59.34kaldemardid you successfully enable sip debug?
08:00.09gavimobilekaldemar: sip set debug on is set
08:00.30gavimobileand im monitoring it to see if I see anything related to the ip or the peer name of this peer
08:00.32gavimobilebut nothing
08:01.36*** join/#asterisk Neptu (~Neptu@mail.avtech.aero)
08:02.11gavimobileI did run sip unregister 0000FFFF0000 when my peer was still unregistered
08:02.46kaldemarsuprprise, unregister does nothing then the peer is not registered.
08:02.46gavimobilehowever I did sip quality peer 0000FFFF0000 after, also restarted asterisk
08:03.00gavimobileI also restarted my pap device
08:03.19kaldemarif you're not seeing messages from the device, then asterisk is not getting any.
08:03.31gavimobileI can ping the device though
08:03.52kaldemarping won't help you if the device is misconfigured.
08:04.41gavimobilekaldemar: ok, so ill re-configure the device but I would like to know what caused the problem if possible.. it was working fine, and today all I did was turn it on after a power outage
08:06.10*** join/#asterisk hehol (~hehol@2001:1438:1009:200:f949:40f4:6f06:2fea)
08:06.28kaldemarif the device does not send messages to asterisk, you won't find a reason for it by digging around on your asterisk server.
08:08.05gavimobilekaldemar: so I should be looking on google for reasons why my spa2102 won't registrer
08:10.18ChannelZWhich side, the FXO or FXS?  Have you made sure there's a proxy typed in for either/both, that it's Line Enabled?
08:10.43schmidtsgavimobile you should first take a look at the status page what it says there about the state of line 1, if it says something like unreachable you know the pap cant reach your asterisk
08:11.05gavimobileChannelZ: my device doesn't have an fxo port only 2 fxs ports
08:11.30gavimobileschmidts: first page of the pap2 device settings?
08:11.42kaldemargavimobile: you should look at the device itself.
08:11.55gavimobileonce again there was no change in the device. im guessing it needs a firmware update
08:12.07kaldemarand not waste time googling for possible reasons.
08:12.31*** join/#asterisk kleszcz (tick@linuxmafia.pl)
08:12.40ChannelZoh right, mixing up the 3102.  Same basic question applied anyway.
08:12.47ChannelZapplies
08:15.40gavimobilekaldemar: im looking at the device, there's only 2 lights on it.. in the settings page I do see that it says under status that registration failed
08:16.49kaldemarcheck addresses. where is it supposed to register itself?
08:17.58gavimobilekaldemar: you mean the domain names? sip.myserver.com
08:18.20gavimobileill change it to a local address just for testing
08:18.26gavimobileshould have thought of that
08:18.42ChannelZshakes his head
08:19.23gavimobileRegistered SIP '0000FFFF0000' at 192.168.0.102:5060
08:19.43gavimobilebut why won't the remote address work?
08:19.53ChannelZAsk your DNS?
08:19.53kaldemarwhat "remote address"?
08:20.34ChannelZOr perhaps the SPA doesn't even know a DNS server to ask.  (psst: *something* changed)
08:21.58gavimobileChannelZ: dns isn't configured on the device, its set to dhcp
08:22.31ChannelZkeep going
08:22.37kaldemarthose are not mutually exclusive.
08:23.25gavimobile?
08:23.28*** join/#asterisk vlad_starkov (~vlad_star@83.149.8.144)
08:23.49ChannelZIt gets -- or should get -- one from the DHCP server.  It'd be on the Status tab as well.
08:23.57kaldemarsomething using dhcp does not mean that it does not have dns configured.
08:24.15kaldemarif it does not, your network is dumb.
08:25.37gavimobilekaldemar: I just looked and the dns is set to the default gateway from my router which should be fine
08:26.42ChannelZ'should be'?  Is the DNS server running, and/or returning the right IP for whatever hostname was in there?
08:26.56kaldemarand what address (if any) does the default gateway give for the domain name you had set?
08:28.22Maliutadepends on how you conf your dhcp server. There is no reason you _have_ to have the dhcp server actually provide a DNS server (or two), it could use a static ... but then it's more sensible to provide the servers via DHCP/BOOTP
08:28.31gavimobilehere is the status information from my default gateway (my router) http://pastebin.com/tukryXGy
08:28.52MaliutaIs the port on the switch configured to the right vlan?
08:29.06kaldemargavimobile: irrelevant
08:29.12Maliuta(mow I'm just being difficult)
08:29.43Maliutayou assigned the device a /32?
08:29.51gavimobileMaliuta: yes, the switch connects to the router directly
08:30.05gavimobileMaliuta: these are the default settings
08:30.15ChannelZWell you've answered your own question insomuch as putting in the IP worked.  So it's a bit up to you to figure out why it doesn't otherwise based on your network and servers and what is what
08:30.29MaliutaI'd say that something is wrong right there, unless the device has a decent routing table it won't know where to route packets to the gateway.
08:31.08bulkorokare there any good (voicemail-)prompts ins spanish out there!?
08:31.25gavimobileMaliuta: so what would you recommend?
08:31.32Maliutabulkorok: the standard * ones aren't good enough
08:31.51bulkorokspanish ones?! I'll check that
08:32.17Maliutagavimobile: well I don't know what your network looks like. But I'd suggest it should at least be on a /30 or /29 that the gateway also has an IP on
08:32.22bulkorokwhere to download!?
08:32.45ChannelZhttp://downloads.asterisk.org/pub/telephony/sounds/  ?
08:32.46gavimobileMaliuta: that's a static ip we have
08:33.01bulkorokgreat :-)
08:33.09kaldemargavimobile: start by finding the problem. try a DNS query for the value you had configured in the pap.
08:34.36gavimobilekaldemar: http://network-tools.com/default.asp?prog=dnsrec&host=sip.shn.co.il
08:34.51Maliutagavimobile: unless there is point-to-point link on that device it's not going to be routing properly. Are the DNS servers your upstreams? or in house?
08:35.33gavimobilemy dns server points to two different locations. to a web server and to my network where my sip server is located...
08:35.46Maliutagavimobile: can you ping the device and get a response? if you can't then your problem is it's network config (and yes you can supply a borked network config via DHCP/BOOTP)
08:35.47kaldemargavimobile: from your network of course, where the device is. and using the dns server that the pap has in its config.
08:35.56gavimobileMaliuta: yes I can ping the device
08:36.49Maliutagavimobile: can you telnet/ssh into the device?
08:37.01gavimobileMaliuta: never tried
08:37.15*** join/#asterisk TobSnyder (~schneider@146-52-60-185-dynip.superkabel.de)
08:37.22Maliutagavimobile: it's a Cisco/Linksys right?
08:37.40gavimobileMaliuta: today its cicso it was once linksys
08:37.40gavimobileright
08:38.58Maliutagavimobile: same diff. There should be a telnet port open (or you can open it with a tftp config). I'd be going in and trying a bunch of stuff from the console
08:39.15Maliutagavimobile: it's standard cisco stuff, the commands are all self help
08:39.46ChannelZthis is off track
08:40.03Maliutagavimobile: it might give you a better feel for where the problem is. I know that using telnet into my Cisco handset has helped in the past
08:40.29MaliutaChannelZ: I'm trying to give him all the diagnosis tools for the device
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08:41.42MaliutaChannelZ: to me the network config looks borked. That would stop it talking to the DNS and SIP servers, and anything else in the world
08:44.18gavimobileapparently I cannot ssh or telnet inside
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08:45.04gavimobileMaliuta: what about other devices that are behind the same nat and DO work
08:45.20Maliutagavimobile: it is a feature that can be turned on and off. I use TFTP to configure my devices.
08:45.21gavimobileif the network config was faulty, other local peers shouldn't be working either
08:45.35gavimobileMaliuta: yea, I scanned the settings for this feature
08:45.58Maliutagavimobile: do they have the same configs? i.e. /32 addresses? do they have routing tables? too many variables
08:46.34MaliutaI'm only going to be here until I finish eating, then I'm off to play Go for a couple of hours.
08:46.43gavimobile:-/
08:47.11Maliutagavimobile: are you assigning this device an address out of the same pool as the other devices?
08:47.26gavimobileMaliuta: yes
08:47.38Maliutagood start
08:47.40gavimobiledhcp gives it that address anyways
08:47.59Maliutaso I'm assuming the address you pastebin'd is not the address on the device
08:48.31Maliutagavimobile: you're assigning out of what? a /24 a /29?
08:48.48gavimobileI don't know cidr blocks :-(
08:48.48Maliutaand then doing NAT somewhere.
08:49.04Maliutawhat is the last 3 digits in the subnet mask?
08:49.05gavimobilebut the same thing that any basic router will do 192.168.x.x
08:49.09gavimobile0
08:49.18gavimobilein the lan its 0
08:49.22gavimobilein the wan its 255
08:49.40Maliutaso the subnet mask on your lan is 255.255.255.0?
08:50.40Maliutaif the device is behind a working nat, forget the WAN side of things and look at the LAN side
08:51.41Maliutait is not unusual for ISP's to assign a /32 (which is a single IP address on a PTP link) to a device, then if you have any other static nets they route them down that
08:52.00Maliutaor you use NAT for all devices behind it
08:52.17gavimobileMaliuta: not sure what that all means
08:52.30Maliutagavimobile: so you don't know networking then?
08:52.55gavimobileMaliuta: not enough :-(
08:53.42Maliutaif you didn't understand that I'd say none, butI have fairly high standards for what constitutes "knowledge" :)
08:54.09Maliutathe device that is handing out the dhcp lease this device is getting. Is it a commodity router?
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08:55.12Maliutagavimobile: you're not assigning it some sort of odd static address? And you're configuring the device via a web interface?
08:55.54gavimobileMaliuta: no odd static address and configuring it through web interface. correct
08:58.25Maliutagavimobile: does the web interface have a network test function somewhere?
08:58.43gavimobileMaliuta: no network testing function
08:58.48Maliutamost of them do.
