IRC log for #asterisk on 20110206

00:05.34*** join/#asterisk erinspice (~erin@207.98.195.107)
00:06.30*** join/#asterisk atan2 (~atan@unaffiliated/atan)
00:06.47atan2Guys having an issue calling voicemail using a grandstream adapter. I get "vm-password (format 0x1 (g723))" error message in console
00:08.18*** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com)
00:08.45atan2I assuem the codec in the device is set wrong. Err, but not sure which to use.
00:12.11atan2Oh great now the phones stream when I pick them up
00:13.33*** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-nllndaccpzcazckw)
00:13.55*** join/#asterisk mr_ian (~mr_ian@S0106001b63f49383.du.shawcable.net)
00:19.50atan2err, my bad. Client left phone off hook. Figures.
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00:34.17*** join/#asterisk GTXComm (~John@cpe-72-128-62-30.kc.res.rr.com)
00:39.38GTXCommQ I am running asterisk 1.6.0.6 as a provider for a Elastix system...  Sip trunk setup, when I dial a to the other box it gives me a message chan_sip.c:15139 handle_response_invite: Failed to authenticate on INVITE to '"WIRELESS CALLER" <sip:8165298686@63.76.55.135>;tag=as64fd44e6'...  I can call outbound, but it doesn't seem to want to take incoming registration on SIP.  I use IAX2 on inbound, and it works, but not on SP inbound.  The Ela
00:39.38GTXCommstix (FreePBx0 is version 1.4.19.  I am not used to setting up asterisk to asterisk SIP.
00:40.56GTXComm1st question. on this.. hope i am okay asking.
00:41.25GTXComm<--- new
00:41.31GTXCommhere
00:42.29GTXCommDo I need to have a special registar string on both servers, on my sip.conf
00:42.35GTXComm?
00:46.30JerJeryou need like defaultuser=  on the peer
00:50.10GTXCommOk, on the other side do I need to have a recognizable DID on ibound?  Because my sip is [c20024-101] so set up defaultuser=[c20024=101] ?
00:50.49GTXCommor sorry defaultuser=c20024-101
00:52.20GTXCommThat was confusing, let me start over .. On the 1.4 server, do I need defaltuser= , and 1.6 pot in Defaultuser=?
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02:08.12RichardLynchHow can I use Dial macro with SIP and force it to go over TCP, without setting up a trunk?
02:12.32JerJeruhh - not sure if you can... i haven't noticed a way
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02:17.33fauxallianceRichardLynch, http://www.voip-info.org/wiki/view/Asterisk+tips+SIP+URI+Dial
02:20.09GTXCommI am having DTMF issues on IAX trunk.  Any suggestions?
02:20.24GTXCommWorks on sip.
02:21.10fauxallianceGTXComm, dtmfmode=rfc2833
02:21.23GTXCommThanks
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06:07.43p3nguinfauxalliance: There is no dtmfmode in IAX2.
06:08.12p3nguingtxcomm: See above statement to fauxalliance.
06:09.03GTXCommk
06:10.35GTXCommYa, I am still having issues; however, I did get my sip registered, and working the way I know how.  It is weird, that it wouldn't take alpha numeric username, for inbound calls.  It needed 10 digit phone number for the sip Peer.
06:11.38GTXCommTherefore, bypassing IAX. Well IAX should work better from what I hear.
06:12.35GTXComm<---- new here, so I don't really know the rules, if I am chatting too much, please let me know. :)
06:14.34GTXCommor Free PBX 1.4.X is just not something I am familiar with
06:16.06GTXCommI am have a free pbx test server that I have hooked up in my co-lo, that I am going to test compatibility with 1.6.x and find out the true issue.
06:16.37GTXCommfor IAX / to IAX
06:19.38GTXCommQ. Does IAX do (default) rfc2833?  I am having a hard time with google, and voip-info on this.
