IRC log for #asterisk on 20131203

00:08.12*** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com)
00:10.04navaismodo i need to worry about this msg: [Dec  2 18:09:23] WARNING[1317][C-00000004]: res_srtp.c:406 ast_srtp_unprotect: SRTP unprotect failed with: authentication failure 10
00:12.53tm1000rofl @ navaismo and by proxy Penguin
00:14.53*** join/#asterisk threesome (~threesome@ip-94-113-13-143.net.upcbroadband.cz)
00:16.03navaismo:P
00:18.18navaismohmm updated to the latest SVN of asterisk 12 and ari wont load--->http://pastebin.com/0cQwSPEX
00:20.07pabelangernavaismo, I believe something is lacking documentation
00:20.16pabelangerand that's stopping the module from loading
00:20.33pabelangernavaismo, leedm777 would know more
00:21.18navaismothanks ill wait
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00:31.43*** join/#asterisk Artyx (~artyx@schizophrenia.googleplex.net)
00:32.04ArtyxIs it possible to hookflash a dahdi trunk (to take advantage of call-waiting provided by the telco)
00:33.25*** join/#asterisk vlad_starkov (~vlad_star@109.188.125.147)
00:34.19[TK]D-FenderArtyx: "core show application flash"
00:42.49*** join/#asterisk matjam (~chrome@styx.stupendous.net)
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01:23.23nextimeanyone using sipml5 or jssip?
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01:25.41*** mode/#asterisk [+o mjordan] by ChanServ
01:29.12*** join/#asterisk bitfury (~bitfury@unaffiliated/bitfury)
01:29.35bitfuryhey guys, anyone familiar with voip voip?
01:29.46bitfuryand their byod plans?
01:30.30bitfuryjust wondering if they're any good..
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01:57.02[TK]D-Fenderbitfury: link?
02:00.24bitfuryhttp://www.voipvoip.com/
02:00.48bitfurythis one is better:http://www.voipvoip.com/byod/
02:02.13[TK]D-Fender$.01/min is pretty decent ... you can find lower if you try hard, and these guys are virtually nameless here... I'd use them as a backup or disposable termination
02:02.49bitfuryok
02:03.17bitfuryit's for a small home office
02:03.31bitfurythanks!
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02:22.59Artyx[TK]D-Fender Thanks, thats exactly what i was trying to figure out
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02:37.15bitfury[TK]D-Fender: any reliable byod provider (pay as you go) provider you may know of?
02:37.48[TK]D-Fenderbitfury: pretty much EVERY provider I've seen is "BYOD"
02:38.18*** join/#asterisk D30 (~deo@58.71.19.178)
02:38.22[TK]D-Fenderbitfury: Only residential-only chump providers treat you like an idiot end-consumer and think the only thing you'll want/need is a single ATA
02:39.18*** join/#asterisk trystero (~al.f.zero@unaffiliated/transfinite)
02:39.19bitfury[TK]D-Fender: yeah, not everyone has pay as you go plans though
02:39.26bitfuryI've been looking into vocalocity
02:39.42[TK]D-FenderNot many don't
02:39.57[TK]D-Fenderyour scale is reversed or your search pool is very "unfortunate"
02:40.01bitfuryI would like to use it for international calling
02:41.32*** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net)
02:41.34[TK]D-Fenderinternation depends where you are, and where you're calling...
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02:44.49MaliutaLapwell I'd say it's more of a billing concept
02:46.09[TK]D-FenderTruer still... given as far as they know .. you could be anywhere
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02:54.32MaliutaLapReally it even depends on the rates you're being charged. I pay the same for calls to canada as I do for calls to australia
02:57.17ArtyxWhats the per minute charge for that call?
02:59.49MaliutaLap$0.08AU untimed
03:00.18MaliutaLapI regularly talk to my mother for 2 hours for $0.08AU
03:00.51MaliutaLapan that's from Brisbane, Queensland, Australia to Fort MacMurray, Alberta, Canada
03:01.32MaliutaLapIf I could get a pap onto their network it wouldn't even cost me that
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03:11.54lvlinuxMaliutaLap: what provider is that?
03:12.00Artyxmost of my dialing is just north america, i have some flat-rate business phone service, its pretty decent call quality, so i use that as dahdi trunk for my box
03:12.40Artyxgv for kicks, an excercise in futility at the time
03:13.39lvlinuxgv works fabulous with yate
03:13.51lvlinuxoops can i mention y*te on #asterisk?
03:13.52MaliutaLaplvlinux: pennytel http://www.pennytel.com.au/
03:14.55Artyxoh i got it working it just didn't want to work, something about having to login with my pc before connecting, that step eluded me
03:15.38lvlinuxyeah you have to go to gmail and use the gtalk applet thingy and make 1 call that way before it will let anything else connect. wierd
03:16.00lvlinuxi guess it initializes your gtalk+gv when you do that or something
03:16.00ArtyxDrove me bonkers
03:25.34XandriXPenguin: your code for the ip check thing gave me an idea earlier i could make it give me current weather if i do it right haha
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03:36.09XandriXnow all i need to do is find the sounds that you reference in a bit of code you shared with me so i can put them in there proper place hehe
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03:53.13Penguinxandrix: You can do all sorts of things with asterisk.  If you can't make it do it natively in dial plan like I did with the IP address thing, you can always script it in any AGI-compatible language and really expand what can be done.
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03:54.57LinoSPhollaaaaaa alguien habla español
03:55.15LinoSPsomeone speaks spanish?
03:56.11Artyxspeak? poquito, type not a word
03:57.18Artyxoh and thats "espanol" ... as in baja/california, mexican i think
03:57.34Artyxnot really spanish at all
03:58.51PenguinHablamos Inglés solamente aquí.
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04:03.02LinoSPsomeones  knows about programmimg a simple app in asterisk
04:03.14[TK]D-Fenderlike?
04:03.24[TK]D-Fendertaht could mean almost anything...
04:04.20LinoSPor where to find examples
04:04.41ChannelZthe apps/ directory of the source
04:16.51bitfuryhmm, weird I have FreePBX running and intrusion detection is missing
04:19.42XandriXPenguin: good to know :)
04:20.08[TK]D-Fenderbitfury: What were you expecting, and what did you actually install?
04:22.34bitfury[TK]D-Fender: PBX in a flash, asterisk 11.6 with FreePBX 2.11.0.11
04:22.45bitfuryexpecting a fail2ban module under intrusion detection
04:23.02bitfurywent over to modules, but not seeing it there either to enable it
04:23.10[TK]D-Fenderbitfury: Good... blame PIAF.  It's their bundle
04:23.17bitfurysweet
04:23.29[TK]D-Fenderbitfury: it is not part of the core of FreePBX.
04:23.44bitfuryi see
04:28.29LiuYanLinoSP: asterisk has some demo dial plans in it's /etc/asterisk/extensions.conf file,
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04:45.38PenguinIf I wanted to parse the value of externhost or externaddr that is set in sip.conf, what would be the best way to do it from dialplan?  Read the conf directly?  Is there an internal mechanism that contains the IP address when using externaddr and the IP address as resolved when using externhost?
04:46.59[TK]D-FenderPenguin: "sip show settings might show what is resolved for it
04:47.21[TK]D-FenderPenguin: I don't have any hostname-based servers to test that with though
04:47.41[TK]D-FenderPenguin: But if it's there you could SHELL+grep it
04:47.54PenguinNetwork Settings section shows Externhost (which is set) and Externaddr (which is resolved).
04:48.41[TK]D-FenderSounds like its a viable target then...
04:50.09PenguinIf you use externaddr instead of externhost, does it still show up there?  I use externhost, and it shows the name and also what it resolved to.
04:50.33PenguinI'd guess the name field is empty there.
04:51.24PenguinOh, I have another host that uses externaddr...
04:51.37Penguin<PROTECTED>
04:51.37Penguin<PROTECTED>
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04:52.05PenguinOkay, so that will work just fine for what I'm going to do.
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05:31.21Penguinasterisk -rx 'sip show settings'|awk -F: '/Externaddr/ {gsub(/[[:space:]]/,""); print $2}'
05:31.39PenguinI had a lot of trouble getting that to strip off the whitespace before the address.
05:31.46PenguinI got it, though.
05:35.12PenguinBUT........