08:59.07Maliutanormally somewhere in the admin or status areas
08:59.23Maliutathat makes it harder to diagnose
08:59.50gavimobilethere is only 2 buttons
08:59.55gavimobilethe rest are fields
09:01.00Maliutagavimobile: what is the device type _exactly_ (i.e. down to hardware revision)?
09:01.52MaliutaI'm about to go out, but I'll take my laptop and I can do some looking while I'm out. If I come up with something I can just jump on IRC and tell you.
09:01.54gavimobilespa 2102. doesn't have a rev but it does say near the fcc logo 1.1.1
09:02.17gavimobileMaliuta: I appreciate your help! thanks man
09:02.54Maliutagavimobile: I try ... when I'm not being trying.
09:03.52Maliutagavimobile: tip. either stay in channel, or re-join when ever your infront of a machine. I'm always here, but not always in front of my machine
09:04.13gavimobileMaliuta: will do! thanks
09:06.14Maliutathere are a lot of helpful people around in here, we just don't always agree on how to track down a problem :) Or the best way to do a certain task.
09:06.45MaliutaOff to get my butt handed too me.
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09:26.10k3asd`hi
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10:40.50gavimobilemy main trunk incomming doesn't seem to be working properly.
10:41.21gavimobilerunning sip show peers says netvision-trunk-66         82.166.66.43       5060     UNREACHABLE netvision-trunk-67         82.166.67.43       5060     UNREACHABLE
10:43.48kaldemarmore network issues
10:44.16gavimobileI am getting the call now but no sound
10:45.28gavimobilekaldemar: before I upgraded the firmware an immeditly my pap device worked
10:45.52kaldemargavimobile: i don't understand what you mean.
10:46.39gavimobilekaldemar: my current problem is when people in the outside call they don't hear the ivr but my phones ring
10:47.26kaldemarsip debug will most likely tell the reason.
10:49.25gavimobilekaldemar:  here's my sip debug http://pastebin.com/aL7saHtf
10:51.00gavimobileit amazes me that only 1 of my istp's won't connect
10:51.04gavimobileand they say its not them
10:53.34kaldemardon't limit the debug
10:54.54kaldemarand naturally you need to look at a sip debug of a call.
10:56.11kaldemarif they don't answer to your OPTIONS messages (which are sent because of qualify=yes), don't send them. disably qualify for the peer. then you won't see UNREACHABLE in sip show peers.
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10:57.26gavimobilekaldemar: this is what I was able to log. http://pastebin.com/znEL4k0b
10:57.49gavimobilekaldemar: qualify=yes and they still show unreachable
10:59.19gavimobileon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
10:59.26gavimobileNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
10:59.28gavimobileline 129
10:59.28kaldemarre-read what i just wrote about the unreachable.
11:00.29gavimobileI disabled qualify and now they show as unmonitored
11:00.48gavimobileshould I make another test call?
11:01.00gavimobilewith qualify disabled?
11:01.03kaldemarno.
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11:01.33gavimobileis there a problem with codec or permissions?
11:01.42gavimobilecause last week I acidently deleted the sounds folder
11:01.59gavimobilebut all was working find till this morning
11:02.23kaldemardoes not seem to be. both have ulaw and asterisk says it plays the sound file.
11:02.39gavimobilekaldemar: in addition if I call the extention locally I hear sound
11:02.46gavimobilebut not from calling using my cellphone
11:03.25kaldemardo you see rtp going out with rtp debug?
11:04.18gavimobileno
11:04.36gavimobile<PROTECTED>
11:04.39gavimobilethen call
11:05.51gavimobilewhat am I looking for in the debug
11:05.58gavimobileits all the same two lines over and over again
11:07.29kaldemarand they are...
11:08.51gavimobilehttp://pastebin.com/ECydMgSZ
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11:13.09kaldemarlooks normal.
11:15.40gavimobilekaldemar: so is this a problem with my provider? if so how can I prove it and what do I tell them
11:17.29kaldemarnot necessarily. might be a firewall that does not let the audio go through or whatever.
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11:21.51gavimobilekaldemar: I am not sure about that.. why was until yesterday my trunks from this provider always listed as ok and not they are not
11:22.14gavimobilequalify was always set to yes. and the show peer results was ok
11:23.20gavimobilewhat's going on here
11:23.44kaldemaryour router is one point of failure. see that you don't have any ALG (application level gateway) for SIP enabled at all.
11:25.01gavimobileso I need to change my router?
11:25.10kaldemardid i say that?
11:25.22kaldemaryou need to make sure that it does not interfere.
11:25.52gavimobileim not ever sure where to start
11:26.15gavimobileopening dmz's for testing?
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11:28.34kaldemarverywiseman: don't target people with private messages like that.
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11:28.58verywisemankaldemar, why ?
11:29.17gavimobilemaking outgoing calls works find on both sides
11:30.10kaldemar~ask
11:30.10infobotQuestions 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.
11:30.35kaldemarverywiseman: and because i am not private consultant.
11:31.11wdoekesyou could be.. for the right amount ;)
11:31.33schmidtskaldemar you are NOT :(
11:33.13kaldemars/not/not your/
11:33.40kaldemarthat was what i really meant, a word was missing. :P
11:34.52wdoekess/your/your free/
11:35.20kaldemargetting better
11:36.24verywisemankaldemar, you must say that when i talk to you in private channel , it is not needing to tell me that in public :(
11:37.11wdoekesbut the ~ask doesn't work in private
11:38.22verywisemanok , it was enough to inform me that when i asked him in private channel , i was not knowing that
11:41.38wdoekesbut now you do; we won't hold it against you
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12:50.23GreenlightHi all. Just wondering if anyone has a UK ISDN30 (Virgin Media) setup and presenting outbound callerid. Virgin assure us that we've "type 5" CLI and we can present ANYTHING, but the only number that seems to present is the main bearer number (or withheld)?
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13:51.24BorjaGVOHi people. Something strange happened some days ago. I had in my full log the following message 245 times in minutes intervals: "NOTICE[25529] chan_sip.c: Sending fake auth rejection for device 1000<sip:1000@IP.IP.IP.IP>;tag=f2a400as". The wird thing is that the IP.IP.IP.IP is my server own IP address. It's a CentOS running just Asterisk, FreePBX and some other Asterisk-related tools. I don't understand how can this happen...
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13:52.12BorjaGVOI have a many operators calling and receiving calls through asterisk but none of them making calls from that box (IP.IP.IP.IP)
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13:57.03kaldemarBorjaGVO: that is not the address where the request is coming from.
14:03.23BorjaGVOwell, when I make calls without registering with domain with random extension, it logs the same message with IP address where request is coming from
14:03.32BorjaGVO(kaldemar)
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14:06.47kaldemarBorjaGVO: care to pastebin the output?
14:07.03BorjaGVOkaldemar: sure
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14:15.53BorjaGVOkaldemar: http://pastebin.com/Rjy44usg. I see now that the output from one case to the other vary. In one there is not port specification.
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14:18.54radenKatty, !!!! :)
14:19.21Kattyhi raden.
14:19.27radenHiya !!!!!!!!!!!
14:19.32radenI need a Katty hug
14:19.40Kattywhy's that?
14:20.22radenthe girl I was / am into has went from sweet , caring , awesome to total ice queen and like yea omg
14:20.47radencause some dude at work is filling her head with a bunch of shit
14:21.04Kattyi bet it's not like that.
14:21.14radenooooo tell me what its like
14:21.30Kattyi would imagine that the doubts have been there all along.
14:21.36BorjaGVOkaldemar: brb, going out to lunch (10 minutes)
14:21.46Kattyelse she'd have no reason to listen to him
14:22.11Kattythat or you've done something to make an ass of yourself.
14:22.37radennope she tells her friends how sweet i am and all this , even her sister is like she is so into you ....
14:22.55radenyet I have got stood up last 2 weekends
14:23.00radenand she always says how sweet i am
14:23.13radenmy head starting to hurt from this shit
14:23.27Kattyif you're not happy, move on.
14:23.36Kattydon't waste your energy.
14:23.39radenyea , thats what im doing
14:23.45Kattygood (=
14:23.48Kattyhugs raden
14:24.31radenI have no problem attracting women , not like im desperate , just shes awesome and fun and things were easy until dude at work started talking shit .....
14:25.05Kattydon't blame some guy.
14:25.11Kattyguys do what guys do.
14:25.34radenlol
14:25.45radenthat is true
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14:25.56radenbut at the same time shes " trusts his udgement "
14:26.06radenand the dude fucking hates me
14:26.13radenthere a lot to it ...
14:26.29radenyet she texts me every freaking day .......
14:26.54radenKatty, why cant these things ever just be like boy meets girl simple ?
14:27.09Kattybecause humans aren't simple.
14:27.23Kattyand having a heart broken makes the process even more complicated.
14:27.37Kattybecause then you have worry and fear running rampant on an otherwise perfectly healthy relationship.
14:27.48radenI showed interest to early and gave her to much control that was my bad , I know how to control women and I did not want to play games with her that was my stupid mistake ....
14:28.02Kattynot everyone can be controlled.
14:28.10radenI shouldnt say control
14:28.12Kattysome it just pisses off, and they will send you home talking to yourself.
14:28.12radenattract
14:28.46radenI dont try to control people , i just know the right things to do to attract them .... and I did none of that with her we just hit it off ....
14:28.57radenlol , send you home talking to yourself :) I like that
14:29.33radenI feel as though I have been put in the imaginary friend zone .... LOL
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14:29.36Kattyi don't think you should have to do anything to attract the right person. just be yourself. that should be enough to attract someone.
14:29.40radenwho knows I should move on
14:29.53Kattymaybe you have.
14:30.00Kattymaybe he is better suited to her than you are.
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14:30.09radenhes older than her by far
14:30.14Kattyso?
14:30.19radengood point
14:30.20Kattymaybe she wants someone to take care of her.
14:30.29Kattyi know someone like that....on both sides.
14:30.33radenI can take care of her way more than he can :P
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14:30.58Kattysome people like that. some people's inner jew bothers them if they don't hold up their half financially.
14:31.09Kattyall types of people in the world.