06:50.31Corydon76-homeGTXComm: No, it does not.  IAX2 does not do ANY signalling or audio on a port other than 4569
06:51.09Corydon76-homeGTXComm: this is also why IAX pierces firewalls so effectively
06:51.55Corydon76-homeThere's no need for a firewall to understand the protocol in order to route the right ports back; it just works.
06:52.22GTXCommThanks
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07:57.14kriegerodhi all. what can you say about asterisk 1.8 stability? can it be adviced for serious production use? thanks
08:38.47*** join/#asterisk TehRabbitt (~TehRabbit@pool-71-172-89-155.nwrknj.fios.verizon.net)
08:39.08TehRabbittQuick question, what is the difference between extensions.ael and extensions.conf?
08:39.16TehRabbittand which one should I be focusing on in 1.8
08:39.56p3nguinael is in, well, ael format... and will get converted to conf internally by asterisk.  You should be using conf unless you have a good reason to use something else.
08:40.10TehRabbittalrighty thanks :)
08:43.13p3nguingtxcomm: There is absolutely no reason whatsoever that you have to use a 10-digit numeric for a device's name.  Both SIP and IAX2 configs will allow you to use alphanumeric peer names.  Also, if this problem is occurring in FreePBX, you should probably either stop using FreePBX or ask about the problem in the more appropriate FreePBX channel (#freepbx).
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09:12.07TehRabbittHm, trying to get chan-sccp-b installed, but when running "make" on the source, it errors out :-\ any ideas?
09:12.38*** join/#asterisk astro__ (astro@silvia.drft.net)
09:13.16fenruspastebin the errors
09:13.25fenrusdid you install all dependencies ?
09:13.39TehRabbittyes, hold on i'm going to pastebin
09:14.15fenrusif you are using a packagesysetem you need the -dev packets of the dependecies aswell
09:14.57TehRabbitthttp://pastebin.com/XpvN1jC2
09:20.06TehRabbittany ideas?
09:22.50TehRabbitt:-\
09:24.26*** join/#asterisk sskyles_ (~sskyles@166.137.8.121)
09:24.42TehRabbittwhat causes this btw:
09:24.43TehRabbitt[Feb  6 04:23:42] WARNING[22666]: db.c:112 dbinit: Unable to open Asterisk database '/var/lib/asterisk/astdb': Permission denied
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09:28.57TehRabbittanyone here still :-\
09:28.58fenrusTehRabbitt, someone that dont have access to /var/lib/asterisk/astdb is trying to access it.
09:29.12fenrus"someone" as perhaps the asterisk user or your user ?
09:29.17fenruswhen does this error occur ?
09:29.25TehRabbittwhen i'm logged into the console
09:29.40fenrusas your user ?
09:29.43TehRabbitti do have a phone that is plugged in
09:29.55TehRabbittnot sure what you mean by as my user, i've just logged into the console as root
09:30.10fenrusokay
09:30.24fenrusthen it seems that the asteriskuser cant access that file. does it exist?
09:30.27fenrusare you suing astdb?
09:30.34TehRabbittnot that i'm aware of
09:31.07TehRabbittall I did was download teh 1.8 source, ran make / make install and then went to /etc/init.d/asterisk start... from there i went to the console (asterisk -r) and now i'm getting those messages
09:31.20fenrusokay
09:32.20fenrusi would unload the module
09:32.31TehRabbittunload what module? i haven't loaded any
09:32.42TehRabbitthaven't even installed the chan-sccp-b one yet
09:32.47TehRabbittit won't compile :-\
09:33.00fenrusare you certain that its not built and loaded by default ?
09:33.19fenruswrite module show
09:33.20TehRabbittpositive, you need to enable it... the .so isn't even listed
09:33.21fenrusin your console
09:33.45fenrusfunc_db.so                     Database (astdb) related dialplan
09:34.17TehRabbittah you mean that module, perhaps it is enabled by default
09:34.50fenrusor create the db file and make sure taht asterisk kan read/write to it
09:35.42*** part/#asterisk sskyles (~sskyles@166.137.8.121)
09:35.45TehRabbittwhat is the point of the db file though?
