IRC log for #asterisk on 20091105

00:00.01*** join/#asterisk darkdrgn2k (n=darkdrgn@bas2-toronto44-1176436941.dsl.bell.ca)
00:00.18[TK]D-Fenderryduh: exten => s,n,Set(DIGITS="") <- bad.  No quotes
00:00.24darkdrgn2khey any one here know how to jtag a vt1005v
00:00.34[TK]D-Fenderryduh: exten => s,n(play),Background(${DIGITS:1}) <_ do NOT actually play anything in here
00:00.35ryduh[TK]D-Fender: ok. thanks
00:00.54ryduh[TK]D-Fender: Ok I got it now.
00:00.56[TK]D-Fenderryduh: 16/17 NO action in the macro
00:01.13[TK]D-FenderryYou do not "escape" it, and you do not make IVR's out of macro's
00:01.44ryduh[TK]D-Fender: ok
00:01.44jayteethere's no escape
00:02.44Kattyhugs jaytee
00:03.02jayteehugs Katty
00:04.07benklopi've got two asterisk boxes, one i set up a while ago and it connects fine to sipgate - http://pastebin.com/m415fc10b - the other i set up recently and it doesn't - http://pastebin.com/m415fc10b
00:04.28benklopi've checked everything I can think of, i can't figure out why one works and the other doesn't
00:04.53benklopdang, gave ythe wrong link for the second, http://pastebin.com/m333daa47
00:07.03benklopnon-working box has the call-id as blahblah@127.0.0.1, rather than it's local IP. I can't figure out why - and this is the only difference I can spot
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00:10.40ryduh[TK]D-Fender: yahoooooooo! thank you.
00:11.09[TK]D-Fenderryduh: You're welcome
00:13.18benklopmy dog just accidentally tuned on my vacuum.. that was frakin hilarious!
00:14.25ryduhbenklop: is your dog deathly afraid of the vacuum like my dog is?
00:14.54benklopoh yes, but he's also a puppy so he got incredibly excited too
00:15.56benklopran into a half closed door
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00:22.28drmessanoMy dog wasnt afraid the of the vacuum until he tried to mate with the business end of it
00:23.32benklophahahaaa.. that story wins...
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00:33.24trebaumdoes anyone here know specifics about the software echo cancellation modules in asterisk?
00:35.32pathI only know oslec
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00:55.53securevoipif one has as analog line into * and an analog station, is it possible to have flash sent back to telco in order to switch to call waiting?
00:56.46[TK]D-Fendersecurevoip: Depends on how the analog line is brought into *
00:56.58securevoipdahdi channel
00:57.40[TK]D-Fendersecurevoip: features.conf dynamic feature + app_flash
00:57.54[TK]D-Fendersecurevoip: I believe there is even a sample on the WIKI
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00:59.59[TK]D-Fender~gsmbug
00:59.59infobot[~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read :  http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39
01:01.24securevoipasterisk.wikia.com?
01:04.24[TK]D-Fender~wikis
01:04.25infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
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01:15.13securevoipthanks, [TK]D-Fender.  now, the wife "owes" me.  ;-)
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01:25.14Qwellsecurevoip: sounds like she "owes" him..
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01:43.25jplohcan anyone help me with asterisk 1.4.26.2 and sipsak? i can't get it to authenticate
01:44.39jplohthese are the parameters sipsak -s sip:test@192.168.100.40:5060 -u testuser -a 123456 -v -U
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02:02.00Daejeo1hello guys
02:02.19Daejeo1anyone awake?
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02:12.46darkdrgn2kany one know where i can get a firmware for  a VT1005?
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02:18.07RichGoldany one use 1.6 in production?
02:19.04NuggetI do.
02:20.19RichGoldhow many calls
02:20.35Nugget*shrug*
02:20.47Nuggetfew hundred a day I guess
02:21.01RichGoldno I mean simultenius
02:21.05RichGoldhow many endpoints
02:21.18Nuggetdozen
02:37.36Holosquit
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03:06.50Kattylooks in
03:08.05Kattyrandomly tickles people.
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03:18.15[TK]D-Fendertee-hee!
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03:33.07korcanummm...
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04:01.18Kattyweeeeeeeeeee!!!
04:01.35korcananyone using hylafax and iaxmodem?
04:06.22[TK]D-Fender~whee
04:06.23infobot[~whee] Weeeeeeeee! http://www.albinoblacksheep.com/flash/weeee
04:08.05Kattyi do believe it is naptime.
04:08.09Kattytime for the napping.
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04:26.18wwalkerI have app_meetme loaded and the dahdi and dahdi_dummy modules loaded, but meetme is still puking with :WARNING[4588]: app_meetme.c:800 build_conf: Unable to open pseudo device
04:26.24wwalkerany ideas?
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04:29.16jplohcan anyone help me with asterisk 1.4.26.2 and sipsak? i can't get it to authenticate
04:29.18jplohthese are the parameters sipsak -s sip:test@192.168.100.40:5060 -u testuser -a 123456 -v -U
04:29.25jplohsorry had to send it again
04:29.34jplohit seems that people were asleep earlier
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04:34.34Trepinonhello, I am having some issues with http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax I don't know how to install the addons or tell if they are installed.
04:35.29TrepinonI am planning on using asterisk 1.4.26
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04:37.29jplcwow, it's quiet in here
04:37.37Trepinonyep
04:38.05TrepinonKnow anything about asterisk and faxing to emails ? :) would help me a bit
04:38.58jplccan anyone recommend an asterisk/iax friendly provider to replace/backup vitelity? their service/tech is slipping these days
04:39.18jplcTrepinon: heh, just your luck. i'm in the midst of setting one up
04:39.43Trepinonsweet I could use someone to share my pain with.