05:35.36PenguinIt dies when putting it in ${SHELL()}
05:35.48bitfuryjust fyi, fail2ban is already loaded in PIAF installations.
05:36.31Penguinsame  => n,Set(IPADDR=${SHELL(asterisk -rx 'sip show settings'|awk -F: '/Externaddr/ {gsub(/[[:space:]]/,""); print $2}')});
05:36.40Penguin[Dec  2 23:34:30] WARNING[15559]: pbx.c:4250 pbx_substitute_variables_helper_full: Error in extension logic (missing '}')
05:36.43Penguin[Dec  2 23:34:30] WARNING[15559]: pbx.c:3877 func_args: Can't find trailing parenthesis for function 'SHELL(asterisk -rx 'sip show settings'|awk -F: '/Externaddr/ {gsub(/[[:space:]]/,"'?
05:36.48PenguinWhat am I doing wrong?
05:37.32WIMPyYou have to escape the ;s.
05:37.41PenguinOh.
05:37.51PenguinShall I use a backslash?
05:38.05WIMPyyes
05:39.11PenguinNice.  That fixed it.
05:41.44*** join/#asterisk suneye (~atcmmi@116.25.196.6)
05:41.44Penguinxandrix: I also found a bug in what I showed you earlier.  Put the SHELL() before the playback to avoid lag in the SayAlpha().
05:42.11PenguinI was doing Playback() and then running the SHELL(), and then the SayAlpha(), and it was lagging the first octet of my IP address.
05:42.24PenguinI switched the order and resolved that problem.
05:43.03PenguinSince all my asterisks are behind NAT, I've changed it up a bit more using sip show settings.
05:47.46WIMPyYes, a local IP announcement could have been helpfull once.
05:49.38PenguinHe's running asterisk on his pfsense router, so I just used ifconfig to get his address.  Mine, on the other hand, have private addresses and wouldn't serve much purpose from the outside.
05:51.08WIMPyIt could still help locally if you don't have a monitor connected and don't know what IP it got via DHCP.
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06:15.22PenguinThat's a possibility, although I wouldn't typically have my asterisk computer configured for DHCP.
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07:21.25yottanamiI want to restart asterisk and I have this error http://dpaste.com/1491031/
07:21.53yottanamibut the /var/run/asterisk/asterisk.ctl exists
07:22.19yottanamiin /var/log/asterisk/full I have this log http://dpaste.com/1491029/
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07:32.46kaldemaryottanami: which version of asterisk are you using? did you just upgrade? did it work before?
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07:35.27yottanamikaldemar:  I did not installed it, How can I check version
07:35.27yottanamiIt works now but I add some code that does not work
07:35.54kaldemarasterisk -V
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07:38.03yottanamikaldemar: Asterisk 1.8.20.0
07:38.55yottanamiI add #include elas.conf in the end of extentions_custom.conf and I create a elas.conf that contains http://dpaste.com/1491048/
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07:39.49kaldemaryottanami: that version has a bug that outputs the "unable to connect" when starting with the init script. it's fixed in the following version. you don't need to worry about it though as long as asterisk starts.
07:39.51yottanamiand I create /var/lib/asterisk/agi-bin/popup.php
07:41.02yottanamikaldemar: Should I do any other thing to run PHP file automatically when somebody called ?
07:41.39kaldemarextentions_custom.conf or extensions_custom.conf? and is that included from extensions.conf? do your calls land in those contexts?
07:42.10kaldemaradding stuff in some file does not automatically make asterisk use it. are you using freepbx?
07:44.39yottanamikaldemar: sorry I mean  /etc/asterisk/extensions_custom.conf
07:45.32yottanamikaldemar: in extensions.conf is #include extensions_custom.conf
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07:47.43yottanamikaldemar: I am not professional in asterisk How can I check about freepbx ?
07:47.56kaldemaryottanami: the includes need to be in the right place in your configs for them to take effect.
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07:49.17kaldemarfreepbx is a GUI for configuring asterisk. it uses a file called extensions_custom.conf, that's why i asked. asterisk does not use such a file by default. if you use freepbx to configure asterisk and want to do modifications by hand, ask in #freepbx where to put them.
07:49.37yottanamikaldemar: in extensions_custom.conf the include is in the end of file and the extensions.conf is default
07:51.12kaldemaryottanami: there really is no default in plain asterisk, so not much to say about that.
07:52.35yottanamikaldemar: Is freepbx the web interface ?
07:52.51yottanamibeause I have a web base interface on 4003 port
07:53.08kaldemaryottanami: it is A web interface. not THE web interface.
07:53.58yottanamikaldemar: God bless me for my bad English :D
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07:56.20kaldemaryottanami: it's not a part of asterisk, but a 3rd party component.
07:57.04yottanamikaldemar: So should I ask my question in freepbx channel ?
07:57.04yottanamiwhat about elastix ?
07:58.45kaldemaras far as i know, elastix uses freepbx too, so that's a "yes".
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09:17.14yottanamiHow can I debug AGI PHP?
09:18.33ChannelZagi set debug on
09:19.07ChannelZand/or turn on PHP logging in your php.ini and look at the php log if that's what is dying
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09:24.27yottanamiChannelZ: What should I do to run a PHP file when somebody called ? Is this code correct? http://dpaste.com/1491048/
09:24.58HinouHello.
09:25.30HinouIs theire some one who have a exemple of how to get Queue_Variables ?
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09:41.35ChannelZyottanami: yeah that should work (though I don't necessarily understand all of your separate contexts and the point of including them all)
09:42.34yottanamiChannelZ: Should I do any other thing to it works ?
09:42.38ChannelZ(although I should mention your PHP script needs to start with #!/bin/php  (or wherever your PHP executable is) and the script should be chmod +x
09:43.03yottanamiChannelZ: Which log file should I read to finding out is it working or no
09:43.24ChannelZ/var/log/php maybe, depends if you have file logging turned on in php.ini
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09:44.30ChannelZsee error_log in php.ini (your CLI PHP probably has its own php.ini separate from the apache one, if you're running it)
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09:44.55yottanamiChannelZ: SHould I define where is popup.php anywhere ?
09:45.02ChannelZyes
09:45.28yottanamiChannelZ: I think my php code is ok, I test it whith sample parameters and it works fine
09:45.32ChannelZunless it's in your agi-bin dir
09:46.01yottanamiChannelZ: I think it did not run after calls
09:46.01ChannelZDid you turn on AGI debug like I said earlier?
09:47.08yottanamiIt is in "/var/lib/asterisk/agi-bin/popup.php" should I define ?
09:48.03ChannelZDid you turn on AGI debug like I said earlier?
09:49.15yottanamiChannelZ: where should I set agi debug on ?
09:49.25ChannelZIs console verbose on, at level 3 or so?  (core set verbose 3)  With that and AGI debug on (agi set debug on) call your extension and pastebin what happens
09:51.11yottanamiChannelZ: dr*CLI> core set verbose 3
09:51.11yottanamiVerbosity is at least 3
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09:52.50yottanamiChannelZ: debug is on. now how can I call extension ?
09:53.00ChannelZyyeesssssss
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09:58.09Martin`wish it was possible to forward a call via apple facetime :P
09:59.04ChannelZLike Apple would ever tell anyone how THAT works...
10:01.12yottanamiChannelZ: sorry I did not get it, How can I call extension for debug now ?
10:01.55ChannelZhowever you've been calling it before
10:03.13Martin`ChannelZ: I believe it is based on sip. but I think they changed to much
10:08.34yottanamiChannelZ: You mean I called the extension ? how ? and how can I check the debug output ?
10:10.16ChannelZyottanami: presumably you've already been testing these extensions trying to figure out why your PHP doesn't work.
10:14.33yottanamiChannelZ: You mean direcyty testing php ? whith php command ?
10:16.29yottanamiChannelZ: Sorry for my poor English
10:17.05ChannelZNO.. dial the extension you're trying to get to run the PHP
10:18.16yottanamiChannelZ: Should I use dial command ? it is not available
10:22.08yottanamiChannelZ: Also there is no commands like dial answer hangup
10:23.29ChannelZDo you not have a functional system with some SIP device you're testing with?  If not, why are you writing PHP code and trying to launch it from a dialplan you don't even have connected to anything?