09:38.41TehRabbitti made the file and i'm still getting the error :-\
09:38.41TehRabbitt[Feb  6 04:37:51] WARNING[22846]: db.c:553 ast_db_gettree: Database unavailable
09:39.01fenrustheres no database index in the file :D
09:39.14TehRabbitthow do i make an index 0_o
09:40.10fenrusi'd try the database -commands in the CLI
09:40.41TehRabbittNo such command 'database -commands' (type 'core show help database -commands' for other possible commands)
09:40.57fenruscore show help database
09:41.43TehRabbittthre isn't anything there to create an index
09:43.20fenrusim not even sure that you need an index, but i guess it complains because you gave it a flat file, and it wants some kinda database format
09:43.32fenrusbut if you ask asterisk to create it itself you might get it right
09:43.41fenrusor disable it if you're not using it.
09:45.40TehRabbittdisabled it lol
09:45.55TehRabbittstill can't seem to figure out why i can't complie chan-sccp-b though
09:47.34TehRabbittchecking if the linker (/usr/bin/ld -m elf_x86_64) is GNU ld... yes checking whether the g++ linker (/usr/bin/ld -m elf_x86_64) supports shared libraries... yes checking for g++ option to produce PIC... -fPIC -DPIC checking if g++ PIC flag -fPIC -DPIC works... yes checking if g++ static flag -static works... yes checking if g++ supports -c -o file.o... yes checking if g++ supports -c -o file.o... (cached) yes checking whe
09:47.36TehRabbittwhoops
09:47.42TehRabbittsorry :-\
09:47.48TehRabbitthttp://pastebin.com/BHvGpbSR
09:47.51TehRabbittmeant to send that link
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10:27.09StaRetjiFolks, what could be the problem where I hear signal tone with dahdi, but if I press button on the phone nothing happens.
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10:27.41StaRetjiI can hear key tone, but signal tone do not change, so I can't even call 600 echo even though it was there
10:28.01as001hi is it possible to setup moh for radio streaming on station which emits program at 128Kbit/s ?
10:28.01StaRetjiasterisk is set to verbose 15
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10:29.43as001When I set it on 24 kbps station it works great, but when i try on 128 or 64 sound is choppy
10:29.56as001I am using 1.6.2.16
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10:33.00StaRetjiit's almost like dial tones are not recognize, CLI doesn't show anything, as it nothing was pressed.
10:33.04StaRetjiAnyone? :/
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11:01.45CaseSensitivedoes anybody know of a resource with an agi script that can load tagged values in csv format by key ? like (var1=value,var2=value)
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12:22.02verywisemanhow can i hide caller id and my extension number when i make call ?
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17:05.05philippel_macping anyone around who has some experience with the CallCompletionRequest/Cancel  app in 1.8 that I could bounce a couple issues off of?
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20:48.02domains25hello everyone
20:52.37domains25anyone can help with a x100p that does not want to call out to a pstn trunk?
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21:13.39MikeHcan anyone identify what the cause of the following error may be?
21:13.40MikeH[Feb 6 00:26:50] WARNING[3872]: chan_dahdi.c:2689 dahdi_enable_ec: Unable to enable echo cancellation on channel 1 (No such device)
21:13.55MikeHgetting terrible echo on connected SIP phones
21:18.24TehRabbittwhen i try to run /etc/init.d/asterisk stop it works, but when I try running start, it doesn't start asterisk, any ideas?