04:39.50Trepinonhttp://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax
04:39.55Trepinonis what I am trying to do
04:40.15jplca beer or two while hackin...er...compliing stuff would be better ;)
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04:40.30saxahi
04:40.46saxais there a way to limit the lenght of the voicemail message ?
04:40.56ketemai receive faxes from www.myipcomms.net they give out free dids and support T.38 faxes come into asterisk box with hylafax
04:41.06saxaand also record it in mp3 or ogg ?
04:41.17Trepinonbeer does make life better.
04:41.36saxathats for sure Trepinon
04:41.44TJNIIsaxa: I believe there is a time limit option in voicemail.conf.
04:42.01saxaTJNII: ok, will try to look into it
04:42.36saxamy problem is that when ppl hang up the Zap continues to registering the beep beep beep
04:42.50TJNIIThat sounds like a Zaptel problem
04:42.53Trepinoncan you use hylafax to do fax to email with t.38 and from a sip line ?
04:42.54saxaand sometimes is for 8 or 27 minutes
04:43.12ketemasaxa: i don't believe you can record n mp3, but you can convert to mp3 when emailing the vmails out, thats how i have mine set up
04:43.24ketemaTrepinon: yes
04:43.47saxaketema: can you point me to a how to or a script ?
04:43.58Trepinonthank you that gives me a good start on something
04:44.22ketemasaxa: sure you can use minegive a sec and i'll post it
04:44.28saxawhere can i put this script to convert the waw to mp3 ?
04:44.35saxagreat
04:44.52jplcTrepinon: i looked into that. i'm planning to use the iaxmodem route
04:45.40ketemaTrepinon: thatsthee exact set up i use...IAXModem, and Hylafax it works great
04:46.28ketemasaxa: in voicemail.conf you will set mailcmd=/var/lib/asterisk/agi-bin/savemail.pl the path is arbitrary thats just what i called my script
04:46.47saxaok
04:46.55Trepinonso how does the IAXModem work ?
04:47.15Trepinonis it a hardware modem or a software modem that you can send IAX2 connections to ?
04:47.22jplcsoftware
04:47.31jplcit still has steve's code in it somewhere
04:48.13jplci just chose it because it seems the less painful route on my * box
04:48.26ketemasaxa: the script can be downloaded here...  http://www.ketema.net/savemail.pl
04:48.31jplcapp_nv* nad hylafax are both in the repos
04:48.54ketemasaxa: in order for this to work you have to have lame, sendmail and base64
04:49.11ketemasaxa: all these things are pretty basic on any distro but you might want to be sure
04:49.13saxaketema: got it thanks
04:49.29saxayup thx ketema
04:49.45saxain case i need something i will recontact you ;)
04:49.58ketemasaxa: cool have fun
04:50.07Trepinonthis is the how to I found http://blogtech.oc9.com/index.php?option=com_content&view=article&catid=4:asterisk&id=77:20071121ast&Itemid=6 is this what you are using  jplc ?
04:51.05saxabye
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04:51.53jplcTrepinon: yep
04:52.22jplci'm somewhere in the middle of that howto
04:54.46TrepinonI have an account with www.myipcomms.net do they support answering faxes and sending them as t.38 ?
04:55.18ketemaTrepinon: yes.  my number works fine and all i use it for is faxes
04:55.38Trepinonvery sweet
04:56.09Trepinonso I would just need to setup a hylafax server and I could receive faxes from them in t.38 ?
04:57.43ketemaTrepinon: thats what I do...call come in from ipcomes I pick it up with IAX modem, which faxgetty on hylafax is listening for and it emails me the fax...never need a shred of paper
04:58.38Trepinonawesome thank you for the advise that helps me a lot.
04:59.08jplcnice to know ketema thanks
04:59.18Trepinonjplc and ketema you two rock I am going to go play with hylafax
04:59.21Trepinonttyl
04:59.25jplcgood luck
04:59.29Trepinonyou too
04:59.31ketemaTrepinon: have fun
04:59.39Trepinonalways ;P
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05:25.16alexshellhi all
05:26.28ChannelZah crap I have to reboot my server
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05:26.38alexshelldo you know if there's a trial/demo version of Skype for Asterisk?
05:27.01drmessanonope
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05:27.45ketemaalexshell: you can talk to skype users through asterisk by setting up a gizmo account
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05:28.41drmessanoThats not nearly as good
05:29.16drmessanoSkype For Asterisk is well worth the $66
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05:30.35alexshellI would like to test skype for asterisk before buy it
05:30.52alexshell:/
05:31.12alexshelldrmessano can you give me a feedback about it?
05:31.19drmessanoI already did
05:31.26drmessano[00:30] <drmessano> Skype For Asterisk is well worth the $66
05:32.14ketemadrmessano: what is your definition of "good" ? i have had no issues with voice quality through gizmo
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05:32.23alexshellwhat about voice quality?
05:33.55drmessanoketema: Gizmo is shady at best, requiring you to give them your credentials, when they've proven they like to change the rules.  They also limit what you are able to do with Skype.. This also integrates directly into your dialplan and is much more functional..
05:34.20drmessanoLets also not forget this is a direct connection to Skype, not reliant on a hack or third party
05:35.02ketemadrmessano: good points...i guess for my needs the gizmo hack works good
05:35.17drmessanoI wouldnt trust Gizmo with a 10 foot pole
05:37.09drmessanoHonestly, it's $66 per channel and you get a free G729 license.. I spend 2/3 that on one night eating indian food
05:38.15jplcdrmessano: lol! yum! garlic naan ;)
05:38.18drmessanoIt a native channel driver with a direct Skype connection.. if you're at all _SERIOUS_ about using Skype, this is the way to go.. anything else is as hacky as running some lamp cord and twine to a Windows box with a Skype phone glued to the USB port
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05:39.18dparkerI just installed an asterisk now instance - my first (and first experience with asterisk).  I'm trying to get a sip phone to register with it but no joy.  I've noticed that nothing is listening on the sip udp port - is there some configuration I need to tweak?  (the other channel is silent)
05:39.31jplchey, in the spirit of hackery, i'd run that lamp cord/twine thingy if it didn't run on windows
05:40.09ChannelZdparker: asterisk NOW is supported in #asterisknow
05:40.33dparkerChannelZ yes - I tried that - they're silent - I was hoping someone here would know
05:40.34ketemadrmessano: i just double checked I did not give any credentials to gizmo...but i do agree with your points, I'm just not a SERIOUS skype user.  The few people I do call i have no issues with.  so its really just up to what you want to do
05:40.58drmessanoketema: Then how do they connect to your Skype account?