14:31.15radenI totally agreee
14:31.34radeni never put to much forward
14:32.18radenJust frustrating , 2 months invested in this BS
14:32.51radenknown her for 20 years
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14:33.00radengah im getting old LOL
14:34.22Katty2 months is nothin
14:34.29Kattylet's take this to private lol
14:34.30radenI know , I know
14:34.43Kattyno one wants to hear this conversation :P
14:35.06schmidtsno one else is saying something, maybe everybody only listens ;)
14:36.03kaldemarwas that about a service level agreement with a client?
14:36.48Kattytotally.
14:37.15kaldemarthe client did not want to pay for it.
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14:43.06pbxManhello
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14:54.21BorjaGVOanyone knows why this kind of messages happen to log? It's strange because the IP address is the server address. It's like the own server made calls. Actually, one of our operators recevied a call from this extension (1000) some hours after. No one was on the other side (no audio stream): http://pastebin.com/Rjy44usg
14:54.47Kattyinfobot: crittercam
14:54.47infobot[crittercam] Katty's Critter Cam http://tinyurl.com/b5k3lt4
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15:27.47wdoekesBorjaGVO: the log message is confusing. it shows the To: field in the header, which usually contains the server address, not the IP you're interested in
15:29.04BorjaGVOHow do you know is the To: field?
15:29.11BorjaGVO(wdoekes)
15:34.57fullstopI'm watching a squirrel eat birdseed.
15:35.15Kattyand peanuts.
15:35.49Kattytho i guess you're right. he's mostly going for the sunflowers.
15:35.58fullstopnice snowman
15:36.04Kattythanks
15:36.07Kattyhttps://sphotos-a.xx.fbcdn.net/hphotos-prn1/154579_10100452825052137_2118825807_n.jpg <- snowman
15:36.20fullstopI'll be honest, though.  It has seen better days.
15:36.32Kattyyesh
15:36.53fullstopalmost looks like a snow-cat.
15:37.10Kattyhe was made to be put on reddit.
15:37.16fullstopI see
15:37.17carrarHELLO SNOW KITTY
15:37.22Kattyhi carrar
15:37.34carrarHi Katty!
15:38.03Kattyhugs carrar
15:38.09carrarWoo Woo!!
15:38.16Kattycarrar: i got my stegosaurus done. i named him Steve, the chronically depressed Rainbow Stegosaurus
15:38.18carrarspins Katty around
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15:38.37carrarheh
15:38.41Kattycarrar: http://42ndknitstreet.blogspot.com/2012/11/stegasaurus.html <- steve.
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15:38.51carrar<PROTECTED>
15:39.04Kattyhe's already built!
15:39.19carrarWoah that looks danergously green
15:39.21Katty(scroll to the bottom to see friends)
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15:39.34wdoekesBorjaGVO: enable: sip set debug on
15:39.48carrardinasur and bunnies
15:40.02wdoekesthen you'll get the entire sip mesage, and you'll see the source IP that's causing the "Fake auth reject" log messages
15:40.06KattyBun, Ham, and Doc Oct
15:40.16Kattythey were trying to console steve.
15:40.21carrarheh
15:40.48BorjaGVOwdoekes: I cannot do that since I don't know when the attepmt is going to happen. It only happened once...
15:41.00Kattynow i'm finishign some garland for the xmas tree i already took down. heh. what are you up to carrar?
15:41.23carrarI just woke up!
15:41.41Kattysounds like it's time for Red Bull in the shower.
15:42.08carrarI have to still make my espresso!
15:42.28Kattydid i tell you my gentlemen friend bought me an espresso machine for xmas?
15:42.39carrargentlemen friend!!
15:42.42carrarPICS
15:42.46carrarof the espresso machine
15:42.57Kattyhttp://www.amazon.com/DeLonghi-EC155-Espresso-Cappuccino-Maker/dp/B000F49XXG/ref=sr_1_1?ie=UTF8&qid=1357659756&sr=8-1&keywords=espresso <- that one
15:43.01carrarI have pics of mine too!
15:43.16carrarOh nice
15:43.17Kattyit's a dream.
15:43.28carrarEspresso Machines really are a dream come true
15:43.33Kattyi got him a burr grinder and an electric tea kettle for xmas.
15:43.42Kattyhe already had a french press
15:43.48Kattyso now he have a coffee bar.
15:43.49fullstopbecause of wifestop's occupation, we have random stuff sent to our house all the time.
15:43.54Kattyfinds photo of coffee bar
15:44.09fullstopApparently there is a blendtec blender on the way... and we get to keep it!
15:44.17carrarMine: http://pics.osburn.com/album/1230
15:44.32carrarburr grinders are the best!
15:44.38*** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net)
15:44.40Kattyhttps://sphotos-b.xx.fbcdn.net/hphotos-prn1/68603_10100447794433537_1245549956_n.jpg <- coffee bar
15:44.57Kattythere's a coffee pot too, but it wouldn't fit
15:45.00carrarholy cafe!
15:45.10Kattywe take our coffee very seriously.
15:45.14carrarIt's like a cafeteria
15:45.18Kattyyesh.
15:45.35carrarWere is the food dispensing machine!
15:45.52Kattybehind the photographer
15:45.54Kattyin the fridge hehe
15:45.58carrarheh
15:46.28Kattythe local grocery store puts their 'old' meat on sale on tuesdays to clear it out
15:46.44Kattythey've been having lots of steak recently.
15:47.01Kattyso the Food Despensing Machine has lots of frozen steaks to pick from
15:47.25Kattylast week they had a package of ribeyes for 6 bucks. 2 of them.
15:47.27carrarheh
15:48.46carrarIt's too danergous to eat meat anymore
15:49.00carrardangerous
15:49.03Kattycause of all the stuff theyput in it?
15:49.07carraryeah
15:49.09Kattywell. into the cows. and chickens. and what not
15:49.15carraror things they didn;t know got in it
15:49.19Kattyyes :<
15:49.21carrarlike crazy cow
15:49.22BorjaGVOwdoekes: I didn't know that letting enabled sip debugging got it on logs. Thanks.
15:49.30Kattycarrar: i've had to switch to organic eggs.
15:49.37Kattycarrar: something about the regular ones makes me nauseous.
15:49.54Kattycarrar: dunno what. my guess is the chemicals in the chicken feed.
15:50.08carrarbiochemically engineered eggs!
15:50.30carraror something like that
15:50.32carrarheh
15:50.41Kattywell they have special feed for chickens.
15:50.47Kattyit has growth hormones and antibiotics in it
15:51.03carrarAll makes for a better super bug!
15:51.07Kattythere's a documentary on netflix where they talk about Tyson chicken.
15:51.45Kattyi should just get a few chickens.
15:51.51Kattyvery spoiled fat little chickens.
15:51.58carraryeah
15:51.58fullstopWe used to get free-range chicken eggs from a friend that had a farm.
15:52.19Kattyfullstop: those are the best.
15:52.19fullstopThe shells were crazy thick and the yolk was bright orange.
15:52.20carrarmake sure to wash them!
15:52.25fullstopNaturally
15:52.33Kattyfullstop: my parents used to have chickens.
15:52.39fullstopThey were awesome.  But then they moved.
15:52.43Kattythey had a big yard too...and just meandered about looking for bugs.
15:53.05fullstopSome other friends had two chickens: Original Recipe and Extra Crispy.
15:53.25Kattyawww now that would be a wonderful name for a couple of hens.
15:53.27KattyOriginal and Crispy
15:53.29carrarYou could raise angus cows too!
15:53.31fullstopOne was eaten by a fox, and the other wasn't much for laying eggs.  They'd get one or two a week.
15:53.35*** join/#asterisk MarKsaitis (~MarKsaiti@81.101.81.114)
15:53.39carrarfeed them grain
15:53.41Kattycarrar: for cheese and milks?
15:53.48carrarfor dah beef!
15:53.48Kattycarrar: why not grasses
15:53.54Kattyoh i wouldn't have the heart to kill it
15:54.02fullstopThat's why you take it to a butcher.
15:54.02carrargrain feed cows the meat taste better
15:54.06fullstopno sir
15:54.09fullstopgrass all the way
15:54.12Kattyi disagree.
15:54.16Kattyi'm a fan of grass fed beefs.
15:54.28carrarwe've done it
15:54.31carrarfor several years
15:54.50carrarwe've (as in my parents)
15:54.51Kattyguess we have different tastes then (=
15:54.58carraryeah
15:55.03coppiceKatty how do you feed it once it has become beef? :-\
15:55.05Kattymore for me. more for you!
15:55.30carrarcoppice, feed it buy opening wide and inserting said meat into mouth!
15:55.39dr0ckthe way you feed it when its still a cow. cut a hole in the side and pour feed inside
15:55.40Kattyyes. then proceed to omnomnom.
15:55.43carrarhaha
15:55.48*** join/#asterisk wonderworld (~w@dsdf-4db541ae.pool.mediaWays.net)
15:55.56Kattylet's not talk about cows.
15:56.02Kattyi feel bad for them.
15:56.03carrarmoo
15:56.15fullstopಠ_ಠ
15:56.24carrarThere is always kale
15:56.38Kattymmm kale chips.
15:56.44Kattysprinkled with sea salt
15:56.44coppiceKatty: do you feel sorry for the ones fed on beer, and massaged every day?
15:56.52fullstopI have two raised garden beds now, 4x8 each.
15:57.00fullstopHopefully expanding to 4 next year.
15:57.02Kattyfullstop: i've been thinking about doing a raised garden
15:57.08Kattyfullstop: what do you grow?
15:57.12fullstopWhere are you, geographically?
15:57.19Kattysouthern missouri.
15:57.24fullstopI made mine from cedar, but it is kind of hard to find here.
15:57.32fullstopeast coast
15:57.33Kattyoh well. i can't do that anyway.
15:57.36Kattyi'm allergic to cedar.
15:57.40fullstop!!
15:57.43coppicemy wife's cousin has a raised garden. it covers the roof of a 30 floor tower
15:57.54Kattycoppice: wow nice.
15:58.05fullstopHopefully they have a long hose.