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10:29.41yottanamiChannelZ: I have that, here is the output http://dpaste.com/1491179/
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10:32.23yottanamiChannelZ: but there is nothing about popup.php
10:37.41ChannelZit's not being called by your dialplan then.  This is a mess, I can't help you more, I have to go to bed
10:39.38yottanamiChannelZ: can you tell me how can I set it to call from dialplan ? or where can I find out more ?
10:40.25yottanamican it be because of freepbx?
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10:50.56eloycotohi all, Anyone know where I can find .spec file for asterisk?
10:58.18WIMPyyottanami: Yes
10:58.22WIMPy~freepbx
10:58.23infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
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11:34.17jonno11Hi guys. My SIP phone won't register with Asterisk, I get a 408 timeout. Oddly, I get nothing in the logs (on verbose level 9). Any ideas where I might be going wrong?
11:41.25WIMPyThe network (settings)
11:42.42jonno11WIMPy: In what sense? I'm on a VM with a bridged network connection
11:45.08jonno11Can can ping the machine fine
11:46.29GreenlightFirewall
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12:02.37wasanzyhello my asterisk crashed and here is a crashed message: http://paste.scsys.co.uk/282661
12:02.45wasanzywhat could be the problem?
12:03.28anonymouz666it seems your box is running out of memory
12:04.31wasanzyoh ok
12:05.10XandriXthe joys of voip and people no longer complaining that i get no reception on my cell phone in the apartment i live in :D
12:05.46anonymouz666wasanzy: oom-killer is a kernel stuff that try to save your box life by killing the process is using all your memory
12:06.28jonno11Greenlight: no firewall is active
12:10.26XandriXhaha someone posted someones full Credit Card info on pastebin
12:11.39wasanzyXandriX:hmm too bad
12:12.14wasanzyanonymouz666: ok, thank you. I had a lot of memory so I wonder why it happen that way
12:12.15XandriXwasanzy: i just find it funny thats its on pastebin u find that shit online all the time but thats the laste place id expect it
12:13.30XandriXanonymouz666: oom-killer ? if you are on deb i know that sometimes cache fills up for no reason one of my machines became way more performant when i added a cron job that emptys out the cache every hour
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12:14.12anonymouz666wasanzy: could be a process that has a memory leak
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12:14.44fr0ggieAhlo asterisk jedis! It is time for me to adventure into the land of secure telephony
12:22.44wasanzyAm on CentOS
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12:32.46cuscohi
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12:55.34wasanzyasterisk-voicechanger doesn't support asterisk 11+?
12:56.24wasanzyam getting this when compiling it: http://paste.scsys.co.uk/282671
12:56.35Greenlightvoicechanger? That sounds very interesting
12:56.54WIMPyIT'S A BUILT IN FEATURE NOW.
12:56.56WIMPyoops
12:57.08GreenlightApparently makes you shout :)
12:57.37WIMPycore show function PITCH_SHIFT
12:58.24wasanzyWIMPy: that is the name?
13:00.58kelebekshi, i have two asterisk server. one of them has no trunk(asterisk-office), other one has a trunk and i can call external numbers(asterisk-pro). i want to add "asterisk-pro" as a trunk on "asterisk-office" to call external number. is it possible?
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13:01.49GreenlightWhy did you have to make me aware of PITCH_SHIFT. Now instead of doing actual work, I'm changing my dialplan so my colleagues callers get pitch shifted based on the hour of the day...
13:02.00anonymouz666WIMPy: the CONNECTEDLINE uses the PAI header to do the "reverse CAllerID". What's the idea behind changing the display in hardphones when hitting some specific IVR option? The same as CONNECTEDLINE?
13:03.20WIMPyNot sure I get that question. Are you talking about PBX phones?
13:03.55wasanzyPlease what is the name of the voice changer module in asterisk now?
13:04.44WIMPyfunction PITCH_SHIFT
13:05.08WIMPySo the module is probably called func_pitch_shift.
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13:14.34jonno11Any more suggestions on my issue?
13:14.47jonno11I'm completely stumped
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13:23.55wasanzyWIMPy: Ok and it works just like voicechanger?
13:24.29wasanzybecause Am reading about it and I didn't see any mention of changing voice or anything of that sort
13:25.24GreenlightI've got it working a treat over here
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13:25.55GreenlightAlthough I'm not sure what functioanlity your voicechanger had other than pitch shifting
13:26.23WIMPyIIRC it was just the same.
13:27.09GreenlightJust biding my time waiting for the next person to call now
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13:28.10wasanzyok
13:28.19wasanzyam checking on that
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13:50.55BeachBallyawns, i should have called in sick
13:50.59BeachBall:Þ
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13:56.40BeachBalli feel fat this morning
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14:00.04anonymouz666do some exercise
14:00.29anonymouz666WIMPy: yes, between some IVR and a SIP device.
14:01.02anonymouz666I got the idea how to update using connectedline using the SIP PAI header as CallerID.
14:01.41WIMPyYes. I always (ab)use the name.
14:01.44BeachBallbounces around
14:01.54BeachBallwell thats enough of that
14:01.57BeachBallsits back down
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14:02.56WIMPyHas anyone tried to do a calculator vie CONNECTEDLINE, btw?
14:02.59WIMPyvia
14:03.27[TK]D-Fender?
14:03.49[TK]D-Fenderas in push answers via callerid?
14:03.56WIMPyUse Asterisk via your phone as a calculator.
14:03.58PenguinYou know what makes a person feel fat?
14:04.03WIMPyOr the entered term as well.
14:04.27[TK]D-FenderWIMPy: Never heard of anyone doing such a thing... don't see why you couldn't...
14:05.03WIMPyJust thought it might be a nice little exercise.
14:05.27WIMPyIIRC LCR has it as a built in function, even.
14:06.08Penguinxandrix: Zrog?
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14:12.13WIMPyActually it doesn't use connectedline. It uses display.
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14:13.38Penguinxandrix: Still awake?
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14:38.26sabestogot internet/iptv/voip from my ISP (3 vlans), so far i have managed to swap out all the crappy equipment they sent me (fiber converter/router), but i still have to use the router for voip. Obviosly i want to use asterisk instead of the router, but my ISP will not tell me what i need to set up my own. Anyone got any tips as to what i could do to find the settings?
14:39.39sabestoi did use wireshark to analyze the traffic on the voip vlan, and it did not seem like there was any login/auth going on
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14:48.43[TK]D-Fendersabesto: What settings?  WWe have no idea what router you're talking about, and if it is in the way you're going to deactivate the VoIP functionality on it.
14:50.18sabestothe router is locked, cant get the settings from there, and it will be removed once i have another solution for voip
14:50.35[TK]D-FenderYou'll have to remove it just to validate any other solution
14:50.55[TK]D-FenderAnd you need to proper settings for your provider.  Don't expect to go sniffing user& passes out of there.
14:51.13[TK]D-FenderIf they don't want to give it to you, start shopping around for another vendor
14:51.26sabestothats the thing, i dont see it authenticating at all
14:51.33sabestois that unusual?
14:51.42[TK]D-FenderNo.
14:52.02sabestoif so, should it not be easy to sniff what i need?
14:52.05[TK]D-Fenderif you have a fixed IP they could jsut be throwing calls at your address
14:52.34sabestothe voip vlan has some internal ip, so i guess its static
14:52.41[TK]D-FenderOf course we have no confirmation of how they are supposedly doing it at all
14:53.12sabestoi will capture some data again, was half a year since i looked into it last
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14:54.18sabestomy previous provider had authentication
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14:55.21sabestonot really an option to change, this is the only ISP with fiber and USP's
14:55.25sabesto*UPS's
15:01.24[TK]D-Fenderperhaps you should remove the assumption that your internet provider has to be your voice provider...
15:01.39[TK]D-FenderOr let us know why it would be otherwise...
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15:02.24sabestoi get 40/40mbit instead of 25/25 if i use internet/iptv and voip from them
15:02.50[TK]D-FenderThat VoIP isn't free...
15:02.57[TK]D-FenderSo far still not sure it's any kind of deal
15:03.19[TK]D-FenderAnd it looks like an uphill battle from people who don't want you doing anything else with what they give you
15:03.43XandriXdown here with our provider if u get phone internet and tv they give you a good deal like 25 down 15 up for your internet but your iptv is on a seperate vlan that is set for 300mbps on that vlan
15:03.48sabestocost 5 usd/month
15:04.18[TK]D-FenderGo look at a call then and see what it's using
15:04.22sabestoyes, the 40/40 limit is for the internet only
15:04.41XandriXsabesto: yeah and i bet they allocate you more bw cuz of voip :P
15:05.08sabestoi just did, and it seems like they now register the "phone" with authorization
15:05.18[TK]D-FenderI wouldn't assume that... we have no description of the service he;s supposed to be getting from them in the first place.