05:41.11dparkerChannelZ - at the point where we talk about listening ports, it's just asterisk anyway isn't it?
05:42.01drmessanodparker: technically, yes, but according to the GUI Nazis, its somehow different, you suck, and your mom (or something)
05:42.08ketemadrmessano: they have their own gateway called opensky
05:42.11ChannelZyes but I've never even seen *NOW so I can't tell you where to start poking around in the GUI
05:42.20drmessanodparker: So youre right, but somehow horribly wrong
05:42.23ChannelZOn regular * we edit config files manually and dig around in the console
05:43.10jplcdparker: yeah, we're old school and hard core, prefering to wade through all the config files and run a debug trace when something's amiss ;)
05:43.13dparkerChannelZ / drmessano - I just switched to regular asterisk!   How do I configure listening ports / interfaces...?
05:43.18dparker:)
05:44.05jplcdparker: sip should default to 5060, check your sip.conf
05:44.11dparkerk
05:44.15drmessanoExcept you use the CLI and debug traces for any GUI install too
05:44.26ketemadrmessano: and your right...they did change their terms...they do not offer opensky to new accounts apparently anymore
05:44.58drmessanoketema: Yeah, not shady at all.. like the shell game with Google Voice they played
05:46.35drmessanoI guess I am not the type to use $25 shit FXO cards and hacky channel drivers.. I barely use Skype, but $66 to play with it was a minor expense, and saved me from hours of "Does it work, can I see it?  PLEASE?  DEMO PLZ"
05:47.06jplcso i take it you missed the beta?
05:47.31drmessanoBeta is over
05:47.42jplcyep. was just asking
05:48.36drmessanoReally wasnt that interested in it at the time.. Only became interested when someone asked me how it worked.. So, I bought it
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05:50.14jplcah i see.
05:54.38jplohanybody with experience on sipsak? i can't get it to authenticate with asterisk
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06:07.34nyrangerhey all.  I am having a weird issue.  when trying to make a bunch of calls using agi and the Async: true flag.  Basically I can reprodue it trying to make 3 phone calls at a time.  THe error I get is:
06:07.34nyranger[Nov  5 01:00:26] ERROR[2583] pbx.c: Received invalid event that had no device IE
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07:44.37awkgood day, I have run make menuconfig and get XXX (dependancy missing) for misdn, I can't seem to find what I am missing even after reading the readme for misdn. I really need to create my rpm with misdn loaded in?
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07:50.20awkI have misdnuser and misdn
07:53.11ChannelZ<PROTECTED>
07:55.38awkChannelZ: I have no idea what package consists of isdnnet and suppserv for centos
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07:58.44ChannelZis there an misdn-dev package or something?
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08:02.12ChannelZand/or did you re-run ./configure if you put the misdn packages after having done so originally?
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08:07.33awkfound it, missing mISDNuser-dev
08:07.34awkthanks!
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08:13.09dparkerChannelZ I'm trying to get a sip phone to register within my intranet with little joy.  Log messages say Registration from <sip url> failed for <sip phone host ip> - No matching peer found.  I was walking up to believing that localnet sip config entries would solve this - am I on the right path?
08:14.42kaldemardparker: you're on the wrong path. that has nothing to do with the localnet setting.
08:15.24dparkerkaldemar - what are localnet entries for?
08:16.00kaldemarfor defining your LAN ip addresses, as a part of NAT settings.
08:16.37Guggedparker: you dont have a peer named <sip url>
08:18.40kaldemardparker: show your sip.conf and a registration attempt with sip debug (command "sip set debug on") in CLI, and well see what is wrong. use a pastebin:
08:18.43kaldemar~pb
08:18.44infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
08:18.45dparkerI guess I'm just generally in the dark on the sip peer notion.  As, if it is the case that one has to specify an IP address paired with a user name to authenticate, you'd never be able to initiate a session from other hosts (roaming) - and I'm sure that's not impossible.
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08:19.34dparkerwhat I'm asking I guess is - isn't it possible to have a sip user that's not tied to a specific ip address?
08:19.48dparkerand then isn't that a function of saying "I trust this and that network in general"?
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08:20.19ChannelZyes
08:20.24ChannelZbut thats not your problem at all
08:20.46ChannelZThink of your SIP phone as a user.. it has a username and a password
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08:21.18dparkerso "peer not found" is actually just a failure to authenticate?
08:21.22mchouasterisk IP acls dont work worth a damn
08:21.23kaldemardparker: no, you don't have to specify an ip address. actually, it's better to specify host=dynamic and let the client be where ever it is.
08:21.36ChannelZdparker: yes and no
08:21.51ChannelZdparker: what does your sip.conf look like
08:22.01mchoudparker: start from the beginning
08:22.26mchoudparker: where is this asterisk box in relation to the phones on the network
08:22.35dparkeryou'll have to allow that the asterisk instance I'm running came from an asterisknow iso :)
08:22.36arekmhi, when someone dials number then he have silence in earphone for some time and then dialing signal appears. Is it possible to put somthing other than silence in the beginning? (people think that nothing is happening while it takes some time)
08:22.48mchoudparker: are they all on the same LAN?