15:58.18coppicethey grow fruit and veg, and raise chickens
15:58.35Kattychickens? on a roof?
15:58.40Kattyseems dangerious.
15:58.49coppicewhy?
15:59.05Kattyi'm afraid the chickens would fly off. and crash land somewhere...30 stories down.
16:00.14coppicethey have an avery sized cage for the chickens, although chickens aren't major flyers anyway
16:00.31Kattyah. well at least they're caged.
16:00.43Kattyand no, they're not major flyers. but they're not exactly graceful when they try either
16:02.50coppiceits kinda weird standing in a market garden, looking down at the streets of GuangZhou 30 floors below
16:03.13navaismoBorjaGVO: enable the sip debug then you can check it in the full log(enable the full log too) and secure your asterisk, allowguest=no, dont use weak passwords, close ports if you dont use it
16:03.15Kattyi imagine.
16:03.27Kattyi love the idea that someone is doing it tho.
16:03.32Kattyi'd gladly buy from them.
16:04.35fullstopcoppice: can you read Cantonese?
16:05.00coppice是的
16:05.51fullstopI'll have to see if I still have it, but my wife received a camera bag for christmas with a chinese newspaper stuffed inside.  There's a comic in it that I was wondering if it would be funny when translated.
16:06.19fullstophopefully she did not pitch it
16:08.43*** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212)
16:21.14Kobazanyone have a problem where you have a loaded system and completion of an attended transfer is very delayed
16:21.57Kobazie: A calls B, B calls C... B hits transfer in order to connect A and C together.  And it takes 5-10 seconds for the audio stream to start from A to C
16:27.11*** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212)
16:27.57carrarok, getting properly caffinated
16:28.17mjordanKobaz: does the log show RTP in probation?
16:30.54Kobazmm, what would probation look like?
16:31.06Kobazi dont like any rtp, but i can log it for an experimental session
16:31.11Kobazs/like/log/
16:33.22mjordanKobaz: it should be just a normal debug message - you don't have to have rtp debug on
16:33.46Kobazah
16:33.50Kobazlemme search
16:34.20mjordanstrictrtp requires a sequential number of RTP packets to be received from a particular source before it trusts that it should send RTP to that source. Basically prevents someone from sneaking into the audio stream and snagging it away from an endpoint.
16:34.44Kobazi dont have strictrpt set
16:34.58Kobazno logs matching *prob* for today
16:35.16Kobazor yesterday
16:36.37*** part/#asterisk rkeene (1011@oc9.org)
16:38.40*** join/#asterisk spriggan (~ircap@excsupercol.supercable.net.co)
16:39.58Kobazthis is on 1.8
16:40.13sprigganhi every one ,, some one can help me ? im using the manager api to get the QueueStatusComplete info.. i need to know if the wait time is the time since the call gets into the queue or if it's since the call gets into the ivr
16:43.46mjordanKobaz: not sure then. I haven't heard of that happening.
16:44.57*** join/#asterisk shadar (~eugene@37.113.202.81)
16:55.29*** join/#asterisk Mon|A|rch (~SBean@72.29.180.35)
16:56.09Mon|A|rchhey
16:56.15Mon|A|rchanyone around that can help me with some issues?
16:56.36WIMPy~ask
16:56.36infobotQuestions 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.
16:57.30Mon|A|rchlol, fair enough
16:58.24Mon|A|rchso, I'm setting up an IVR, and within the local network, I can call the asterisk server's extension (502) and it'll route me through wherever I want to go
16:58.45Mon|A|rchthe issue I'm having is that I can't originate calls with the ami, because my sip channel status is 4
16:58.50Mon|A|rchor unavailable i suppose
16:59.02Mon|A|rchsip show peers shows me my sip channels
16:59.05Mon|A|rchsip show registry does not
16:59.30Mon|A|rchI'm taking over someone elses project, so their code is a little wonky in extensions.conf
16:59.39Mon|A|rchI'm thinking I might need to just go ahead and replace the whole file
16:59.39Kobazsip show registry will show you registrations that asterisk is set up to make with other devices
16:59.50Mon|A|rchokay
17:00.27Mon|A|rchI was a little confused on the difference
17:00.29Mon|A|rchanyway
17:00.38Mon|A|rchI think I set up my incoming trunk incorrectly
17:00.38Kobazlike, say an itsp requires registration, you'll add the registration in your configs and then it'll show up in sip show registry
17:01.02Mon|A|rchokay
17:01.35Kobazsip show peers will show configured sip peer devices and their status
17:01.53Kobaznot necessarily all available sip channels because there is sip show users as well
17:02.42Mon|A|rchi see
17:02.49Mon|A|rchso I've got one more user than i have peers
17:03.04Mon|A|rchthe extensions i want to access are listed in sip show users/peers
17:03.05Kobazif you do type=friend in the sip.conf then it will add a peer, and a user
17:03.14Mon|A|rchk
17:03.14Kobazif you do type=user, then it's just a user
17:03.24Mon|A|rchI've been using type=friend
17:03.45Kobazthere's generally very little need to specifically do type=user
17:04.21Mon|A|rchi would assume so
17:04.45Mon|A|rchso, I'm assuming I need to set up a user, or a sip channel that represents my incoming/outgoing trunks?
17:04.48Kobazyou can think of users as only inbound
17:04.56Mon|A|rchokay
17:04.58*** join/#asterisk ujjain (ujjain@unaffiliated/ujjain)
17:05.09Kobazfriend as inbound/outbound
17:05.43*** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212)
17:07.34Kobazand it's better to refer to your setup as sip peers and users, not really channels
17:07.41Kobazbecause channels have a specific meaning
17:07.50Kobazit's more like available to use channels, but not really
17:08.06Kobaza channel is a leg of a call, (a call in progress)
17:08.17Kobazsip show peers, is more like, showing you available devices
17:10.40Mon|A|rchso, are there any configs, aside from sip.conf and extensions.conf that actually determine whether or not a sip friend can be accessed?
17:10.57*** join/#asterisk retentiveboy (~retentive@74-95-28-34-Atlanta.hfc.comcastbusiness.net)
17:11.26Mon|A|rchI've set the default context for my trunk and the friends associated with the extensions I'm using to the context that the extensions are in
17:11.48Mon|A|rchdo i need to declare my trunk within sip.conf?
17:20.33*** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com)
17:24.34Mon|A|rchfrom what i can tell, I need to register my sip provider (which is a cisco server we're using) in sip.conf?
17:29.10*** join/#asterisk qakhan (~qakhan@208.253.91.58)
17:29.17qakhanhi all
17:29.33qakhananyone has polycom .cfg files
17:29.44qakhantamplates
17:37.19Mon|A|rch*sigh* so much lurking
17:37.27Mon|A|rchwell, if anyone has a suggestion, let me know
17:37.48Mon|A|rchI'm just unsure how to properly set up outbound and inbound
17:43.53kaldemarMon|A|rch: you need to edit both sip.conf and extensions.conf for inbound and outbound: http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/OutsideConnectivity_id291235.html#OutsideConnectivity_id291268
17:45.05Mon|A|rchkaldemar, thanks
17:45.07Mon|A|rchi appreciate it
17:48.09*** join/#asterisk Dibbler (~Dibbler@host109-148-34-244.range109-148.btcentralplus.com)
17:48.25Mon|A|rchkaldemar, so, if i were to set a friend to host=dynamic, it would receive all calls?
17:49.32*** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey)
17:50.26*** join/#asterisk citrusfizz (~chatzilla@70.184.40.66)
17:51.11citrusfizzI setup a conference room on Asterisknow,  with out fail, everyone gets kicked off after 30 minutes exactly.   what could be the cause of this?
17:51.42Qwella timer on the room?
17:55.03kaldemarMon|A|rch: setting something host=dynamic has little to do with receiving calls. it only tells asterisk that the device should register to it. after registering, asterisk knows an ip address and a port to reach the device.
17:55.32Mon|A|rchokay
17:56.09Mon|A|rchdo i need to actually use users.conf?
17:57.29Mon|A|rchsorry about my confusion, I've had to figure this all out pretty quickly
17:58.51*** join/#asterisk JustinAiken (~JustinAik@justinaiken.com)
17:59.18JustinAikenHi all, having an issue with DTMF detection on ast 1.8 i was looking for some help on
17:59.31JustinAikensometimes a digit is doubled; pressing it once receives it twice
17:59.42JustinAikenif i look at the dtmf logging, i see stuff like:
17:59.52JustinAiken'9' has duration 39 but want minimum 80, emulating on SIP/vitel-inbound2-0000014a
18:00.06JustinAikeni do i stop it from emulating if it's below the minimum?
18:00.21*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
18:01.05[TK]D-Fender<Mon|A|rch> do i need to actually use users.conf? <- no
18:01.39[TK]D-FenderMon|A|rch, Only AsteriskGUI really needs it.  Aside from that I wouldn't touch it ever.
18:02.02Mon|A|rchalright
18:02.58Mon|A|rchso, i want to originate calls to outside numbers, which are passed in the originate command from some php code, to an extension in my incoming context
18:03.18[TK]D-Fender<qakhan> anyone has polycom .cfg files <---- polycom.com.  It comes with the firmware
18:03.36[TK]D-FenderMon|A|rch, cool.
18:03.46Mon|A|rchdo i need to set up a SIP friend for that extension specifically? i figure i need to set up a trunk for my provider
18:04.26[TK]D-FenderMon|A|rch, for the actuall calls.. yes
18:04.59*** join/#asterisk retentiveboy (~retentive@74-95-28-34-Atlanta.hfc.comcastbusiness.net)
18:05.28[TK]D-FenderMon|A|rch, What "extension"?
18:05.46[TK]D-FenderMon|A|rch, you set up SIP entires to devices and providers you need to call/get calls from
18:06.16Mon|A|rchokay
18:06.52*** join/#asterisk slidesinger (~slidesing@c-69-141-208-250.hsd1.nj.comcast.net)
18:07.13Mon|A|rchso what i need to do is set up a single sip entry for my provider server, to route calls to my incoming extension
18:07.36[TK]D-FenderWhat is this "incoming extension" you are referring to?