15:05.19XandriXoh shit voipms does the 911 thing and i havent set it up on my system
15:05.39XandriX[TK]D-Fender: very good point there :)
15:06.05XandriX[TK]D-Fender: i know however for our iptv down here in montreal if u get an extra wireless reciever they give you an extra 5mbps on your connection to "compensate"
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15:06.59[TK]D-FenderI like my FTTH in the basement and 12/10 I get already unrestricted with a subnet....
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15:10.32XandriX[TK]D-Fender: i get 10/1 on fttn and its like 29,99 a month it was an upgrade so for a year i save 20$ a month cuz the 10 mbps baseplan is 10 bucks cheaper then the 5 i was on before and they pay my 10 dollar a month dry loop for a year
15:10.58XandriXso after a year i only save 10 bucks to get well double my speed haha
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15:13.12youjelly~media gateway
15:13.35youjellyany recommendations for a ss7 to SIP media gateway
15:15.29[TK]D-Fenderyoujelly: AudioCodes Mediant series
15:16.19youjellythanks tk
15:17.34sabesto[TK]D-Fender: well, it is using digest auth with md5 now, so i guess i cant do it
15:21.05sabestowell, there might of course be a slight chance that the password is actually blank
15:21.50XandriXread ss7 and hisses
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15:22.19XandriXim sure im not the only one here who finds that ss7 == crap :P
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15:29.30XandriXmore i read about asterisk more features i discover and more shit i see i will need to configure hehe i am loving this
15:29.54XandriXalso allowed me to better understand and debug commercial telephony systems like avaya bs and the like
15:30.13XandriX#asterisk <3
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15:32.43hfpHi guys, I have a DID where I just ring it, it catches the CallerID and calls it back without ever answering the call. I use Hangup() to send a busy tone. But I changed cellular networks recently and when I call the DID, my phone gets confused. When it reaches Hangup(), the cell just keeps ringing a busy tone and I can't hangup the line on the calling end. Is there a better way to return a busy tone without answering the channel?
15:33.38sabestohm, the "nonce" part of the authentication is static for every register the router does, nice
15:34.02[TK]D-FenderhpYou don't return a "busy tone".. If you are transmitting sound to them then you have answered the call
15:34.17[TK]D-Fenderhfp: You don't return a "busy tone".. If you are transmitting sound to them then you have answered the call
15:35.12WIMPyhpf: what technology are you receiving that call on?
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15:36.12hfpI am not answering the calls, I have never been billed for the call. When the DID rings, I have some logic that will extract the CallerID, setup a callback and then Hangup() the channel. It was never answered.
15:36.15hfpI am using VoIP
15:36.18hfpSIP
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15:36.48WIMPyOk, so the call is over then.
15:37.03hfpWould posting the extension logic help?
15:37.41WIMPyIf the caller side doesn't end the call that's got nothing to do with your Asterisk.
15:38.05WIMPyBut you could try to see if it reacts differently if you use other hangup cuases.
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15:38.48hfpThat is what I am trying, I was just wondering if there was a better practice way of handling that scenario. I thought maybe it is not right because I Hangup() a call that was never answered to begin with
15:39.17WIMPyNo. That's ok.
15:39.46hfpAnd because it takes forever for the cell to drop the call after the Hangup(), I miss the call back that goes to the voicemail...
15:40.05WIMPyHang up yourself.
15:40.27hfpThis is it, I can't
15:40.44WIMPyThen get the hangup button fixed.
15:40.59[TK]D-FenderEither your ITSP isn't passing back the abort, or your Cell carrier isn't respecting it.
15:41.21[TK]D-FenderOr what you telling us is innacurate.
15:41.24hfpIs there a way to verify which side the problem lays on?
15:41.48[TK]D-Fenderhfp: Look at the call on your side.  that is all you can prove
15:41.56WIMPycall from another phone.
15:42.11[TK]D-Fenderthat could work too..
15:42.19[TK]D-FenderI'd recommend a land-line from another carrier
15:43.16hfpI will try this. What would be the proper hangup cause for that scenario?
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15:44.39WIMPythe default 16 should be ok. You could try 21 for a reject.
15:45.01WIMPyBut there's always a chance you get some announcement.
15:48.10hfpSeems to work with 21
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15:53.06hfpAlright, thanks, I'll try with other lines and carriers to make sure 21 is fine
15:55.49PenguinHave you heard of something called Hud Lite Fonality?
15:57.12[TK]D-FenderAMI client like FOP
15:57.29[TK]D-FenderBased off a central tenet-based server
15:58.00PenguinIs it any good?  Someone asked me about upgrading FROM that to something else.
16:00.36[TK]D-FenderPenguin: Guess it depends on what they actually need and what they're comparing it to.
16:01.38Penguin"... and do some programming that integrates our current SQL Server database with an upgraded asterisk interface (Hud Lite Fonality currently used)."
16:03.21PenguinAnyone need some work?  I don't write programs.
16:05.25PenguinMaybe I'm misinterpreting the statement.  Maybe they want to hack the Hud Lite Fonality to work with their SQL.
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16:14.52igcewielingIs there a practical way to have multiple networks permit'ed for the same peer in sippeers realtime table?  I've only ever needed to allow access from one block of IP addresses so far.
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16:16.41BeachBallgently bounces
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16:23.30Synx|hm_Anyone using google voice? I am having issues getting DTMF back correctly in a call. I can dial out and send DTMF find, but getting DTMF back from an automated system is not working even though I can hear the digits in the audio path.
16:23.55Penguinigcewieling: In sip.conf, you just specify multiple permit lines.  Perhaps realtime will allow a comma separated list or something.
16:24.24igcewielingPenguin: I can think of several "perhaps" situations, I was hoping someone knew.
16:24.48PenguinIf I knew, I would have given you a definitive answer.
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16:34.41garymcAnyone know if there are differnt ways of performing auto outbound calls. At the moment when we drop a .call file into the outgoing folder our operatives dont get a call until the customer picks up the phone. Can we do this the other way around. E.g the call file gets dropped in the operators phone rings. They pick up then it start to call the customer?
16:34.43garymcThis would be so the customer is not answering the phone to a ringing sound or automated message?
16:35.25[TK]D-Fendergarymc: answered in #freepbx
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16:49.58yottanamiI add this code http://dpaste.com/1491480/ in /etc/astrisk/extensions_custom.conf and the output in call time is http://dpaste.com/1491479/ Is it running my popup.php code ? and [popup1] ext ?
16:50.45[TK]D-Fenderyottanami: do you SEE your PHP script being called in there?
16:51.36yottanami[TK]D-Fender: No I did not, but I dont know what is the output if the PHP file  called
16:52.12[TK]D-Fenderyottanami: You'd see the dialplan apps in your dialplan .. being executed.... and they clearly aren't
16:52.31[TK]D-Fenderyottanami: there is no AGI in your pastebin of the call trace provider.  Nor is there "php"
16:52.45[TK]D-Fenderyoujelly: Whatever code you've shown us is not involved in that call.
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16:52.49[TK]D-Fenderyottanami: ^
16:52.53yottanami[TK]D-Fender: Is it running http://dpaste.com/1491480/ ?? because there is some output about Goto (ext-queues,500,1)
16:53.06NerdOutSo this is the modem I have, can I use it with asterisk? http://www.ebay.com/itm/US-Robotics-3Com-56K-V-90-PCI-Fax-Modem-3CP2977-OEM-Model-0726-w-Drivers-/321043284744?pt=US_Internal_Modem_Card&hash=item4abfabc308
16:53.40[TK]D-Fenderyottanami: Stop pointing to that code sample.  We SEE the output.  NONE of the commands in your code are being executed there at all.  We can all see that.
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16:53.43NerdOutI'm just doing this for some nerdy boredom, so if I can't I can't..just figured it would be neat to answer my house phone from a PC.
16:54.03[TK]D-FenderyooNNO AGI is being called.  At all..  A text search on the pastebin page shows that.  It isn't even a question....