21:18.49b14ckTehRabbitt: look at /var/log/asterisk/full
21:18.57b14ckerr: /var/log/asterisk/messages sorry
21:19.01TehRabbittalso for some reason, it keeps trying to run chan_skinny even though I set it for noload => chan_skinny.so
21:19.31TehRabbittb14ck: nothing in the messages :-\
21:19.36b14ckThen do:
21:19.47b14ck/etc/init.d/asterisk start; asterisk -r (maybe you'll connect to the term)
21:19.52b14ckand then see what happens
21:19.56b14cksometimes it'lls tartup
21:19.57b14ckthen fail immediately
21:20.09b14ckbut in general, look in all of the /var/log/asterisk/* logs
21:20.16b14ckit will have errors in there somewhere
21:20.19TehRabbitt[Feb  6 16:19:33] ERROR[17157]: logger.c:1042 init_logger: Unable to create event log: Permission denied
21:20.29TehRabbitti guess that's why
21:20.33b14ckthere you do =p
21:20.35b14ck*go
21:20.42b14cklook at /etc/init.d/asterisk
21:20.46b14ckwhat USER it is set to run as?
21:20.57TehRabbittnot sure, how do I set that?
21:21.06b14ckits a variable defined in /etc/init.d/asterisk
21:21.09TehRabbitti'm running the command as root, but i'm not sure if it's actually trying to run as root
21:21.10TehRabbitthm
21:21.16b14cklook at that flie
21:21.17b14ck*file
21:21.20b14ckvim /etc/init.d/asterisk
21:21.37TehRabbittUSER=$NAME GROUP=$USER
21:21.45TehRabbittAh
21:21.48TehRabbittNAME= Asterisk
21:22.43TehRabbittchanged it to root, and now I get this lovely message: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
21:22.57b14ckdo:
21:23.11b14ck/etc/init.d/asterisk stop; /etc/init.d/asterisk start; asterisk -r;
21:23.29TehRabbittGrr :-\ looks like the files are all owned by asterisk
21:23.39b14ckthat doesn't matter
21:23.42b14ckroot can read any file
21:23.45TehRabbittStarting as root not supported.
21:23.54b14ckalthough, if you want to change it, you can do: `chown -R root:root /etc/asterisk`
21:24.04b14ckHow did you install asterisk? And what OS? :o
21:24.12TehRabbittDebian, installed 1.6 from source
21:24.17TehRabbittlogged in as root
21:24.21b14ckcompiled as root?
21:24.23TehRabbittyes
21:24.30b14cksomething bad happened in your install somewhere
21:24.44b14ckcan you wipe your box and start fresh?
21:24.55b14ckI've got some good install instructions if you want them.
21:24.56TehRabbittyou mean remove asterisk?
21:25.01b14cki mean wipe the os
21:25.05b14ckand start from scratch
21:25.13TehRabbittNoooo
21:25.26b14ckAre you running other services on this box too? :o
21:25.31TehRabbittYes... :-\
21:25.35b14cknot good =/
21:25.49TehRabbittwell it runs a couple VMs
21:25.57b14ckwell, you can attempt to rm everything
21:26.11TehRabbittwell thats what i did last time, i had 1.4 installed originally
21:26.21b14ckrm -rf /etc/asterisk /var/lib/asterisk /etc/init.d/asterisk /etc/init.d/dahdi /var/spool/asterisk
21:26.36TehRabbittand then recompile i guess? :-\
21:26.44b14ckyah, but follow my instructions
21:26.50b14cksec
21:26.53b14ckhttp://projectb14ck.org/transparent-telephony-part-2-installing-aster
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21:28.34TehRabbittit says for centos
21:28.46TehRabbittah theres an ubuntu one
21:30.47TehRabbitthm, would running make menuconfig before running make screw stuff up?
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21:38.20TehRabbittok, i've installed asterisk using source, ran the make install, and now when I try to do /etc/init.d/asterisk start it says it doesn't exist :-\
21:38.45superbeefyou'll have to make your own init.d script
21:38.51superbeefit uses a script called safe_asterisk
21:38.55superbeefwhich is basically a loop
21:38.58TehRabbitthow would I do that?