18:08.54Mon|A|rchsorry, incoming is just a context in extensions.conf, and there's an extension in there that is the IVR
18:09.51[TK]D-FenderMon|A|rch, then that is just dialplan.... so just the SIP entry to match the incoming call.
18:12.02Mon|A|rchalright
18:13.31Mon|A|rchwill i need to change the entry for outgoing? will type=friend cover that?
18:19.58gustoso
18:20.12Mon|A|rchI'm getting a -1 status when i use extensionstate
18:20.19Mon|A|rchdoes that generally mean it can't find the extension?
18:20.30gustono idea
18:20.36gustowhat extension?
18:20.41gustoare you in right context
18:20.43gustoha?
18:21.05[TK]D-Fender~pb
18:21.05infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
18:21.07[TK]D-FenderMon|A|rch, ^^^
18:21.21[TK]D-FenderShow us precisely what you're doing and what's happening...
18:21.49Mon|A|rchalright
18:27.32*** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net)
18:27.59Mon|A|rch[TK]D-Fender, http://pastebin.com/nZhaSCqg
18:28.25Mon|A|rchI'll paste the originate command in a sec
18:28.36*** join/#asterisk Defraz (~Defraz@168.103.142.217)
18:28.43[TK]D-FenderMon|A|rch, exten => 502,1, goto(incoming,500,1) <- you have no 500,1
18:28.57[TK]D-FenderMon|A|rch, You started with "n" and have no step 1
18:29.25[TK]D-Fenderexten => 500,n, Goto(diab-exh-are-you,s,1) <- I'd avoid the space there too...
18:29.49Mon|A|rchoh, there was a 1 priority, it just wasn't necessary
18:29.51Mon|A|rchdeleted it for the paste
18:29.52Mon|A|rchsorry
18:30.13Mon|A|rchbeen amended
18:30.23Mon|A|rchI've tested that extension by dialing internally
18:30.28Mon|A|rchand it works pretty much as advertised
18:30.29[TK]D-FenderMon|A|rch, Show a new PB with the relevant bits as well as precisely what you're checking for and status dumps to match
18:30.32Mon|A|rchI'll avoid the spacing
18:30.41[TK]D-Fenderwe need to see the error
18:31.03Mon|A|rchalright
18:31.28Mon|A|rchalso, i don't know what a PB is
18:32.17[TK]D-FenderPASTEBIN
18:32.30Mon|A|rchsorry
18:34.30*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
18:34.56*** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e)
18:35.21*** join/#asterisk justdave (~dave@unaffiliated/justdave)
18:36.28*** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com)
18:36.55Mon|A|rchhttp://pastebin.com/e6TbwnRK
18:37.28Mon|A|rch[TK]D-Fender, let me know if that's not enough information
18:38.59jacekowskiehh, i've got a sip provider that uses multiple ip addresses for incoming calls
18:39.12jacekowskias in, i may register with .65 and then have stuff coming from .66
18:39.14[TK]D-FenderMon|A|rch, What's the actual goal?
18:39.35jacekowskiand asterisk is rejecting packets from that provider
18:40.11qakhan[TK]D-Fender web.cfg and license.cfg is not included in their package
18:40.57*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
18:41.26[TK]D-Fenderqakhan, Who said it was supposed to?
18:41.30Mon|A|rchthe eventual goal is to originate a call from asterisk to a phone outside the network, when they pickup, they get led through the IVR, and enter answers to questions n' things
18:41.37[TK]D-Fenderqakhan, How do I know which one you even grabbed?
18:41.45Mon|A|rchi can dial into the extension from outside the network successfully
18:41.56[TK]D-FenderMon|A|rch, Ok, and the "extension_state" bit
18:41.57[TK]D-Fender?
18:42.13Mon|A|rchmaking sure the channel isn't congested or unavailable
18:43.10qakhan[TK]D-Fender do you have?
18:43.32Mon|A|rcham I being dumb here?
18:43.32[TK]D-Fenderqakhan, You are not being informative at all....
18:43.46[TK]D-FenderMon|A|rch, It's always available ... it's just dialplan....
18:43.58[TK]D-FenderMon|A|rch, You can 1000 calls there if you want as far as I can see...
18:44.04[TK]D-FenderMon|A|rch, What do you see that should limit you?
18:45.06[TK]D-Fenderjacekowski, Add another peer or match by name for type=user
18:45.14Mon|A|rchit's only asterisk informing me that the extension is unavailable
18:45.23Mon|A|rchI'll remove that check, and see if the calls work
18:45.38Kobazdo de do... so i'm back
18:46.42Kobazconfiguring service pack, do not turn off your computer
18:49.21Mon|A|rch[TK]D-Fender
18:49.35Mon|A|rchhad some errors, will pastebin
18:53.16Mon|A|rchhttp://pastebin.com/rsevQ8jr
18:53.33Mon|A|rchthe calls are falling through
18:54.43[TK]D-FenderMon|A|rch, and everything around it please... I see a call landing on s@default... not the "500" you showed earlier
18:54.49*** join/#asterisk camelCase (~camelCase@unaffiliated/camelcase)
18:54.56Mon|A|rchoh, sorry
18:55.38camelCaseanyone see this: http://blog.exodusintel.com/2013/01/07/who-was-phone/
18:56.41QwellcamelCase: It was sent to the asterisk-announce mailing list, so yes, everybody using Asterisk saw it.  Unless they're bad and not subscribed to that list.
18:56.45Mon|A|rchhttp://pastebin.com/sSJxR9Ww
18:56.49Mon|A|rch[TK]D-Fender, there you go
18:56.52qakhan[TK]D-Fender i downloaded UC_Software_4_0_3F_release_sig_split from Polycom
18:56.56Mon|A|rchi appreciate your patience
18:57.11Mon|A|rchi realize this is probably nerve-grinding
18:57.45qakhanthen i made copy of phone1.cfg from internet
18:57.50[TK]D-FenderMon|A|rch, Only for you... np here...
18:58.01Mon|A|rchfair enough
18:58.06Mon|A|rchthanks anyway though
18:58.11[TK]D-Fender== Starting SIP/10.3.1.1-00000000 at default,,1 failed so falling back to exte                                                                             n 's'
18:58.19[TK]D-Fenderthat's clearly not good...
18:58.25[TK]D-FenderI need to see your originate, etc...
18:58.33Mon|A|rchokay
18:58.45[TK]D-FenderALL the backup for what started this call and everything you think it should be doing...
18:59.00qakhannot when i upload to phone this require MAC-web.cfg and MAC-license.cfg
19:01.04*** join/#asterisk pa (~pa@unaffiliated/pa)
19:05.08*** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e)
19:10.43Mon|A|rch[TK]D-Fender, http://pastebin.com/pkUjrF2s
19:10.50sprigganhi every one ,, some one can help me ? im using the manager api to get the QueueStatusComplete info.. i need to know if the wait time is the time since the call gets into the queue or if it's since the call gets into the ivr
19:11.06Mon|A|rchI can paste the entire method, but that's the meat of it
19:11.24Mon|A|rchlater on it'll take different extensions and contexts for different functionality
19:11.30Mon|A|rchbut i need to get originating working first
19:12.12Mon|A|rchoh
19:12.12Mon|A|rchwhoa
19:12.14Mon|A|rchcaptcha
19:12.17Mon|A|rchwasn't paying attention
19:12.18Mon|A|rchsorry
19:12.30Mon|A|rchhttp://pastebin.com/pkUjrF2s
19:12.31Mon|A|rchthere you go
19:13.55Mon|A|rchusers.conf was modified by the guy who started the project, i can rename the file for now if you think it's messing with the sip config
19:14.54[TK]D-FenderMon|A|rch, Your original dialplan showed me extension 500
19:15.03[TK]D-Fenderttp://192.168.3.103:8088/asterisk/rawman?action=originate&channel=sip/10.3.1.1/918059941511&extension=500&context=default&priority=1
19:15.06*** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk)
19:15.19[TK]D-FenderMon|A|rch, And this isn't pointing to the same context at all
19:15.33[TK]D-FenderMon|A|rch, "default" != "incoming"
19:15.48[TK]D-FenderMon|A|rch, Which is probably the mistake right of the bat
19:15.56Mon|A|rchfacepalms to death
19:17.14Mon|A|rchhm
19:17.18Mon|A|rchstill getting the same error
19:17.19Mon|A|rchhold on
19:17.24*** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius)
19:17.51fullstopMon|A|rch: This is your next option: http://i.imgur.com/3L3r2.gif
19:18.57danfromukHi, Ive got a client that received fax calls over ISDN (ports 1 and 2 on an ISDN card) and routes them out of port 4 to a fax server running on a windows machine. Its not working. How can I go about diagnosing it? Any tips? I've only ever dealt with SIP.
19:19.10danfromukNormal calls are coming in fine.
19:19.44Mon|A|rchit's still trying to work with the default context ><
19:19.56[TK]D-FenderPB
19:20.22Mon|A|rchk
19:20.58*** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e)
19:22.40*** join/#asterisk lifeforms (~walter@tau.lfms.nl)
19:22.49Mon|A|rchhttp://pastebin.com/QFLxLZ22
19:22.53Mon|A|rch[TK]D-Fender, k
19:23.18[TK]D-Fenderqakhan, It doesn't come with the provisioning.  It comes when you LICENSE the phones.
19:23.42Mon|A|rchit's like it doesn't see my extension
19:23.50Mon|A|rch"failed at incoming,,1"
19:24.06[TK]D-Fender<PROTECTED>
19:24.13[TK]D-Fendernow no EXTENSION
19:24.28Mon|A|rchhm
19:24.35[TK]D-Fenderincoming (comma) (comma) 1
19:24.44[TK]D-FenderMon|A|rch, Watch for more typos
19:25.06Mon|A|rchit all looks good, vardumping, hold on
19:25.23[TK]D-FenderMon|A|rch, I see no reason for it to be variable so far at all...
19:25.30[TK]D-FenderMon|A|rch, You have a fixed target it seems...