16:54.03navaismouse a free softphone
16:54.17NerdOutOr smartphone, etc.
16:54.36navaismoyes
16:54.40NerdOutWell, the problem with a softphone and soft provider, they don't answer my house POTS line that people will be calling.
16:55.14navaismouse a gateway
16:55.18NerdOutIf the you thing the modem will work, asterisk could answer the POTS line and ring some softphones.
16:55.35WIMPyPort the number.
16:56.24NerdOutAre these just suggestions for better ways or a hint that the modem is useless?
16:56.41navaismoNerdOut, only the old modems with Tiger chipset are recognized by DAHDI as x100p, but for the sanity of your calls and your mind(and us) dont use that.
16:57.00WIMPyAsterisk hasn't supported modems in more than 5 years.
16:57.06yottanami[TK]D-Fender: What should I do to find out what is wrong ?
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16:58.48NerdOutAnd I guess there is no such thing as a Poor Man's PBX, which uses modems instead of costle gateways.
16:58.55NerdOut*costly
16:59.16WIMPyGet digital and somethign exists.
16:59.45WIMPyIt's just the analogue stuff that's expensive.
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17:00.39NerdOutMy internet is shit so I don't really want to rely on a VOIP provider. But, this was a boredom project anyway, guess I will just have to keep walking to the phone :D
17:02.04navaismoor buy a chinese-crap x100p on ebay
17:02.09WIMPyFor some of us there's ISDN. Reliable and cheap.
17:02.56navaismoyeah first world
17:02.58navaismo:'(
17:04.10[TK]D-Fenderyottanami: You showed an IVR processing a call.. nothing in your dialplan code there is used by that ivr natively.  There is no reason to expect any of your code to be looked at at all.
17:04.36[TK]D-Fenderyottanami: and that is FreePBX you are integrating with and is not something supported here.
17:04.41[TK]D-Fender~freepbx
17:04.41infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
17:05.01[TK]D-Fenderyottanami: But you contexts are NOT magically used by an IVR based on what you've written
17:05.32NerdOutSo to the end goal of, people call my POTS line, it rings my house phones and asterisk, asterisk rings my computer/iphone, I can answer from any of them, and outgoing calls from my softphones go out the POTS line, what would I need? TLDR, what would I need to turn asterisk backed softphones into additional handsets?
17:06.28[TK]D-FenderNerdOut: Linksys SPA-3102 < $70 USD and lets you take in your home line and also lets you plug in an analog phone, each functioning independent of the other
17:06.33WIMPySome piece of hardware to connect to your line.
17:06.52WIMPyEither as PCI card or as an external appliance via VOIP.
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17:08.59NerdOutWIMPy: Any specific recommendation of hardware? PCI card sounds good, fits into my box instead of taking up more plugs.
17:09.05nextimeis there any chance to get sipml5 or jssip to work with asterisk decently without using webrtc2sip?
17:09.26navaismonextime, yes
17:09.34navaismoprepar your asterisk to work with webrtc
17:09.43navaismouse latest 11 version
17:09.44nextimenavaismo : already done
17:09.47nextimei'm using 11.6
17:09.52navaismoand its done
17:09.52nextimeit work with webrtc
17:09.55nextimebut no audio.
17:10.07navaismocheck your ICE settings and Nat Settings
17:10.26nextimenavaismo : already done and re-done multiple times
17:10.36navaismoso rephrase your question please
17:11.21nextimenavaismo : basically i can't get audio working on a setup of asterisk with sipml5/jssip, not even with just calling a test extension on the asterisk side with just an Echo()
17:11.24navaismoand show the configs, and the call and the debug from asterisk and the used API
17:11.28NerdOutActually, on another note, I'm looking to get a Raspberry Pi(http://www.raspberry-asterisk.org/) for Christmas, so any USB devices for this purpose?
17:12.05navaismoNerdOut, no rapberry pi doent have enough power for USB, so use a SIP gateway
17:12.08nextimelet me write all things
17:12.40navaismoNerdOut, and the only usb-fxo is not supported by arm platforms
17:12.48*** join/#asterisk pointer (~Any@aj.catt.com)
17:12.59navaismostupid sock structure
17:15.04NerdOutOK, so back to PCI cards then.
17:16.08navaismoraspberry do not have pci
17:16.44navaismoso better use gateway and you can use it with your PC or RPi
17:17.17NerdOutGateways aren't exactly cheap though( I don't expect PCI cards to be, but I was hoping for cheaper)
17:18.24navaismoonly Chinese cards are cheap and you get waht you buy so...
17:19.08navaismohttp://www.ebay.com/itm/Linksys-Cisco-SPA3102-Voice-Gateway-with-Router-SPA3102-NA-NEW-/121224688677?pt=US_VoIP_Phone_Adapters&hash=item1c398df425
17:19.57[TK]D-Fender[12:06][TK]D-FenderNerdOut: Linksys SPA-3102 < $70 USD and lets you take in your home line and also lets you plug in an analog phone, each functioning independent of the other
17:20.02*** part/#asterisk yottanami (~yottanami@5.237.113.75)
17:20.04[TK]D-FenderNot like I didn't say it already...
17:20.15navaismoThe Obihai is cheaper, I never used it but you may want to check
17:21.10navaismohttp://www.obihai.com/matrix
17:22.45*** part/#asterisk Blashyrkh (~michaeld@043-054-094-081.as39912.net)
17:23.32NerdOuthttp://amzn.to/1dNKRvr This would be the Obi?
17:24.07[TK]D-FenderNerdOut: Clearly not.
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17:24.53nextimethis is my asterisk config
17:24.54NerdOutAh, my mistake, 110, not 100.
17:24.55nextimehttp://danex.nexlab.it/asterisk-webrtc.txt
17:25.09nextimeand this the call trace: http://danex.nexlab.it/full.txt
17:27.16navaismoforbidden ^
17:28.09nextimeuhm?
17:28.27NerdOuthttp://amzn.to/1hvYQXg So from my rather new understanding of this, the Obi110 would bridge my POTS line to "a VoIP service", which would be Asterisk. And through much configuration, I could turn my PCs and smartphone apps into replacements for my cordless handsets that recently died.
17:28.51nextimea ok
17:28.54nextimethe full.txt
17:28.56nextimenow it work
17:29.21WIMPyNerdOut: Yes, but you should be aware, that wifi is not good for realtime voice.
17:29.28WIMPyUnless you don;t have neighbours.
17:29.44[TK]D-FenderNerdOut: There is very little documentation on the OBi for setup and I've never heard anyone use one for PSTN before....
17:30.46NerdOutWell, I do occasionally take calls over Google Voice and Talkatone, which is a VoIP service, and the quality is good enough for me over my WiFi. (Using a NETGEAR R6300)
17:31.01NerdOutSo I don't think the WiFi will be an issue
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17:32.58navaismonextime, is this ok?---><--- Transmitting (no NAT) to 87.0.155.191:5060 --->
17:33.33navaismoyour sipml5 client is on the public IP 87.0.155.191?
17:33.39nextimenavaismo this the public ip address of the client side, using stun
17:33.53nextimeif i don't use stun it try to trasmit to my private ip
17:34.27navaismoyes yes, i just want to know if that client is inside or outside the pbxx network
17:34.35nextimeoutside
17:34.53nextimethe asterisk is on a OVH server, the client is on my home adsl connection
17:35.15*** join/#asterisk endemic (~endemic@lynx.onvox.net)
17:36.30navaismonat =  force_rport,comedia  didn't help? Your router has disabled sip alg stuff?
17:37.43nextimenavaismo : the router is a linux machine, nothing special and it work on "normal" sip calls and other webrtc tests, nat is already nat=comedia,force_rport but i have tested it also with no nat helper
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17:44.04navaismonextime, did you installed the uuid-devel for ice support?
17:45.05navaismook forget that
17:45.22navaismoi already saw the ice-ufrag, ice-pwd
17:46.07navaismobut the rtp debug dont say (VIA ICE) like: Sent RTP packet to      10.0.1.102:55518 (via ICE) (type 08, seq 023106, ts 088800, len 4294967284)
17:46.25nextimeuhmm
17:46.33nextimetrue
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17:59.34NerdOutSo this would suggest that it at least works. http://tech.iprock.com/?p=3208
18:01.22igcewielingIs there a practical way to have multiple networks permit'ed for the same peer in sippeers realtime table?  I've only ever needed to allow access from one block of IP addresses so far.