21:39.06superbeefwhat os
21:39.10TehRabbittdebian
21:39.13TehRabbittsquzeeze
21:39.19TehRabbittsqueeze*
21:40.49superbeefhere's one i use on ubuntu
21:40.51superbeefshoudl be close enogh
21:40.52superbeefhttp://pastebin.com/cSrJ035v
21:41.03superbeefmodify it as needed and add it with rc-update or whatever it is that debian uses
21:41.54TehRabbitt-bash: /etc/init.d/asterisk: Permission denied
21:42.08superbeefewps
21:42.13superbeefthat script isn't so good anyway because that's for freepbx
21:42.20TehRabbittlol
21:42.46superbeefmost people
21:42.49superbeefjsut run safe_asterisk
21:43.08TehRabbittand how do i do that thoguh
21:43.19superbeefand add /usr/sbin/safe_asterisk into /etc/rc.local
21:43.38superbeefjust run it from the shell
21:43.42superbeefshould be in your path
21:43.45superbeef"safe_asterisk"
21:43.55TehRabbittah it works
21:48.52TehRabbittso what is safe_asterisk in comparison to asterisk?
21:49.59drmessanoIt's a script that runs asterisk
21:50.28drmessanohttp://linux.die.net/man/8/safe_asterisk
21:50.33TehRabbittdrmessano it seems like there is nothign iniside /etc/asterisk anymore :-\
21:51.24TehRabbittwhere would the .conf files be?
21:51.57TehRabbitthm nvm it's there, weird
21:52.06drmessanoO.o
21:52.28*** join/#asterisk DelphiWorld (~VoIpGuy@41.200.16.55)
21:52.30DelphiWorldhi guys
21:52.31TehRabbitti did a cd /etc/asterisk and it said it couldn't be found...  i did just cd /etc then cd asterisk and it worked 0_o
21:52.33DelphiWorldhere we go aguin.
21:52.41DelphiWorldmodule iax2 is unable to be loaded for me
21:52.44DelphiWorldalways same issue
21:52.52DelphiWorldi'm sure i'm missing something else. any help?
21:53.41superbeeftechrabbit.. did you do make samples
21:53.51superbeefdo install example config scripts
21:55.04TehRabbittyea they're there now
21:55.27TehRabbittfor some reason the files were there, maybe i mispelt it or something 0_o
21:55.35*** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com)
21:55.46TehRabbittnow i just have to find out why it's rejecting my phone :-\
21:56.05DelphiWorldsuperbeef: my issue is that sip and iax channel isn't loaded :(
21:56.22superbeefnot loaded or not configured?
21:56.39DelphiWorldsuperbeef: you you rock rock
21:56.43DelphiWorldsuperbeef: make samples fixed it
21:56.58superbeefword
21:58.33TehRabbitthow can I confirm the module is installed?
21:58.40TehRabbittfor chan_sccp-b
21:59.05DelphiWorldTehRabbitt: module load chan_you.so :-P
21:59.26TehRabbittyea just figured that out 0_o lol and yet another error... i swear it can never just *work* lol
21:59.27TehRabbitt[Feb  6 16:58:57] WARNING[1610]: loader.c:428 load_dynamic_module: Error loading module 'chan_sccp': /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: ast_rtp_change_source
21:59.45Kobazdo you need sccp?
21:59.52TehRabbitt[Feb  6 16:58:57] WARNING[1610]: loader.c:781 load_resource: Module 'chan_sccp' could not be loaded.
21:59.56TehRabbittKobaz: yes I do
22:00.10Kobazokay so, the easy way isn't possible (just not loading it)
22:00.14TehRabbittlol
22:00.38Kobazsccp probably has a dependency on another module
22:00.52Kobazeither that, or it's broken in your asterisk version
22:01.11TehRabbitthttp://chan-sccp-b.sourceforge.net/
22:01.14TehRabbittthats what i'm trying to use
22:01.22TehRabbitti'm using the latest trunk via SVN
22:02.34Kobazno guarantees that trunk will work at all
22:02.47TehRabbittAh
22:02.52TehRabbittso go for one of the stable ones i guess?
22:03.01Kobazyes, most definitly
22:03.28TehRabbittaight, so how should I go about removing the old module :-\
22:03.39TehRabbittor would running make clean do it?
22:03.42Kobazwipe out /var/lib/asterisk/modules
22:04.00TehRabbittand reinstall asterisk again?