08:22.56ChannelZdparker: the error you're getting is because your phone is trying to register using the peer name "whatever" but you have no corresponding entry in sip.conf
08:23.01dparkermchou - no - it's a pretty complex network actually
08:23.08mchoudparker: lol
08:23.20mchoudparker: that's why we need this info
08:23.32mchoudparker: it could be a firewall issue
08:23.39dparkerno - traffic is fine
08:23.47dparkerthe networking is functioning properly
08:24.07dparkerjust think of an intranet with 3  /26 networks
08:24.12mchoudparker: how did you figure that out?
08:24.24mchouI'm talking sip nat traversal
08:24.25dparkerthe asterisk server is on  one of them, the sip phone is on another
08:24.38dparkerI sniffed packets at the asterisk host
08:24.43dparkerlooks fine
08:24.54dparkertraffic is bi-directional between asterisk and the soft phone
08:25.09mchoudude
08:25.12ChannelZthis is a config problem
08:25.22mchouwe know it's bidirectional
08:25.24kaldemarmchou: so far nothing indicates that there is a network problem. let's concentrate on the peer matching issue first.
08:26.12dparkerconstructs a pastebin...
08:26.20mchouI'm fine with that
08:26.37mchouno pb no help
08:28.18dparkerhttp://svdasein.pastebin.com/m73f3b720
08:28.40arekmany ideas?
08:29.06mchoudparker: wtf is that?
08:29.08kaldemardparker: you need to define a client for the phone
08:29.33ChannelZyou have no peers defined
08:29.33dparkerkaldemar - bear in mind this is a freepbx installation..
08:29.40mchoudparker: ~book
08:29.42ChannelZ[MyPhone]
08:29.45kaldemardparker: off to #freepbx then :)
08:29.47ChannelZsecret=barf
08:29.48ChannelZand such
08:29.50mchou~book
08:29.51infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
08:30.11dparkerkaldemar - yes - I know - I'm trying to ignore the ui at the moment :)
08:30.43mchouhears the sound of flushing approaching
08:30.48jplcdparker: i recommend scrapping that sip.conf and building your own from scratch ;)
08:30.59jplcor flushing...works just as well
08:31.08mchoujplc: you think? :)
08:31.12kaldemardparker: you can't ignore it if it generates your configs
08:31.19ChannelZand/or scrapping freepbx because the second you fire up the GUI again it will probably trash everything
08:31.26dparkerfunny thing is that there was no sip.conf or anything
08:31.29jplcChannelZ: lolz
08:31.31dparkerwhich seems wrong
08:32.00dparkerso what I pasted there is just some stuff I culled together with some googling
08:32.06ChannelZit is, but probably because you never set anything up first.
08:32.07jplcthat's the gui's fault
08:32.08mchouit's beyond wrong
08:32.59mchoudparker: you're lucky we dont give you the silent treat ment
08:33.22mchouor refer you to esr's "Samrt Questions"
08:33.26dparker:) ok - I appreciate your patience.  I'll start over :)
08:33.32mchouSmart*
08:34.12jplcwe may be hard-core build-all-conf-files-from-scratch * guys but we're patient alright
08:35.21mchounobody wants to wade thru a 50 page dial plan just to solve your "problem"
08:35.35mchouthat's really the bottom line
08:36.13mchouand the problem with all guis
08:36.41JAMMAN2110Well, one of the problems with guis anyway
08:36.46jplcthey break up the conf files to "streamline" configuration...
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08:37.10JAMMAN2110My main problem with FreePBX is I cant decide if I should put it on the work PBX when I build it....
08:37.35JAMMAN2110Its a debate between, what the GUI does to configs / etc and how much faster things will be to use the config...
08:37.50jplcit's a toss up
08:38.04JAMMAN2110Well, the main thing would be adding extensions and changing voicemail passwords
08:38.13jplcif you can trace what the gui is doing, you're probably better off without it
08:38.27jplcah yes. that too
08:38.38JAMMAN2110Im sure it would be easy enough to build a form, just a form, that appends to extensions.conf
08:39.14JAMMAN2110But Id rather avoid having Apache on the box
08:39.28JAMMAN2110Perhaps a basic C++ app, over SSH
08:39.37mchouwhat??
08:39.44jplcor add extensions??.conf to the bottom?
08:39.50mchouall this stuff should be DB driven
08:39.57jplcyou'll end up with another gui-like problem hehe
08:40.04mchouno need for local apache
08:40.40mchoujplc: I totally disagree
08:40.50mchouDB is structured
08:41.00jplcno no, i meant JAMMAN2110's c++ app
08:41.02dparkerwell I'll tell you (for what it's worth) - 'til about 4 hours ago I'd never so much as thought of installing asterisk before, so I figured I'll get something close to an appliance to get some kind of feel for how the thing is configured.  The idea being that it'd be a working instance and all...
08:41.04mchouconfig files arent
08:41.26dparkerobviously my plan had a flaw
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08:41.44mchoudparker: go use windoze
08:42.00mchoudparker: try 3cx for GUI
08:42.05ChannelZit's not a horrible plan if you're just farting around but you didn't even configure it all the way
08:42.32dparkerChannelZ - I just did what the quick start guide says - no more no less.
08:42.36jplci seem to remember a howto somewhere...boot up one of the * on cd distros and configure a softphone...
08:42.38mchouChannelZ: all the way??  he didnt even _start_
08:43.57mchouChannelZ: a two year old can copy and past from google more competently
08:44.04mchoupaste*
08:44.15jplchehe - paste ...
08:44.40JAMMAN2110Hmm
08:44.54JAMMAN2110It would be nice if the entire asterisk config file collection was in a mysql database
08:44.56mchouof course with the DB approach there will be other issues
08:45.08JAMMAN2110Yes their would be issues
08:45.14JAMMAN2110Wrong there / their ?