18:02.25navaismoigcewieling, can i answer that just assuming?
18:02.53igcewielingnavaismo: you can assume no.
18:03.01igcewieling8-|
18:03.40navaismoNerdOut, yes but if you have questions you can ask to the obihai dudes, they can answer you without problem also you can contact them via twitter
18:04.50navaismoigcewieling, have your tried with separated values by ; in the DB
18:05.01navaismoand yes im guessing
18:05.10navaismobased on my codecs fields
18:05.15igcewielingah, like the codecs
18:06.34igcewielingthanks navaismo and Penguin
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18:41.21JamKoGreetings. I recently upgraded to asterisk 11.6 from 1.8.. This version is ignoring config changes to the udptl.conf file, and sets default. I see no errors on startup pointing to an issue with it.
18:41.29JamKoAnyone seen an issue with this?
18:42.44PenguinIt seems familiar.
18:42.55*** join/#asterisk u0m3_ (~u0m3@92.80.74.24)
18:44.12PenguinOh.  It seems familiar because you were asking about it before.
18:45.10*** join/#asterisk trystero (~al.f.zero@unaffiliated/transfinite)
18:48.37pabelangerJamKo: read CHANGES, I think something about udptl changed
18:48.40pabelangerIIRC
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19:26.06volga629Hello Everyone, I am trying optimize asterisk  for use on vmware and those option in kernel clock=pit nosmp noapic nolapic ans other suggestions ?
19:26.49Chainsawvolga629: ESXi/vSphere or Workstation?
19:26.55volga629esxi
19:27.00Chainsawvolga629: Because certainly for ESXi that seems to be dumbing it down *way* too much.
19:27.43volga629so it should handle without it ?
19:28.56Chainsawvolga629: It certainly does where I use it.
19:29.17Chainsawvolga629: The main thing is to set the kernel virtualisation awareness up for vmware. If you use GrSec/Pax, that also has a vmware preset for you.
19:30.15ChainsawCONFIG_VMWARE_VMCI_VSOCKETS=y CONFIG_VMWARE_BALLOON=y CONFIG_VMWARE_VMCI=y CONFIG_VMWARE_PVSCSI=y CONFIG_GRKERNSEC_CONFIG_VIRT_VMWARE=y
19:30.49volga629this kernel config right ?
19:30.55Chainsawvolga629: Yeah.
19:31.17Chainsawvolga629: Unless you're dealing with some sort of horrendous vendor kernel whose config you cannot tweak, I cannot see how that nosmp/noapic madness is justified.
19:32.08volga629which kernel it is ?
19:32.28Chainsawvolga629: Mine? Linux green 3.10.1-hardened-r1 #1 Fri Nov 8 09:22:29 GMT 2013 x86_64 AMD Opteron(TM) Processor 6238 AuthenticAMD GNU/Linux
19:32.51Chainsawvolga629: ESXi 5.5
19:33.24volga629<PROTECTED>
19:33.36ChainsawRedHat Expensive Linux. Quite.
19:35.01volga629yes RHEL  need check menu config if I have those options
19:35.48volga629from where you got those options ?
19:39.17volga629it  3.10 kernel and I am using 3.2
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19:40.01ChainsawI have one or two 3.2 kernels around for those annoying FusionIO boards.
19:40.03ChainsawBut those aren't VMs.
19:40.33ChainsawI suppose it's more of a RedHat question, volga629. Is there no #redhat, #rhel or equivalent where you could ask this?
19:42.03volga629I know 100 %  that no kind options on rhel kernel, I just was wondering which one you using and right now I found that from kernel 3.9 those options was added
19:44.08volga629Fedora kernel have those options config-3.11.9-200.fc19.x86_64
19:51.19volga629ok tnk for info
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19:55.32*** join/#asterisk Flametail (481748b8@gateway/web/freenode/ip.72.23.72.184)
19:55.59FlametailIs this the fabled X100p supported by asterisk? http://www.x100p.com/products/FXO.php
19:56.21[TK]D-FenderflNo, that is just one of the more popular clones
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19:57.57FlametailSo what would be the best one to get and where to get it?
19:58.10[TK]D-FenderFlametail: None from anywhere.
19:58.30[TK]D-FenderFlametail: What are you actually looking to do?
19:58.32FlametailAll additional gateway hardware these days huh?
19:59.00[TK]D-FenderNo, just not "crap based off a flakey winmodem"
19:59.36*** join/#asterisk jsjc (~Adium@192.Red-79-151-33.dynamicIP.rima-tde.net)
19:59.58FlametailWell, when you've got a "flakey win"-box running asteriskwin32, you look at the hardware the webpages says it has support for.
20:00.14[TK]D-FenderFlametail: You are compounding shit with crap there
20:00.52[TK]D-FenderFlametail: It is difficult, but not impossible to put into words just how much bile is generated here at the mere mention of asteriskwin32
20:00.53navaismothought that asteriskwin32 was a myth
20:01.20FlametailServer 2003 is currently the only supported way to run an Active Directory domain from a single box :) Samba(4?) is not stable enough for me yet.
20:02.00[TK]D-FenderFlametail: No excuse for running Asterisk under windows.  Get a proper box for it or virtualize it sanely at least
20:02.49FlametailAs to your original question, I am setting up the asterisk to be a sort of rooms intercom based on iPad Mini's I have laying around..call between rooms. Figured I would at lease look at the card needed to take it that extra step and turn them into full phones.
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20:03.58[TK]D-Fenderthe thought of trying to gt HARDWARE support on that platform is also insane
20:04.10FlametailWhat would be the minimum specs for a virtual linux running a proper asterisk setup? I have embarassingly low quality hardware available to me..I play with people's refused old PCs.
20:04.14[TK]D-Fenderuse a SIP gateway of some sort like a Linksys SPA-3102, etc
20:04.59Flametailhttp://asteriskwin32.com/hardware
20:06.36newtonrFlametail, http://www.voip-info.org/wiki/view/Asterisk+hardware+recommendations has some old example use-cases with hardware specs for most of them.
20:06.40[TK]D-FenderBest of luck with that...
20:06.51[TK]D-FenderYou are piling junk on junk.
20:06.58[TK]D-Fenderthose cards aren't recommended for any use
20:08.31navaismoloves the asteriskwin32 page with the old html style!
20:09.25Flametailnavaismo: It is actually supposed to look like this, but with the frames style nav I couldn't link directly to the page. http://asteriskwin32.com/
20:09.28navaismoLOL Version 0.66 compiled from version 1.2
20:10.39newtonrFlametail, I also highly recommend not using asteriskwin32, as if you run into any issues it will be difficult to get help. If you run into a bug, know that there is no official support for Asterisk on windows, not even considering the fact that it uses an old 1.2 version of Asterisk that isn't supported.  Therefore investing in hardware to work specifically with an ancient version of Asterisk on an unsupported platform is unwise.
20:12.16newtonrhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for reference
20:12.21Flametailnewtonr: Then I will just go with my original intercom plan, and save out/in calling for when someone dumps newer hardware on me(or just any more hardware, since the sysreqs of asterisk are not that steep.) And when I invest in a gateway/
20:14.31newtonrProbably a good idea.
20:16.02JamKoPenguin: Thanks I'll take a look.
20:16.37navaismow00t!! the support plan for 1 server is very affordable --> $199/year
20:17.14drmessanoor you can just use an ITSP and forget analog crap
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20:18.16drmessanoapologizes somewhat to those in the business producing/supporting cards for crap analog lines.
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20:27.32nextimetired to fight
20:27.36FlametailStupid question I'm sure, but what is the voicemail call in number in a basic Asterisk system like this?
20:27.57nextimei will probably search an alternative rtp proxy to put in front of asterisk for webrtc
20:28.15nextimesomething different from the crap that is webrtc2sip
20:28.44navaismowhy ?