22:04.20Kobaza stable asterisk version
22:04.41TehRabbittwell it's just the chan-sccp-b one that i got from svn
22:04.47TehRabbittasterisk is 1.6 via the site
22:04.54DelphiWorlduse 1.8
22:05.11TehRabbittcan't use 1.8 with chan_sccp-b :-\ it just doesn't compile the module
22:06.14DelphiWorldTehRabbitt: contact cisco :-D
22:06.37TehRabbittDelphiWorld: Funny... lol
22:06.57DelphiWorld:)
22:07.52*** join/#asterisk Dr-Linux (~Dr-Linux@182.177.131.100)
22:08.15Dr-Linuxwhy should i go to 1.6.2 and not for 1.8.x?
22:08.29Dr-Linuxshould go for*
22:28.00*** join/#asterisk Faithful (~Faithful@carame.lnk.telstra.net)
22:40.05*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
22:50.48*** join/#asterisk IsUp (IsUp@unaffiliated/isup)
23:03.59DelphiWorldif i see a peer unmonitored
23:04.02DelphiWorldhow do i make it monitored?
23:04.07DelphiWorldpuzzled: :)
23:06.47DelphiWorldchanged from peer to friend...
23:09.15Tozz_add qualify=yes to the peer
23:09.17Tozz_in sip.conf
23:10.51DelphiWorldTozz_: is iax... but i allready got it
23:10.54DelphiWorldthank you Tozz_
23:10.56Tozz_ah ok
23:10.57Tozz_np
23:11.43*** join/#asterisk jql (~jql@12.9a.344a.static.theplanet.com)
23:12.42DelphiWorldTozz_: now i'm stuck in iax registration, is not registering itself to the provider
23:13.37Tozz_u hgave a register => line?
23:14.12DelphiWorldTozz_: yeah and unable to receyv call :(
23:15.29IsUp'iax2 show registry' on your CLI
23:18.48DelphiWorldIsUp: iax2 show registry isn't this show iax2 client registered to my asterisk?
23:19.16DelphiWorldIsUp: zero iax2 reg...
23:21.52*** join/#asterisk infobot (ibot@rikers.org)
23:21.52*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.3 (2010/01/26), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/
23:22.28DelphiWorldstrange.
23:22.43IsUpDelphiWorld: you have register => in your iax conf, right?
23:22.54DelphiWorldyes, IsUp
23:23.06IsUpso you are trying to register to your provider,
23:23.30IsUpyou need to use 'iax2 show registry'
23:23.36IsUptry 'iax2 reload' first
23:24.16DelphiWorldIsUp: yeah, registering to my providers
23:24.30DelphiWorldIsUp: still zero iax2 reg
23:25.05IsUpcan you post your iax.conf to pastebin please?
23:25.15IsUpmask any passwords before posting
23:26.41shortcircuitScenario: I have VoicePulse connecting PSTN to my Asterisk server with a static IPv4 address. I have reliable 6to4 on the same server, and reliable 6to4 at home. Can Asterisk/SIP work directly, IPv6 to IPv6? It saves me setting up and managing a proxy SIP gateway on my router.
23:27.04DelphiWorldIsUp: iax.conf or only the peer?
23:27.43IsUpDelphiWorld: whole iax.conf
23:29.13DelphiWorldIsUp: here is my peer http://dpaste.de/UyaG/
23:29.25DelphiWorldIsUp: while give iax2.conf... iax3 ;-)
23:30.33IsUpDelphiWorld: make sure your register => is under [general] context, not a peer
23:30.45IsUpalso, do 'core set verbose 10'
23:30.50IsUpand 'iax2 reload'
23:31.42IsUpif you cant see any on 'iax2 show registry', do 'iax2 set debug on' and then 'iax2 reload'
23:31.56DelphiWorldIsUp: http://dpaste.de/UUqL/
23:32.00DelphiWorldIsUp: is my iax.conf
23:32.58IsUpgo to line 212, put your 'register => user:pass@london.voip.ms' there
23:33.54DelphiWorldIsUp: i did... let see
23:34.20DelphiWorldwow, IsUp
23:34.30MikeHhrm
23:34.34DelphiWorldIsUp: why isn't registering me using my peer definition and is working in line 212?