08:45.30mchouevery time you revise the schema goodbye to compatibility with older versions
08:45.36JAMMAN2110But it would make scripting things like new extensions etc much easier
08:45.42JAMMAN2110mchou - that would be the problem
08:45.58JAMMAN2110But then, there are plenty of apps out there that update databases in mysql when they get upgraded
08:46.17mchounono, upgrading is not the issue
08:46.24mchoudowngrading is
08:46.42JAMMAN2110Production PBX boxes shouldnt need to be downgraded
08:46.47mchouyou dont want to upgrade and be holding bag if shit dont work
08:47.00JAMMAN2110And hey, its easy enough to dump a database and reimport it later
08:47.28mchouJAMMAN2110: I think we're in violent agreement
08:47.41ChannelZEVERYTHING XML!
08:47.51mchoubut I was suprise I couldnt downgrade quassel
08:47.56JAMMAN2110XML would cause chaos
08:48.07JAMMAN2110Youd tripple the size of the configs
08:48.10mchousurprised*
08:48.10ChannelZheh isn't that what freeswitch uses?
08:48.42JAMMAN2110Not sure
08:49.22JAMMAN2110The kind of things I do with Asterisk would be a lot easier with a MySQL backend though
08:49.29JAMMAN2110Is there a way to convert the current configs to MySQL?
08:49.43mchouhaha
08:49.49JAMMAN2110I know you can call on say a PHP script and it can do MySQL functions in dialplans etc
08:50.01mchouif there were there wont be so many sill gui projects
08:50.10mchousilly*
08:50.59JAMMAN2110My first and very messy asterisk project: http://blog.jamesforman.co.nz/2009/01/controlling-applications-with-asterisk/
08:51.07JAMMAN2110Ive been thinking about revising that
08:51.23JAMMAN2110But then, I havent posted any of my more recent ones
08:51.37JAMMAN2110Currently working on a web app for Contact ID Alarm Codes
08:52.01JAMMAN2110And an internal extension to send SMS messages through my gsm gateway
08:52.18JAMMAN2110Considering TTSing incoming ones as well :)
08:52.36jplcJAMMAN2110: wow. talk about a rube goldberg solution ;)
08:52.50JAMMAN2110Can just imagine it though "U hav 4 nw bstmts"
08:53.02JAMMAN2110Thats the kind of SMSs you get from Mobile providers here
08:53.07JAMMAN2110Only they use the full 160 char
08:53.29jplcyou're gonna make the poor TTS engine die from tongue twisters
08:53.44JAMMAN2110jplc: yeah, it was definately rough around the edges, but it was quick easy and gave me some better insight into some * stuff
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08:53.56JAMMAN2110Thats the problem, I wouldnt want to call and listen to them
08:54.02jplcme neither
08:54.21JAMMAN2110The TTS and SMS Extension would be an addon to the webapp Im doing for SMSing through the gateway (Project #4)
08:54.30JAMMAN2110But it just.. would serve no purpose
08:54.55jplcjust keep em txt. less data overall and less brainpower needed to decipher the message
08:55.14JAMMAN2110Though, being able to hit Speaker phone, dial 767, <phonenumber>, <Message> would be nice
08:55.56JAMMAN2110I have VoIP phones everywhere
08:56.19jplcbut it'll be easier to parse a txt msg than a STT dump
08:56.43JAMMAN2110The main issue Im having is getting the web app to actually submit the form on the gateway
08:56.56JAMMAN2110The sending is handled by a CGI script, but it only listens to localhost
08:57.25JAMMAN2110Telnet with AT commands, just isnt fun nor something I want to continue exploring
08:57.25Nuggettelnet is eeeeeeevil!
08:57.31jplclol
08:58.03jplchave you tried the * manager interface?
08:58.13mchouJAMMAN2110: you have any experience with LG_Nortel 6812?
08:58.16JAMMAN2110Actually, if any of you have a PHP script lying around that submits to a remote HTML form, that would be insanely helpful
08:58.25JAMMAN2110mchou - Unfortunately no
08:58.58JAMMAN2110I did however sucessfully convert a Grandstream Budgetone 100 into a FM radio killer
08:58.59mchouJAMMAN2110: what voip phone you use?
08:59.09mchouphones*
08:59.49JAMMAN2110I have 2 ATAs, 2 GXP 2000s (Grandstream), 4 Soft phones, 8 GStrm BT100s and 101s <- inherited all those phones :)
09:00.01mchoubah
09:00.03JAMMAN2110I plan to get some nice Aastra's
09:00.11JAMMAN2110But Id rather get the system working perfect first
09:00.21JAMMAN2110And right now, the GXP2000s work fine for me
09:00.25mchouGrandsuck
09:00.31JAMMAN2110Yes, I agree
09:00.45JAMMAN2110But you cant argue with FREE
09:00.47JAMMAN2110;)
09:01.05mchouyes you can
09:01.22JAMMAN2110Not really
09:01.22mchouif you spend time fixing their bugs
09:01.46JAMMAN2110I havent found any bugs with the phones, honestly, all I use them for is inbound and outbound calling
09:01.52JAMMAN2110I dont use any other feature of them
09:02.05JAMMAN2110Not even the speeddial
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09:02.48mchousomeone is selling 6812s for $30 here
09:03.06mchouI have no idea if they are good or not
09:03.15JAMMAN2110Im going to have an insane budget for works VoIP rollout :)
09:04.38JAMMAN2110mchou / jplc - Do either of you have a GSM Gateway?