19:26.06*** join/#asterisk Russ (~russ@md20536d0.tmodns.net)
19:26.15Mon|A|rchit'll be variable later
19:26.22Mon|A|rchbut for now I've set it static at 500
19:27.01[TK]D-FenderApparently you "oopsed" in there somewhere...
19:27.33Mon|A|rchapparently
19:27.35lifeformsI'm having trouble with iax.conf, a new section seems to 'inherit' settings from my VOIP provider.. I have a [providername] sort of 'template' as received from my VOIP provider, I'm using it for redundancy as in [prov01](providername) and [prov02](providername)... now my new [office] section doesn't work because when the office connects, I am getting: 'chan_iax2.c:11015 socket_process: Host 192.0.2.1 failed to authenticate as providername' even though that
19:27.53*** join/#asterisk Docfxit (~Docfxit@netblock-75-79-6-10.dslextreme.com)
19:27.58Qwelllifeforms: pastebin your config
19:28.07[TK]D-Fender~pb
19:28.07infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
19:28.11[TK]D-Fenderlifeforms, ^^^
19:28.13lifeformsok hold on :)
19:28.44Mon|A|rch[TK]D-Fender, http://pastebin.com/hgWgJvj7
19:28.49Mon|A|rchthat's all i have for the originate string
19:28.56Mon|A|rchafter that it's all CURL
19:29.45[TK]D-FenderMon|A|rch, more complete code, more complete call please.
19:29.54Mon|A|rchalright
19:30.27lifeformshttp://pastebin.com/urcUyadL there goes!
19:31.19lifeformsso my problem is, 'office' connects great but my home asterisk says: Host <ip> failed to authenticate as speakup-lifeforms
19:31.47lifeformsand I am a bit puzzled why it would even expect that :) but I don't know that "(speakup-lifeforms)" notation at all, my iax provider gave it to me, and it's very hard to google on ( ) :)
19:31.51Qwelllifeforms: (!) makes something a template
19:32.03Qwell[speakup-lifeforms](!)
19:32.08lifeforms(!)
19:32.14lifeformsaha...
19:32.18QwellI doubt it'll actually fix your problem, but it's b0rked
19:32.29Mon|A|rchhttp://pastebin.com/3ijP9si6
19:32.33FLeiXiuSWill events after my dialplan hangup occur?
19:32.43lifeformslet's see what happens if I make that a proper template then
19:32.58Mon|A|rch[TK]D-Fender, I'm sure this is going to be some horrible typo somewhere ;_;
19:33.49lifeformsokay, something did change :)
19:33.55[TK]D-FenderMon|A|rch, extension=500 <--- IIRC this is supposed to be "exten", not "extension"
19:34.09Mon|A|rchlol
19:34.20*** join/#asterisk elico (~Thunderbi@109.67.209.186)
19:34.29Mon|A|rchrofl
19:34.34lifeformsI did [speakup-lifeforms](!) now, this seems better. but now, when 'office' tries to connect, it says: Host <ip> failed to authenticate as speakup01 :')
19:34.48lifeformsthe sarcastic smiley was not part of the log message
19:35.06lifeformsso [office] seems to reuse some part of the earlier section
19:35.35Mon|A|rch[TK]D-Fender, it all works no
19:35.48[TK]D-FenderMon|A|rch, Please go stand on that large plastic sheet in the corner while I grab my pistol.....
19:35.50[TK]D-Fender:p
19:36.02Mon|A|rchrofl
19:36.12Mon|A|rchseriously
19:36.13[TK]D-Fenderhated the cleaning bill from the last time he forgot the sheet
19:36.20[TK]D-FenderMon|A|rch, You're welcome.
19:36.45Qwelllifeforms: should office maybe be a peer, and not a user?
19:37.28lifeformsQwell: could be nice in the future, but for now, I just want office to reach me at home
19:38.12DocfxitI would like to record a voice prompt.  I added extension 130 to extensions.conf. The lines I added are at pastebin.com/t3Yqgqzp  I have restarted Asterisk.  It isn't working.  Could someone please suggest what I need to do to get it working?
19:38.43DocfxitIt isn't recording the file.
19:39.01citrusfizzQwell: you asked if i had a timer on the room, but i don't see any place to set that in the asterisknow web interface
19:40.07navaismoDocfxit: did you mis the SET in the pastebin?
19:40.13navaismoor is actuallñy your diaplan
19:41.02Docfxitnavaismo: on the recordedfile line?
19:41.07navaismoyes
19:41.22[TK]D-FenderDocfxit, RecordedFile=/tmp/asterisk-recording <---- this is not a dialplan extension calling Set() ......
19:41.43lifeformsI could get rid of the templates altogether, I guess, let's try that first
19:42.55ghost75is it not possible to hold call on specific channels over ami ?
19:43.39ghost75found only mute
19:44.23DocfxitDoes this look better? pastebin.com/7rJ4fkKG
19:45.13navaismosame => n,Set(RecordedFile=/tmp/asterisk-recording)
19:45.17[TK]D-FenderDocfxit, NO
19:45.33[TK]D-FenderDocfxit, EXTEN =>  ............
19:45.59[TK]D-FenderDocfxit, It is a dialplan app.  It requires all the same basic formatting as every other line of dialplan
19:49.49lifeformsokay, removing the templates didn't help, still my new iax2 connection 'failed to authenticate as speakup01' (the topmost section).. but if I remove [speakup01] and [speakup02] sections from iax.conf, office can connect!
19:49.57lifeformsvery interesting
19:56.28*** join/#asterisk Penguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
19:58.09Qwelllifeforms: should office maybe be a peer, and not a user?
20:01.31qakhan[TK]D-Fender can you tell me how to setup auto answer on exts
20:01.52lifeformsQwell: if I remember correctly, the receiving machine would call the caller a 'user', right?
20:02.03lifeformsbut let's try something else, yeah
20:02.03PenguinYou don't want auto-answer on extensions.
20:02.05PenguinYou want auto-answer on PHONES.
20:02.29PenguinThe PHONE is the important part.
20:02.31[TK]D-Fenderqakhan, http://www.voip-info.org/wiki/view/Polycom+auto-answer+config
20:02.43[TK]D-Fenderqakhan, for POLYCOM's only
20:02.46FLeiXiuSWhat would be the cause of my extension going from a Hangup to the beginning of an extension?  It goes into the extension name H then back into S after it hangs up.
20:03.03[TK]D-Fenderqakhan, Every phone maker that even offers the feature does it differently.
20:03.19Penguinfleixius: What does h have in it?  What else is in the context?
20:03.20[TK]D-FenderFLeiXiuS, Show us
20:03.38PenguinExtension patterns like _. are BAD.
20:05.04FLeiXiuS[TK]D-Fender, Penguin http://pastie.org/5650059#18
20:05.27[TK]D-FenderFLeiXiuS, and the call.......
20:06.45FLeiXiuS[TK]D-Fender, http://pastie.org/5650068
20:07.05[TK]D-FenderFLeiXiuS,  the COMPLETE call
20:07.18FLeiXiuSBah!  Woops
20:08.48PenguinI'd also like to know the context of those extensions that you showed.
20:10.32FLeiXiuS[TK]D-Fender, http://pastie.org/5650092
20:13.55[TK]D-FenderFLeiXiuS, Apparently calling "hangup" in "h" = bad
20:14.50FLeiXiuS[TK]D-Fender, That what I was going too..but how else would I execute a quit..or a stop?  Not putting anything at all.
20:14.55FLeiXiuS? *
20:15.15[TK]D-FenderFLeiXiuS, Run out .
20:15.16*** join/#asterisk timahvo1 (~rogue@41.212.120.45)
20:15.26lifeformsQwell: okay, apparently the problem goes away when each section has a 'username='... I am guessing when it's not set, asterisk (incorrectly?!) uses the username= setting of a former iax.conf section... pretty strange!
20:15.29FLeiXiuSPenguin, The context of this call is tricky ;-P
20:15.44PenguinNo it isn't.
20:15.59FLeiXiuSPenguin, It's doing some silly things, whcih makes it unlike a phone call
20:16.25PenguinThe call can only have one context per extension involved.
20:16.32PenguinThere are two extensions.
20:16.32FLeiXiuS[TK]D-Fender, I'll try without the hangups within the 'h'
20:17.32FLeiXiuSPenguin, how so?
20:17.58Mon|A|rch[TK]D-Fender, any idea how to improve call quality on outbound calls? I'm in the us, so I'd imagine i should use ulaw, and disable alaw
20:18.00PenguinWhile I can see the context of 'control' being asserted, I doubt that is the context within which the extension actually resides.  I feel there is an include involved.
20:18.10Mon|A|rchfrom what i understand they're meant for us and european connections respectively
20:18.24[TK]D-FenderMon|A|rch, ULAW vs ALAW should have no real quality difference at all.
20:18.38Mon|A|rchhm
20:18.41Mon|A|rchgsm?
20:19.01Mon|A|rchdo i need to poke around in my provider server?
20:19.04[TK]D-FenderMon|A|rch, If it's stuttery it could be network jitter or bandwidth.  May sure other things don't cut in on that (QoS), etc
20:19.09Mon|A|rchI just sort of assumed this was an asterisk issue
20:19.13FLeiXiuSPenguin, The call is made through AMI, directly to this context / 's' extension.
20:19.25Mon|A|rchgood call
20:19.29FLeiXiuSI believe removing the Hangup() from the 'h' exten worked.
20:19.36PenguinThe main difference between ulaw and alaw in the US is that almost no one in the US uses alaw.
20:20.00Penguinfleixius: So the call goes to s@control, then.
20:20.09[TK]D-Fenderthe companding between ULAW and ALAW is almost identical and transcoding shows hardly nay quality loss at all.
20:20.20FLeiXiuSPenguin, Correct.
20:20.31Mon|A|rchgood to know
20:20.42PenguinThat broke a theory I was trying to work on.