23:34.58IsUpregister => definition must be under [general] context
23:35.10MikeHis it considered secure to have a SIP port open to the internet?
23:35.24DelphiWorldMikeH: if no, avoid using VoIp.
23:35.26IsUpyou can do #include "iax_registrations.conf" on your line 212
23:35.40MikeHDelphiWorld?
23:35.58DelphiWorldMikeH: is sure, but you have to take care about your PBX security
23:36.13MikeHDelphiWorld, This is my concern
23:36.29MikeHI have no issues with opening it up - all accounts have passwords
23:36.31DelphiWorldIsUp: :)
23:36.31MikeHand are fairly secure
23:36.33DelphiWorldIsUp: good catch
23:36.44MikeHbut the concern is still there
23:37.04MikeHperhaps I'll use SSH tunelling
23:37.23IsUpDelphiWorld: ;)
23:38.04IsUpMikeH: sip scanning is going wild, they can hammer your server. i prefer close 5060 to strangers.
23:38.13IsUpor use fail2ban if your 5060 open
23:39.02MikeHI'll use ssh tunnelling I think :)
23:40.28IsUpi dont know how it works but lemme say again, always keep your 5060 closed!
23:40.51DelphiWorldMikeH: ssl/tls
23:42.05MikeHalso
23:42.12MikeHcan anyone recommend a windows based SIP client?
23:42.47DelphiWorldMikeH: phoner
23:43.11IsUpMikeH: xlite/eyebeam
23:44.04DelphiWorldIsUp: exten => 883510009900637,1,Echo()
23:44.06DelphiWorldIsUp: is this correct?
23:44.23IsUpDelphiWorld: yes, but you dont need ()
23:46.19DelphiWorldIsUp: hanging up directly
23:46.27DelphiWorldexten => 883510009900637,1,Answer()
23:46.27DelphiWorldexten => 883510009900637,1,Echo
23:47.39IsUpDelphiWorld: can you see any output on CLI when call hits your PBX?
23:47.50DelphiWorldyeah, IsUp... i see :-)
23:49.11DelphiWorld-- Call accepted by 78.129.153.20 (format ulaw)
23:49.11DelphiWorld-- Format for call is (ulaw)
23:49.11DelphiWorld-- Hungup 'IAX2/voipms-18460'
23:49.17DelphiWorldIsUp: this is the log
23:49.21DelphiWorldleifmadsen: don't ban me :-)
23:50.08DelphiWorldIsUp: ok, IsUp! when thay will be down? :-)
23:50.22IsDown-no-IsUpIsDown-no-IsUp: :P
23:50.27IsDown-no-IsUpIsUp: :)
23:51.12DelphiWorldi think while give up on it now and go to sleep, is 1am
23:51.46IsUpyeap i am back
23:51.53IsUpworking or not?
23:53.30DelphiWorldIsUp: no, not working :-)
23:53.49MikeHIsUp, What is the "Domain" field on x-lite?
23:54.18IsUpIP of PBX
23:54.49IsUpDelphiWorld: paste your iax.conf to pastebin again, please
23:55.04DelphiWorldspsure
23:55.06DelphiWorldIsUp: sure
23:56.01DelphiWorldIsUp: http://dpaste.de/166A/
23:56.17IsUpDelphiWorld: and to 'core set verbose 10' and place a call to your DID and then paste output to pastebin
23:56.45IsUpDelphiWorld: i cant see voip.ms in your iax.conf
23:57.01DelphiWorldIsUp: because i do include :-)
23:58.13IsUppaste your provider context too
23:58.52DelphiWorldIsUp: i did put it in default
23:59.06DelphiWorldIsUp: log: http://dpaste.de/aRJP/

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