09:04.46JAMMAN2110Or similar
09:04.47mchounot me
09:04.50jplcnope
09:05.05JAMMAN2110Bugger
09:05.17JAMMAN2110Retailed for 950$, got it for 600$ less than that :)
09:05.33mchounice
09:05.35tuxx-eugh, gsm gateways
09:05.36tuxx-:D
09:05.45JAMMAN2110The joy of being friends with the owner of one of the only VoIP companies in the country
09:05.47tuxx-luckily my coworker handles those
09:06.05JAMMAN2110tuxx- Cant help with my SMS gateway issue can you? :)
09:06.18JAMMAN2110Ive had 3 programmers look at it, and spent 3 months trying
09:06.21tuxx-not really familiar with them
09:06.33mchoulol
09:06.36*** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:7dc7:ee9e:3e12:233a)
09:06.47mchoumaybe that's why it was on sale :)
09:06.51JAMMAN2110Around work and being on morphine (tissue graft)
09:06.55JAMMAN2110Nah, the gateway is fine
09:07.03tuxx-what seems to be the problem?
09:07.06JAMMAN2110Im just not exactely using it for its description
09:07.14tuxx-its not connecting to your sip proxy or something?
09:07.28JAMMAN2110Oh no, calling works fine, it works fine with its own web app
09:07.37tuxx-mkay
09:07.37JAMMAN2110Im trying to integrate it into my home automation app though
09:08.00JAMMAN2110So I can send SMS and receive SMS through the HAA (home automation app - I know, lame acronym)
09:08.09tuxx-hehe
09:08.09tuxx-;D
09:08.25JAMMAN2110Gateway has a basic HTML form, that submits to the Gateway's smsSendNow.cgi script
09:08.36JAMMAN2110But unfortunately its only accepting connections from localhost
09:08.43tuxx-heh
09:08.53JAMMAN2110So I have to submit through the actual HTML form, and not the cgi script
09:09.13JAMMAN2110smsSend.htm -> smsSendNow.cgi
09:09.14dparkerso on the off chance that this girly-man thing I'm re-installing doesn't work, what approach would you guys suggest for getting a basic instance running?  Is the default configuration in fedora/ubuntu/whatever reasonable?
09:09.17tuxx-well, that should be possible right? :P
09:09.49JAMMAN2110dparker - If this is your first go, download PBX in a flash, Elastix, trixbox... etc, try as many as you can
09:10.00JAMMAN2110tuxx-, you'd think so, but its suprisingly difficult
09:10.14dparkerJAMMAN2110 - roger, thanks
09:10.23JAMMAN2110I figured it would be easy as eating cake, but 3 months later, and 3 programmer friends later...
09:10.32jplcJAMMAN2110: if you knew perl, i'd point you to a module that does what you want...crawl a form
09:10.49JAMMAN2110I can pull the incoming SMSs off the gateway
09:10.59JAMMAN2110Just not push outgoing ones
09:11.23jplcyou just can't remotely access the cgi script...
09:11.42JAMMAN2110If I could remotely access the cgi script, it would be fine
09:11.58JAMMAN2110Unfortunately I cant modify the source on the device either
09:12.35dparkerJAMMAN2110 what are you trying to do?  Post form data from flatfiles or something?
09:13.07JAMMAN2110Well, its dual SIM, so the plan is to have a web app (half complete) which you can send and recieve SMS with
09:13.29JAMMAN2110And then Im going to build onto that, and have incoming commands via SMS and outgoing alerts such as alarm activations etc
09:13.47*** join/#asterisk viq (n=viq@unaffiliated/viq)
09:13.59JAMMAN2110So really what I need to do is have myAppsSendingScript.php -> smsSend.htm -> smsSendNow.cgi
09:14.16dparkeroh ok
09:14.45JAMMAN2110It doesnt have to be PHP, it just has to be easy to implement and maintain
09:14.56jplcif only php had something similar to perl's WWW::Mechanize
09:14.57dparkerwhere smsSendNow.cgi sends form data to some other page/site/whatever?
09:15.19JAMMAN2110smsSend.htm and smsSendNow.cgi are on the GSM Gateway itself
09:15.39dparkerso you can actually use command line stuff then?
09:15.50dparkeror pipes or whatever?
09:16.06JAMMAN2110jplc - I could call a Perl script from the PHP script couldnt I?
09:16.13jplcoverkill ;)
09:16.15JAMMAN2110dparker, what do you mean by command line?
09:16.22jplcah, how about http://www.lastcraft.com/browser_documentation.php?
09:16.24JAMMAN2110jplc - Yes, but well, Im rather desperate
09:16.45jplcit's about as close to WWW::Mechanize as you can get on php
09:17.26jplcor heck, install activestate perl and copy the test perl scripts
09:17.34dparkerJAMMAN2110 - does the GSM gateway have command line utilities that do smsSendNow-like stuff?
09:17.45jplcthey're less than a page long. not complex
09:17.47JAMMAN2110Eekads jplc - that could be just what Im looking for!
09:18.10jplcgood ;)
09:18.21JAMMAN2110dparker, It has Telnet, but Im not the biggest fan of AT commands, and have tried sending SMS through AT commands multiple times with no sucess
09:18.38jplcif that box was with me, and three programmers failed...i'd load backtrack and crash the http server on it hehe
09:19.06JAMMAN2110I considered doing similar, with a hammer
09:19.12JAMMAN2110But then, 350$ box
09:19.13dparkerso what's the host you're telnetting to?
09:19.18jplcnot very constructive >:)
09:19.34JAMMAN2110192.168.1.68 / gsm.lan
09:20.26dparkernot telnet localhost... then
09:20.51JAMMAN2110Yeah, I did wonder if the cgi script telnets to localhost
09:20.58dparkeroh - it is?
09:21.00dparkerit's doing that?