20:28.55navaismoYou dont need a proxy at all
20:29.10navaismoand so far the webrtc2sip is designed to work only with sipml5
20:29.37nextimenavaismo : well, i need to make it work with sipml5 ( or jssip if the first doesn't work )
20:29.40navaismoFlametail, what ever you set in your dialplan
20:29.46nextimebut webrtc2sip isn't so good
20:29.56nextimeit require an old version of libavcodec and so on
20:30.12nextimeit's crap for me
20:30.24nextimei would prefer to run only asterisk with nothing in front of it
20:30.30nextimebut i can't make it work actually
20:30.30navaismothat is a Point of view you can ask to the people using it in the discuss group and many will tell you the opposite
20:30.39*** part/#asterisk SuperNull (~NigggaaaP@24-148-101-238.ip.mhcable.com)
20:31.04nextimenavaismo : of course it's my point of view
20:31.10navaismoso switch to Jssip and configure propelry both the api and asterisk
20:31.29navaismothere are many people using JSSIP and asterisk successfully
20:31.31nextimethe fact is that i already have a recent ffmpeg that i need to have on those machines, i can't mix multiple version of ffmpeg just for webrtc2sip
20:31.40navaismodont use it
20:31.44nextimenavaismo : with jssipo i have the very same issue
20:31.48navaismoand dont use sipml5
20:32.00nextimenavaismo : sadly jssip gave me the same issue
20:32.22navaismoso again maybe the issue is in your settings and not in the api
20:32.46navaismohave your tried testing the jssip with the demo page?
20:32.57nextimenavaismo : yes, probably you right. But anyway, i don't really know what to try, it's 2 days that i'm fightinhg with this
20:33.04nextimenavaismo : yes.
20:33.05navaismohttp://tryit.jssip.net/
20:33.17nextimenavaismo : i'm using the tryit demo on my own server too
20:34.20navaismoYou should start to test locally, then migrate the stuff to the cloud and so on. Normally with the no audio and webrtc the issue are bad nat settings
20:34.58navaismoor not ice support i ran on that issue more than 10 times in the last semester.
20:35.04navaismoand all was my fault
20:35.11navaismoNAT or ICE
20:35.15nextimenavaismo : well, as it work on other online tests and "normal" sip work on the same server
20:35.17navaismobut always my config
20:35.21nextimei'm pretty sure isn't a nat issue
20:35.30nextimebut anyway
20:35.57nextimeif nat give so many issues isn't a great thing if i need to make people use it where i don't have control of they're nat settings
20:37.12navaismoright, there is no such thing as "turn key" for development apis or protocols
20:38.02nextimenavaismo : well, at least if i know that "chrome version 31" work
20:38.12nextimei can say to all "use only chrome 31"
20:38.18nextimeit's ok this way
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20:41.29navaismoI have been working with chrome since 28 now is 32 & it works at least locally
20:42.15nextimenavaismo : ok, i can confirm that locally it work
20:42.32nextime( via vpn i can call the server with no nat )
20:42.55nextimeso, it's a nat/ice/stun/something issue
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20:44.05navaismoyep. Try with jssip and check the debug from jssip too, look for the audio address in the SDP exchange and see if match
20:44.19navaismowith the asterisk rtp/sip debug
20:44.44navaismoalso look for ice stun errors
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20:45.18nextime( audoi work, video not with a grandstream phone, but probably it's cause of h263/vp8, no matter anout video for the moment )
20:45.32nextimei\ll re-try with jssip
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20:51.08nextimewell, jssip isn't working well even in local, one way audio
20:51.48navaismocheck if it is using the public ip instead the local
20:53.49adeelnI'm trying to use the Transfer() function and am passing it the 'j' option to jump to n+101, yet when i test it; it doesn't jump...what am i missing something?
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20:56.48navaismoIn Asterisk 12, is valid in the contact field for the AOR to have more than sip:name@ip:port? Like sip:name@ip:port;option1;option2
20:59.52saliakI have a remote SIP phone that's behind a NAT.  audio that goes external (if someone calls in or they call out) is 2 way.  if i call them internally, it's one way audio
21:00.19saliakwhat would cause that?
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21:05.51Weezeyhave a weird problem, one asterisk box won't talk to another.  They're both type=friend and have their host as the IP address of the other
21:07.56[TK]D-FenderShow the actual call...
21:09.14Weezeyit's a port number problem.
21:09.36Weezeyi just did insecure=port,invite and the call worked
21:09.49Weezeythe call was coming in as :1024
21:12.00[TK]D-Fenderthen remove "invite"
21:12.23Weezeydone
21:12.25Weezeyworks too
21:12.44Weezeydamn nat.
21:13.41Weezeybiggest problem with cloud hosting this stuff is that nat layer
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21:14.49saliakThis is the sip debug from my call where I call the other extension.  Answers and I can hear "hello", but then all audio ceases to exist.  http://pastebin.com/TN0xxCEm
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21:21.40[TK]D-FenderCheckout time, BBIAB
21:25.04saliakI see, it's using the local IP of my remote peer.. i have nat=yes in my sip.conf, isn't that supposed to make the public ip the one that it uses?
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22:03.24nextimeok after some investigations
22:03.43nextimeif i remove ice/stun and try to use only common nat helper
22:04.03nextimesipml5 send in sdp my private ip as rtp endpoint, as expected
22:04.21nextimebut asterisk should use the external address as it has nat=force_rport,comedia
22:04.41nextimebut instead it doesn't use the extern ip of my client, but uses the internal one as endpoint for rtp
22:04.45nextimethis cause my issue
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22:10.39Synx|hm_Need a little help getting an outbound flowroute.com sip peer setup. I have the config, i am able to register the peer and i have a dial plan setup and working in extensions.conf, however i keep getting 484 returns. And on the actual sip messages i still see 1800xxxxxxx@mylocaldomain in the TO field, should asterisk strip that out before it goes out the WAN?
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22:16.07[TK]D-FenderSynx|hm_: * strips what you tell it to strip....
22:16.45[TK]D-FenderSynx|hm_: pastebin the complete call attempt with SIP debug enabled.
22:16.47[TK]D-Fender~pb
22:16.47infobotA "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.
22:19.39Synx|hm_[TK]D-Fender: doing it now, not sure if there is something else going on here, as i have only ever used googlevoice with this asterisk box
22:20.42Synx|hm_[TK]D-Fender: http://pastebin.com/kSru7Mqs
22:22.10[TK]D-FenderSynx|hm_: That is * telling your phone that it does like what it dialed....
22:22.49[TK]D-FenderSynx|hm_: That has nothing to do with Flowroute.... the call coming in isn't accepted let alone apporaching going out.
22:23.38[TK]D-FenderSynx|hm_: Time to debug your dialplan
22:23.56Synx|hm_[TK]D-Fender: hrm i keep seeing options pings in there so i assumed it was 216.115.69.144 rejecting (flowroute)
22:24.07Synx|hm_so thats actually not even making it out the wan
22:24.48Synx|hm_[TK]D-Fender: my dial plan is right out of the flowroute example... exten => _NXXNXXXXXXX,1,Dial(SIP/1${EXTEN}@flowroute)
22:24.51Synx|hm_exten => _1NXXNXXXXXXX,1,Dial(SIP/${EXTEN}@flowroute)
22:25.30[TK]D-FenderSynx|hm_: Bad assumption especially since you can SEE who is getting refused there
22:25.37[TK]D-FenderSynx|hm_: It isn't in doubt.
22:25.44[TK]D-FenderSynx|hm_: And your dialplan is wrong....
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22:26.15[TK]D-FenderSynx|hm_: You should pay much closer attention to it...
22:27.00Synx|hm_[TK]D-Fender: damn it yup there it is and its obvious
22:27.13Synx|hm_[TK]D-Fender: thats what i get for reading that in the console with all the other station registrations going by
22:27.16[TK]D-FenderSynx|hm_: pretty much :)
22:27.38Synx|hm_[TK]D-Fender: well poop now to figure out whats wrong with that dialplan
22:27.58Synx|hm_it seems fairly streight forward but then again its probably something obvious i dont understand there
22:28.08[TK]D-FenderSynx|hm_: Sounded like you had just found it.....
22:28.29Synx|hm_no i ment it was obvious that asterisk was 484ing and not the carrier
22:28.47[TK]D-FenderSynx|hm_: Well stare at your dilaplan some more...