20:21.07Mon|A|rchbtw, when I'm passing in my originate string, can i set channel variables? I'd like to be able to get transfers to whomever opens a session on the front-end
20:21.17Mon|A|rchtheir sessions have their particular extension
20:21.32[TK]D-FenderMon|A|rch, &setvar=var=value
20:21.32Mon|A|rchis it something like: "variable=var1=5|var2=4"?
20:21.40Mon|A|rchk
20:22.15Mon|A|rchthen access it like s,1,goto(incoming,${var},1)?
20:22.24Mon|A|rchor dial or w/e
20:22.26FLeiXiuSPenguin, [TK]D-Fender Removing the Hangup from the 'h' exten worked.  I guess it was acting as it was intended too.
20:23.06[TK]D-FenderMon|A|rch, yup
20:23.27Mon|A|rchcool beans
20:23.55*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
20:24.59DocfxitI am still having trouble getting it to record a prompt the latest version is at pastebin.com/ukN4WAv2
20:25.58navaismowhy? what sows the cli
20:26.12[TK]D-FenderDocfxit, Why are you not showing us the FAILURE?
20:26.28DocfxitI'd be happy to.
20:26.43DocfxitLet me see if I can get that.
20:26.48[TK]D-Fender"if"?
20:27.27[TK]D-FenderYou could be hitting a failure than isn't directly related to your code, but rather the working environment.
20:27.48[TK]D-FenderWhich makes this the equivalent of showing a brochure for a car ... and then why asking why YOURS crashed.
20:28.35Mon|A|rch[TK]D-Fender, after a call is originated, do i need to use answer() before dialing someone to put on the other end?
20:28.56Mon|A|rchI'm not entirely sure what answer does, guessing just general config for the call
20:29.03[TK]D-FenderMon|A|rch, Depends exactly where, and exactly why.
20:29.13[TK]D-FenderMon|A|rch, There are times you'll want to, others you specifcally don't
20:29.18PenguinIt answers the line, as strange as that may sound.
20:30.04Mon|A|rchwell, i suppose since i don't have a specific reason not to use it, I'll use it.
20:30.18[TK]D-FenderUsually one would do the REVERSE
20:30.22PenguinYour thinking is backward.
20:30.24Mon|A|rchall this extension will do is call someone out in the world, then patch them through to a CSR that will try to sell them crap
20:30.26[TK]D-FenderYou do things because you need to....
20:30.35*** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone)
20:30.42Mon|A|rchwell, generally i assume i need to answer the line
20:30.48PenguinLet the callee answer it.
20:31.06Mon|A|rchmakes sense
20:33.12[TK]D-FenderMon|A|rch, It can affect billing, success codes, retries, etc...
20:34.09DocfxitAs I am sure you can tell I don't work with Asterisk every day. (Actually very rarely) It's been very solid and I don't make changes very often.  I'm sorry I'm not more experienced with asterisk.  I am in root now.  I entered asterisk -vvvvvvr.  I made the call.  It doesn't show in the terminal window.  Isn't that the correct way to show cLi?
20:34.34Mon|A|rchgood to know tk
20:37.12[TK]D-FenderDocfxit, If you connect via that command as listed and see nothing then your call isn't even being processed
20:37.27navaismoDocfxit: try with the cmd: core set verbose 3
20:38.13[TK]D-Fender"I entered asterisk -vvvvvvr" <- that's already *6*
20:40.28navaismoLOL
20:40.34navaismomissed that
20:46.43Docfxitnavaismo: I'm getting command not found.  I am running an old version 1.4.22.  I do have a new computer that I am almost ready to load the latest version on.
20:47.38PenguinDisregard his suggestion anyway.
20:47.45navaismoyep ^
20:49.12[TK]D-FenderDocfxit, "set verbose 10
20:52.11PenguinSet it to 9000, if you want.
20:53.18FLeiXiuS[TK]D-Fender, Thanks btw.
20:53.30[TK]D-FenderFLeiXiuS, np
20:57.37JustinAikenAnyone know how to sort out occasional DTMF-doubled tones?
20:57.56JustinAikenLooking at a tcp dump in wireshark, it looks like we occasonally get a DTMF end packet that shouldn't be there,
20:58.04JustinAikengiving us really short tones (30ms) or so
20:58.10JustinAikenis there a way to reject those?
21:01.27DocfxitI'm at root@ubuntuAsterisk:~# set verbose 9000       I made the call.  Nothing is showing in terminal.
21:01.57PenguinThat is not the asterisk cli.
21:02.56DocfxitOh good.  How can I get to the cli?
21:03.07Penguinasterisk -r
21:03.31navaismoyou said "asterisk -vvvvvvr"
21:03.39navaismodo it again
21:05.09DocfxitWhen I put in asterisk -r I get what is in pastebin.com/cJFVcHVR
21:05.33PenguinUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
21:05.46PenguinThat means asterisk might not be running!
21:05.55DocfxitThere is a file in /var/run/asterisk/asterisk.ctl
21:06.08DocfxitThat is the only file in that folder.
21:06.35Penguinpgrep asterisk
21:06.41PenguinDoes it return the PID?
21:07.11DocfxitYes.  It returns two pid's.
21:07.29Docfxit5367      and     5377
21:08.03Penguinps -C asterisk u
21:08.09PenguinWhich user/uid is asterisk running under?
21:09.05Docfxitroot 5377
21:09.16PenguinRoot?  Good grief.
21:09.38PenguinWho is the owner of /var/run/asterisk/asterisk.ctl?
21:11.01DocfxitThe /var/run/asterisk folder was there last night.  It isn't there today.
21:11.26PenguinCreate it and then restart asterisk.
21:12.19DocfxitI am currently signed in as root.  Is it ok to create it with the user root?
21:12.32PenguinYes.
21:12.56PenguinI'm not a big fan of your running asterisk with root, but that's another lesson.
21:13.42*** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke)
21:16.01DocfxitHopefully soon I will find someone to help me build another box correctly and these problems will go away.
21:16.16*** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com)
21:16.45PenguinFor an easy deployment, I would use AsteriskNOW with the "No GUI" install option.
21:17.12PenguinIt's been a while since I did it, but it went rather well that last time I did it.
21:17.45DocfxitWhy the "No GUI"
21:18.06*** join/#asterisk pbxbrian (~pbxbrian@79.97.2.26)
21:18.23PenguinGUIs are junk, they break things, make it impossible to work out some problems, and we don't support them here.
21:18.37Docfxitok.
21:18.59DocfxitSomeone else suggested I go with FreePBX
21:19.09PenguinI'm sure they did.
21:19.18carrarFreeSwitch!
21:19.25ChainsawDocfxit: Then it's up to the someone else to do your FreePBX support work really.
21:19.50DocfxitI'd rather be here.
21:20.14*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
21:23.44DocfxitI created the folder /var/run/asterisk   I re-booted,  The folder isn't there now.
21:24.04PenguinI don't remember advising you to reboot.
21:24.04[TK]D-FenderI think you should look at what you need out of yoru system, who'll maintain, it and prioritize accordingly/.
21:25.26*** join/#asterisk Russ (~russ@md20536d0.tmodns.net)
21:27.02DocfxitPenguin: I'm sorry.  You asked me to restart asterisk and since asterisk -r gives me an error the only way I know how is to re-boot.  I will re-create the folder now.
21:28.10DocfxitI currently have the folder /var/run/asterisk.  How should I restart asterisk?
21:30.06PenguinWith the directory existing, start or restart asterisk.
21:32.26DocfxitThe directory does exist now.  I tried sudo asterisk -vvvvvvr     It didn't create a file in /var/run/asterisk    It did give me the error saying asterisk.ctl doesn't exist.
21:32.52PenguinWhen you start or restart asterisk, it will output information.
21:33.09PenguinIf it cannot create the ctl file, it should say so.
21:33.45PenguinHow are you starting and/or restarting asterisk?
21:34.33DocfxitThe information it outputs is in pastebin.com/jz4pxsTv
21:35.04DocfxitI'm restarting it with sudo asterisk -vvvvvvr
21:35.10PenguinThat isn't starting OR restarting asterisk.
21:35.19PenguinAnd you're running that command as a user anyway.
21:35.33Penguinasterisk -r connects to a RUNNING ASTERISK.
21:35.40PenguinYou have to actually be running asterisk first.
21:36.48DocfxitI believe asterisk is running.  When I call it answers the phone, plays the prompts and lets me connect to an extension.
21:36.52Penguinsudo -i
21:36.54Penguin/etc/init.d/asterisk restart
21:36.56Penguinasterisk -r
21:37.34PenguinYou've failed to restart asterisk.  That was key in the instructions.
21:38.32DocfxitI'm sorry.
21:38.44DocfxitI just ran the commands you gave me.
21:39.51DocfxitThe results are in pastebin.com/FyVrudeu
21:40.12PenguinDoes the file exist?
21:40.31PenguinDoes the directory still exist?
21:41.10DocfxitNothing is in /var/run/asterisk
21:41.41PenguinPastebin the contents of your asterisk.conf.
21:42.13DocfxitYes the directory does exist.
21:43.57DocfxitIt's in pastebin.com/myXN6QLc
21:45.40DocfxitIs the astrundir => supposed to be /var/run/asterisk    or is it supposed to be /var/run?
21:46.17*** part/#asterisk mjordan (~mjordan@nat/digium/x-mangkwmxpwbejrkx)
21:51.58*** join/#asterisk Docfxit (~Docfxit@netblock-75-79-6-10.dslextreme.com)
21:54.31*** join/#asterisk k1920 (~k1920@160-6.5-85.cust.bluewin.ch)
21:55.43*** part/#asterisk navaismo (~navaismo@189.191.2.44)
21:57.29JustinAikenanyone know a fix for occasional doubled DTMF tones?
22:01.31k1920more informations about "occasional"? always same phone/number? always same time? do you have a single line? did you try a verbose debug?
22:01.46*** join/#asterisk navaismo (~navaismo@189.191.2.44)
22:02.27JustinAikenabout 1 in 10 digits will be doubled, any number
22:02.35JustinAikenso you type 1234 into the phone, get 122345
22:02.50JustinAikenseems nearly completely random
22:03.22JustinAikenlike in this: https://gist.github.com/c7b483f9cd5dc37e7a33
22:04.30Penguin/var/run/asterisk would be good.