09:21.03JAMMAN2110I have no idea
09:21.06dparkerok - well
09:21.14JAMMAN2110The telnet interface is a tad, crazy
09:21.33dparkermy thinking is that if it *is*, then whatever it's connecting to...  there's probably some command-line ish way to do whatever you're doing through the telnet interface
09:21.38JAMMAN2110Its only got 2 SIM cards, but the telnet interface gives you the choice between "module0" "module1" and "module2"
09:21.38jplcthey all are...those embedded mini oses
09:21.50dparkerit might even be a stdin like service just being served out of inet.d
09:21.58jplcdparker: you have a point
09:22.21JAMMAN2110I cant confirm the cgi script is connecting to the telnet server, theres no way for me to intercept it
09:22.36jplcJAMMAN2110: have you tried to port scan it?
09:22.54JAMMAN2110Yes
09:22.57JAMMAN2110HTTP and Telnet
09:23.01jplcheh
09:23.04JAMMAN2110Full port scan
09:23.05jplcbloody telnet
09:23.07dparkerdid you scan the loopback interface?
09:23.20JAMMAN2110No
09:23.33dparkertcpdump -lvvi lo
09:23.38JAMMAN2110Im going to take that as a trick question ;)
09:23.42jplclol
09:23.51JAMMAN2110I cant issue such commands through its telnet interface
09:24.07jplcdparker: i doubt tcpdump is installed on that thing
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09:24.19JAMMAN2110Thus I cant check the lo interface
09:24.27dparkerah
09:24.32dparkernetstat -an | grep LISTEN
09:25.09JAMMAN2110"bad command!!"
09:25.14dparkerhrm
09:25.15JAMMAN2110Nice default response
09:25.25jplci have a gsm wireless device i ordered from china a long while ago. same problem.
09:25.43jplcno ports open but telnet and http
09:25.54jplcand the documentation was crap
09:25.55JAMMAN2110I considered some GSM USB modems instead
09:26.09JAMMAN2110But this seemed like a much more... acceptable approach
09:26.14JAMMAN2110What with proper SIP connections and all
09:26.26*** join/#asterisk ruyo (n=sayo@195-23-253-223.net.novis.pt)
09:26.58JAMMAN2110Just going to try that stuff jplc linked me too
09:27.29jplcgood luck with it, if it doesn't meet your needs, i'd recommend trying the perl route
09:28.05dparkerif it's just a simple form, it's just a libwwwperl thing - very straightforward
09:28.29jplcdparker: i recommended him using WWW:Mechanize
09:28.44jplcit'll make filling out the form easier
09:28.59dparkerroger
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09:29.57JAMMAN2110Id rather keep it in the same language, so the PHP option gets tested first ;)
09:29.57*** join/#asterisk sgimeno (n=sgimeno@226.Red-80-33-64.staticIP.rima-tde.net)
09:30.03nicola_pavAmorsen: there?
09:30.53AmorsenYes
09:31.24jplcJAMMAN2110: agreed. ScriptableBrowser can fill out forms too so it should work
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09:32.17nicola_pavremember me?
09:32.20AmorsenYes
09:32.23nicola_pavgood
09:32.40JAMMAN2110jplc - Obviously the 3 web dev guys that tried it out, dont test their websites through automated means ;)
09:32.48nicola_pavyestrday i was using xlite to test
09:32.51JAMMAN2110Sockets != fun
09:32.57nicola_pavnow i am trying landline and mobile
09:33.02nicola_pavmobile works fine
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09:33.09jplcJAMMAN2110: egad. sockets.
09:33.11nicola_pavbut with landline when i call
09:33.15nicola_pavno voice at all
09:33.35JAMMAN2110I dislike sockets
09:33.40JAMMAN2110They arent easy enough
09:33.40nicola_pavi have to put answer() in the script so i can hear messages
09:33.50nicola_pavany hint?
09:34.05nicola_pavand if i put answer() i will be at the same point as yesterday
09:34.16nicola_pavit will deduct from pin in a worng way
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09:34.55nicola_pavsince mobile and xlite works fine, it has to be pstn issue
09:35.05nicola_pavright?
09:35.40JAMMAN2110Partner just came up to me, just got back, said hi, my response "busy finishing 3 month project"
09:35.45JAMMAN2110<3 is fun
09:35.46mchounicola_pav: how are you hooked up to PSTN?
09:36.03nicola_pavthere is a toll free number 1800
09:36.10mchounonp
09:36.15nicola_pavconnected to pstn and from pstn to cisco
09:36.24nicola_pavcisco gateway as5400
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09:37.11Amorsennicola_pav: Try adding a Progress() like I said yesterday
09:37.29nicola_pavinstead of answer Progress()?
09:37.35AmorsenIf it doesn't help, talking to your provider. If that doesn't help, you'll have to go the ResetCDR route
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09:37.48AmorsenYes, Progress() where you used to Answer()
09:37.59nicola_pavi will try that, ok
09:38.58AmorsenI suppose you could invent an Answer() command which didn't actually tell the CDR subsystem that the call is answered, but it'd be a bit of a hack
09:39.39nicola_pavAmorsen: i am thinking that it could be cisco configuration
09:39.50nicola_pavconfiguring correctly the voice-port
09:39.52nicola_pav?
09:39.56nicola_pavit could be right?
09:40.57AmorsenYou could, but early media isn't universally supported on the PSTN
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09:41.28AmorsenI can't help you with the Ciscos, I've never touched their voice stuff
09:42.17TSManyone had issues with indications not changing from the US default?
09:43.18AmorsenTSM: I did, but it turned out to be user (me) error
09:45.04Amorsencountry=dk but [da] in the list...
09:45.08TSMi having issues, ive set the country=uk in the indications.conf, but any incomming call or internal call all comes up with us rings
09:45.59AmorsenTSM: Internal call? from a SIP phone?
09:46.30AmorsenThe phone might be playing their own tones
09:48.04JAMMAN2110Hmm... Fatal error: Using $this when not in object context
09:48.34TSMAmorsen: naa its not,  im able to play MOH to the user if i wanted to
09:49.01JAMMAN2110At least these are reasonably easy errors :) Fatal error: Class 'WebTestCase' not found
09:50.02JAMMAN2110Oh, Im an idiot, Im pointing it at the cgi script and not the actual form *headdesk*
09:50.14AmorsenTSM: I'd probably stick in an Answer() and then add ,r to the Dial command, just to see if that fixed the problem.