22:29.49Synx|hm_[TK]D-Fender: this is a bit new to me sorry, is it the @flowroute? i assume that was in ref to [flowroute] in sip.conf
22:30.36[TK]D-FenderSynx|hm_: that is part of the Dial... but the dial is not happening because that line is NOT getting used
22:32.00[TK]D-FenderSynx|hm_: None of the4m match.  The 484 is telling you that straight-up
22:33.57Synx|hm_*FACEPALM*
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22:34.44Synx|hm_you know whats really stupid about that... i said you know let me make sure there are note too many match points let me compare it to the first one, they both had one too man
22:34.47Synx|hm_many
22:35.30Synx|hm_[TK]D-Fender: please forgive my stupidness and lack of attention to detail, thanks for letting me figure it out on my own with a few hints, DOH
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22:36.17[TK]D-FenderSynx|hm_: So you actually have it this time ... right?
22:36.38Synx|hm_ya 11 digit dial plan ;)
22:37.20[TK]D-FenderMATH .. it ain't just a suggestion
22:38.39Synx|hm_ya i should have figured when i had to start using my toes to count the x's that it was too man ;)
22:38.48Synx|hm_s/man/many
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22:56.04Synx|hm_[TK]D-Fender: well all of that work setting up a new outbound SIP route to bypass googlevoice as i thought it was the source of my DTMF issues, nope still same problem :(
22:56.35jonno11Hmm how can I debug not having any audio on a mobile call?
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22:57.14FlametailI was researching using GV for my external calling, and I came accross this: https://plus.google.com/+NikhylSinghal/posts/MjyncJEbzxK
22:57.28FlametailWill asterisk be affected by this loss of XMPP functionality?
22:57.31[TK]D-Fenderjonno11: Step 1: properly identify what this "mobile call" is, because that is not a "thing" as worded.
22:57.48jonno11[TK]D-Fender: Sorry I realised as I hit enter how unhelpful that was haha
22:58.22[TK]D-FenderFlametail: When XMPP goes, so oes GV which is moving to hangouts
22:58.34[TK]D-Fenderjonno11: And that's only one side of the cal.
22:59.18FlametailI've look at Hangouts, to me it's just a replacement Chat. Surely the new protocol can be learned or Google will offer the asterisk devs an alternative?
22:59.51navaismoFlametail, GV is dead for us period. Use an ITSP there are many cheap ITSP outside
23:00.47FlametailYeah, but Cheap vs Free...damn corporations luring us in with freebies and then rubbing it in how it is up to them to control the "free" world.
23:00.59jonno11[TK]D-Fender: bit of a brain dump. Basically I have a sip client on my iPhone. I can't hear any audio on the phone, but the microphone is working fine. Things are working fine on my laptop (running Telephone).
23:01.10[TK]D-FenderFlametail: Nothing warms out hearts like blind optimism.....
23:01.14[TK]D-Fenderour*
23:01.18FlametailGoogle at least was good...everyone else will probably be "you get what you pay for"
23:01.30navaismoFlametail, there is no such thing as free for PSTN
23:01.43jonno11Audio's fine both ways on the laptop, only the microphone is working on the mobile SIP
23:02.10[TK]D-Fenderjonno11: No idea where your * server is, nor your laptop.
23:03.10jonno11[TK]D-Fender: You only have to ask... Asterisk is on a VM on my local network (bridged). Both sip clients are accessing via wifi.
23:03.11[TK]D-Fenderjonno11: Go look at the call with SIP DEBUG enabled and give a proper description of the bits involved
23:03.28navaismois not sure about people changing their nicknames and enter to ask the same question about free pstn...
23:03.42FlametailSucks because GV is screwing everyone over. This is going to kill Talkatone and stuff too.
23:04.09navaismoand wondering what type of info is out there that in the past month many guys came to ask for free pstn sevices ?¿?¿?¿
23:04.29jonno11[TK]D-Fender: How do I enable that? `sip debug` is not a command.
23:05.01[TK]D-Fender"sip set debug on"
23:06.09Flametailnavaismo: This "GV and Free VoIP are dead in 6 months" stuff started 10/31...so that explains your month of alternative seekers.
23:07.05navaismosince GV only works for USA, I never see that as a real alternative.
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23:09.07jonno11[TK]D-Fender: http://pastebin.com/bvaJz528
23:09.08Flametailnavaismo: Well, for a lot of people, USA and Canada are all we ever need :D
23:11.44[TK]D-Fenderjonno11: Ok, everything loks like local LAN...
23:11.54jonno11[TK]D-Fender: it is :)
23:11.57[TK]D-Fenderjonno11: verify your firewall isn't in the way.
23:12.03[TK]D-Fenderjonno11: "iptables --list"
23:12.14jonno11[TK]D-Fender: ip tables isn't active
23:12.28[TK]D-Fenderjonno11: verify that.
23:13.25[TK]D-Fenderjonno11: And well.. we don't get to see it answer at all...
23:13.34navaismoFlametail, seems like is time to buy a real service
23:14.05jonno11[TK]D-Fender: What do you mean?
23:14.43[TK]D-Fenderjonno11: You talk about "no audio"... well we only see it reach "ringing" not "answer"
23:14.50[TK]D-Fenderjonnthat is not a COMPLETE call
23:15.15jonno11[TK]D-Fender: Hmm well I physically answer
23:15.27jonno112 secs I'll try again
23:15.33[TK]D-Fenderwell we don't see it answer. or the call end.
23:15.52jonno11[TK]D-Fender: I copied the log after it answered
23:16.13[TK]D-Fenderwe don't see an answer.  Or an end
23:16.17[TK]D-FenderGo get a complete call
23:16.19jonno11[TK]D-Fender: Before the call ended - reckoned the call ending wasn't important
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23:21.25FlametailIn all of your experienced opinions with this stuff, who would you recommend as the cheapest but still usable replacement for a VoIP service like GV?
23:22.03jonno11[TK]D-Fender: Okay right. Just made progress. Basically I'm trying to configure ConfBridge. Before, I made a call to the conference line, worked fine on my mac. Then, dialled in with the phone, and the same audio issue occurred. From then on, calling any extension replicated the problem. It took a restart, which I did accidentally to reset it. However the confbridge still creates the problem again.
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23:25.58navaismoFlametail, you better check many ITSP and compare their services here is a list
23:26.06navaismo~itspus-list
23:26.12navaismo~itsp
23:26.13infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.
23:26.24navaismo~itsplist-us
23:26.25infobotHere are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com
23:26.28navaismo^
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23:32.45jonno11[TK]D-Fender: http://pastebin.com/MhWHqnuK this is the logs. No audio!
23:37.40XandriX[TK]D-Fender: quick q uestion for ya if i wer to use the playback function to read a message if you dial an extension if the text its suposed to read is for say current forecast is will it say that or will it say c u r r e n t f o r e etc etc ? if say i use no predefined sounds to do so
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23:49.21jonno11Going to ask this question again... Basically I'm trying to configure ConfBridge. I can dial in and hear things fine on my mac. When I use a sip client on my phone, I get no audio, but micrp
23:49.30jonno11microphone works fine*. I'm on a LAN
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23:50.39Synx|hm_Need some help with a DTMF issue, i can send DTMF outbound just fine, its receiving dtmf back that is not working at all, i can hear the dtmf in the return audio but cannot figure out why asterisk is not picking it up and forwarding via rfc2388
23:51.01ChannelZDoes the audio otherwise work, just not in the conf?  (IE if you do a Playback of something do you hear it, and have you done an Echo test?)
23:51.09ChannelZSynx|hm_: where is the call coming from that you hear it?
23:51.36Synx|hm_ChannelZ: i dial out to an IVR and the IVR responds back with some DTMF at a few points
23:51.55Synx|hm_ChannelZ: its not carrier related as this is now happening on both xmpp googlevoice and sip carrier calls
23:52.42atanAnyone in here use a Cisco phone with a bluetooth headset? Does it work as well as your standard corded headset or is it more trouble than it's worth?
23:52.45ChannelZguess I don't follow.
23:53.35Synx|hm_ChannelZ: sorry its a bit odd, so heres the call flow, i dial out into an IVR and use DTMF to navigate the IVR (this works fine) the IVR then routes my call to an automated agent that sends DTMF digits back down the line, its the return DTMF path that is not working
23:54.44ChannelZbut what do you mean by "not working"?  What is supposed to happen?  Do you think it's supposed to start navigating your own dialplan or.. I don't get it
23:55.56Synx|hm_ChannelZ: i have an auto mated test suite that is placeing the call, it is listening for DTMF to come back, it never gets anything
23:55.59Synx|hm_automated
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