22:04.39lifeformsbyeee!
22:04.50DocfxitPenguin: great.
22:05.00DocfxitThat's where it is.
22:06.33DocfxitPenguin:  It sounds like it should be writing the file asterisk.ctl to that folder?
22:07.23Penguinwell, directory, but yes.
22:08.18PenguinGo ahead and kill off any asterisk processes you have.  Kill them forcefully if necessary.
22:09.20DocfxitWith pkill asterisk ?
22:09.34PenguinThat should be fine, yes.
22:10.00PenguinBe sure they are gone using pgrep or any other way you know how to ensure asterisk is dead.
22:12.13Docfxitpgrep asterisk doesn't return anything and the phone doesn't answer.  So it's dead.
22:12.54k1920Well done Penguin.. little tired.. bye all
22:13.48PenguinNow start asterisk using the following command:  asterisk -vvvvddddc
22:13.53DocfxitPenguin:  Thank you for sticking with me.
22:14.36DocfxitI am now into cli.    Yey!!!
22:14.46PenguinI don't know about that.
22:14.54*** join/#asterisk charley (charley@epicboise.com)
22:14.58PenguinYou should have just started asterisk and that's all.
22:16.02DocfxitThere is a lot scrolling on the screen.
22:16.07PenguinOnce that is running, then, in another terminal, find out if the asterisk.ctl file exists.
22:16.47charleyI know this isn't the right channel.. but I can't find any Broadsoft users. My VoIP engineer needs to convert a *.vml file to .wav, or something that doesn't need a 3rd party player to listen to it. Anyone know where I can get one? He's scoured the web and hasn't had any luck.
22:17.30PenguinI recall running asterisk 1.4.22 and there was no problem with the socket file.
22:17.44DocfxitPenguin:  asterisk.ctl is in there and asterisk.pid
22:17.51PenguinGreat!
22:17.58PenguinSo asterisk isn't broken.
22:18.11DocfxitIt's been a while since I've seen them.
22:18.13PenguinIt's something else.  Could be something in the init fiel.
22:18.22Penguins/fiel/file/
22:19.25navaismoor selinux
22:19.32DocfxitWhat directory is the init file in?
22:19.35PenguinNot on ubuntu.
22:19.42Penguin/etc/init.d/
22:20.42PenguinUbuntu does have some security thing, though.  I think it's called App Armor.
22:20.54Penguindoesn't use ubuntu.
22:21.47DocfxitWhat do you use?
22:22.00PenguinPrimarily Arch Linux.
22:22.07PenguinBut I don't mind CentOS.
22:22.45charleyanything is better than ubuntu, really.
22:23.34PenguinAmen.
22:24.58charleyWas it Ubuntu's logo that was so appealing when it blew up? I don't understand. Was it firefox fanboys that were on Ubuntu's nuts?
22:26.08sruffellI think it was it's willingness to make proprietary software available…especially when the open source video drivers were not existent or lacking.
22:26.20Penguinits
22:26.28sruffellthanks
22:26.46charleylol
22:27.03DocfxitThere are a few warnings in the cli.  pastebin.com./iwPUuFYT
22:27.17PenguinWarnings are usually okay.  Errors are bad.
22:27.40DocfxitI will call the extension now to see what is happening.
22:28.43DocfxitAsterisk isn't answering the phone.
22:32.07*** join/#asterisk navaismo (~navaismo@189.191.2.44)
22:32.22*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2)
22:33.57ghost75did you know there will be ubuntu phone?
22:34.08*** join/#asterisk Russ (~russ@md20536d0.tmodns.net)
22:35.19DocfxitNow I really have a problem because the phone isn't answering.
22:47.57DocfxitI restarted asterisk.  I called the extension 130.  I found a warning.  You can see it at pastebin.com./pGbVpGC7
22:48.50[TK]D-FenderDocfxit: paste FULL URLS's
22:49.01[TK]D-Fenderwe have to copy/paste it manually all the time
22:49.07navaismoyou dont have an extension 13 in your context voicemenu-custom-5
22:49.09Qwell^^^^^^
22:49.15navaismonot mine
22:49.37DocfxitI have included the new Voice_Pronpt_That_I_Recorded     in voicemenu-custom-5
22:49.43[TK]D-Fenderdocfxit: [Jan  8 15:02:11] WARNING[6134]: pbx.c:2514 __ast_pbx_run: Invalid extension '13', but no rule 'i' in context 'voicemenu-custom-5'    -- Hungup 'DAHDI/7-1'
22:49.47Qwellalso since when do browsers allow a trailing . in the domain?  That's funky.
22:49.53[TK]D-FenderDocfxit: that is 13 ... NOT 130
22:50.09DocfxitIt should be 130
22:50.40[TK]D-FenderDocfxit: It isn't   Pastebin entire calls, not tiny broken little snippets
22:50.54[TK]D-FenderDocfxit: And we are now in completely different contexts than before.
22:51.05[TK]D-FenderDocfxit: Be thorough in the configs and debug you show us.
22:52.46DocfxitOk.
22:53.41DocfxitI tried starting asterisk with asterisk -vvvvvvr       It removed the files from /var/run/asterisk.
23:01.19*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
23:02.06DocfxitWhen I use asterisk -vvvvddddc    Asterisk will start and the files come back
23:03.56DocfxitI tried a new call to extension 130.  Pastebin.com says I reached my limit of 10 pastes per 24 hr.  I guess I can sign up?
23:04.47JustinAikengist.github.com
23:14.46DocfxitJustinAiken:  Thank you.
23:15.25*** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com)
23:15.36[TK]D-Fender~pb
23:15.36infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
23:15.46[TK]D-Fender4 more right there
23:18.22DocfxitI started Asterisk with asterisk -vvvvddddc   I called x130 I pasted the results at https://gist.github.com/4488923   You will see the problem with x130 and asterisk is shutting down.. Why?
23:19.24*** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir)
23:23.13kaldemarDocfxit: what problem?
23:24.07[TK]D-FenderDocfxit:  Executing [130@voicemenu-custom-5:1] Answer("DAHDI/7-1", "") in new stack
23:24.12[TK]D-FenderDocfxit: It answers....
23:24.24[TK]D-FenderDocfxit: and you have NOT shown us your dialplan.  We have no idea what's in there now.
23:24.35kaldemarand falls through.
23:24.36[TK]D-FenderDocfxit: what part about "show us complete configs and debug" was unclear?
23:24.53DocfxitWhen I dial x130 asterisk should record a file in /tmp
23:24.58[TK]D-FenderDocfxit: How can we tell you what's wrong with your code ro call without seeing them both?
23:25.01*** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com)
23:25.07[TK]D-FenderDocfxit: WHERE IS THE DIALPLAN?
23:25.08*** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri)
23:26.26DocfxitThe dial plan in at http://pastebin.com/JJPYZwRx
23:27.34[TK]D-Fender[Voice_Prompt_That_I_Recorded]
23:27.39[TK]D-Fender<PROTECTED>
23:27.42kaldemarwrong context.
23:27.54[TK]D-Fenderthe context you are in has nothing to do with the dialplan you created in that OTHER context
23:28.54DocfxitI don't understand.  What do I need to do to fix it?
23:28.58[TK]D-Fendervoicemenu-custom-5  <---- YOU ARE HERE
23:29.05DocfxitYes.
23:29.10[TK]D-Fender[Voice_Prompt_That_I_Recorded] <--- your CODE is here
23:29.31[TK]D-FenderIf I tell you the papers are in the top shelf... and they AREN'T there.  Your search is going to FAIL
23:29.41DocfxitI included Voice_Prompt_That_I_Recorded in voicemenu-custom-5
23:29.48[TK]D-FenderWhere do WE see this?
23:30.03DocfxitI will paste it.
23:30.07[TK]D-Fender\docYou seem to have a very large problem with the idea of showing us COMPLETE information
23:30.17kaldemara misspelled include was mentioned earlier.
23:31.04kaldemarpron pt
23:34.25Docfxithttp://bin.cakephp.org/view/1230663065
23:35.30kaldemarand what do you have in [default]?
23:35.52DocfxitI will paste that for you.
23:38.14kaldemarif it has a matching extension, it will always be used first because you include it first. besides using default in the first place is often bad practice.
23:39.20DocfxitThat is the way the first person that set it for me did it.
23:39.39DocfxitI have updated http://bin.cakephp.org/view/1230663065   to include default
23:40.06[TK]D-FenderDocfxit:  ....
23:40.13[TK]D-FenderDocfxit: You're running 1.4.2 ... right?
23:40.19DocfxitYes.
23:40.23[TK]D-FenderDocfxit: "same" did not even EXIST THEN
23:40.25[TK]D-Fender^^^^^^^^^6
23:40.37[TK]D-FenderYou are using NEW config options thta do not exist at all
23:40.57[TK]D-FenderDocfxit: All of those lines are BROKEN
23:40.59*** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey)
23:41.49DocfxitI was just following what someone told me to do yesterday.  I will change it back now.  Thanks for catching that.
23:45.45*** join/#asterisk doctorray (~ray@72.26.99.19)
23:47.30[TK]D-Fenderheads out for a few hours
23:50.13Docfxit[TK]D-Fender:  Thank you for your help.
23:52.16DocfxitI have updated http://bin.cakephp.org/view/1230663065 to show the dial plan and I have the cli error here  http://bin.cakephp.org/view/675566775
23:53.43*** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)
23:53.43*** mode/#asterisk [+o pabelanger] by ChanServ
23:57.21navaismo[TK]D-Fender,  said you need to remove same and use exten
23:57.42Docfxitnavaismo:   I did that.
23:58.04DocfxitYou can see that in my paste.
23:58.12navaismohere http://bin.cakephp.org/view/1230663065??
23:58.30navaismocuase im seein same string there
23:58.42navaismoseeing*
23:59.03*** join/#asterisk fisted_ (~fisted@unaffiliated/fisted)
23:59.45DocfxitSorry.   It's in http://bin.cakephp.org/save/85620

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