09:50.38AmorsenIf it does fix the problem, at least you have excluded indications.conf from the problem area
09:53.29TSMit fixed the problem, but this is deffo a bug, seems the indications do not work with on hook inband audio, this does exist as im able to do onhook inband MOH
09:53.51JAMMAN2110And.. Im stuck :|
09:54.27AmorsenTSM: You have just proved that the audio is generated from somewhere else than indications.conf
09:55.18AmorsenIf you want further proof, wireshark is great
09:55.25TSMAmorsen: no i did not, I can play MOH to the caller without answering the call, i just cahnge the dial command so instead of 'r' to 'm(default)'
09:55.43jplcJAMMAN2110: errors?
09:55.59JAMMAN2110Fatal error: Class 'WebTestCase' not found in C:\wamp\www\testSMS.php on line 8
09:56.20JAMMAN2110I copy pasted from an example or two, could be the problem
09:56.21JAMMAN2110Ill pastebin
09:57.02JAMMAN2110http://pastebin.com/d5e6f60c
09:57.08AmorsenTSM: But you just said that adding ,r + Answer() fixed the problem.
09:57.27AmorsenThat forces the phone to use whatever audio Asterisk sends it
09:57.53TSMAmorsen: read what i wrote, i can play MOC (Music on hold) to the user without using the Answer() command, and the time counter does not start on my phone
09:58.28TSMso somthing is inconsistant
09:58.59JAMMAN2110jplc - Did I mention Im doing most of this to learn more PHP? :)
09:59.13JAMMAN2110Im a network tech, not a programmer
10:00.30jplci picked up on that ;)
10:00.37JAMMAN2110ROFL!
10:00.46JAMMAN2110My cats magnetic collar just got stuck to my server cabinet :D
10:00.52jplclol
10:00.55AmorsenTSM: Asterisk doesn't generate audio for ringing if it isn't forced to do so, AFAIK
10:01.50AmorsenIt just passes whatever the other end sends, and if the other end doesn't send anything, hopefully the calling phone will generate its own audio
10:01.56TSMwell, our old phone system again without answering the call would ring a UK ring style, so clearly its a * problem
10:02.34JAMMAN2110Or it could be your phones TSM (I apologise if Ive missed a point and come into a conversation midway)
10:03.08AmorsenTSM: If you want Asterisk to generate sound, use ,r. But it would be better to fix the phones.
10:03.31TSMthis is happening to external clients calling our office
10:03.39TSMnot only our phones in the office
10:03.44TSMthats why i say its an * problem
10:04.00TSMall our clients on tuesday started to think we had moved abroad
10:04.01JAMMAN2110TSM - You dont live in the UK I take it?
10:04.05TSMi do
10:04.56kaldemarTSM: how do the clients dial in?
10:05.00jplcJAMMAN2110: not so obvious question but did you download and install simpletest first? imho it doesn't come bundled in WAMP
10:05.01JAMMAN2110Im confused, Ill assume Ive missed something and stop commenting ;)
10:05.07TSMkaldemar: they dial our main number
10:05.13JAMMAN2110jplc - downloaded and extracted etc
10:05.14AmorsenHeh.
10:05.21kaldemarfacepalms
10:05.28kaldemarTSM: what technology?
10:05.36jplcJAMMAN2110: ok, so we're missing a step.
10:05.38JAMMAN2110Used requires rather than includes ;)
10:05.45TSMkaldemar: PRI lines
10:05.53JAMMAN2110Indeed
10:06.53jplcJAMMAN2110: did you try the .. um .. simple test cases?
10:06.58jplcsimpletest.org
10:07.20kaldemarTSM: and how did you configure tone zone?
10:07.37TSMkaldemar: country=uk in the indications.conf
10:08.13TSMive checked the full log and it seems the system is setting the default to UK as well
10:08.22kaldemarTSM: what about system.conf for DAHDi?
10:08.38JAMMAN2110Hmm, looks like I included the wrong file, but it still doesnt work
10:08.42JAMMAN2110Ill have a play jplc ;)
10:09.10*** join/#asterisk TJNII (n=TJNII@207.189.199.62)
10:09.32*** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-54660266eb349165)
10:09.37TSMkaldemar: ok this is what im talking about, i did not notice that the sangoma software had put us in the system.conf file
10:10.34TSMjust need to wait till no one is making a call so i can reload dahdi
10:10.34JAMMAN2110jplc -> Fail: -> Bad TestSuite [testSMS.php] with error [No runnable test cases in [testSMS.php]]
10:14.06gr0mitJAMMAN2110, what is the prb again with tones?
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10:14.30JAMMAN2110gr0mit, Im not having an issue with tones
10:15.10gr0mitah ok
10:15.21gr0mitoh, it is TSM
10:15.32JAMMAN2110:)
10:16.20gr0mitTSM, is this a new install?
10:17.34TSMyey finaly, thanks kaldemar, my incomming PSTN calls are now ringing correctly :)
10:18.52TSMis there any other file i have to put loadzone defaultzone into?
10:20.06JAMMAN2110jplc - what time is it where you are? I might head off and give this some thought tomorrow (in about 40 minutes)
10:20.11kaldemarTSM: that's it, iirc
10:20.35jplcPST ;)
10:21.14jplcand that would be 2 am
10:21.29JAMMAN2110Nice, your sleeping patterns might just be as bad as mine
10:21.32TSMkaldemar: good, now im off to track down the internal ring stuff, it may be phone related as there is a load of stuff in the XML but no documentation from poly regarding it
10:21.38jplclol
10:22.14jplcfunny thing is i onced worked in hongkong and the timezone didn't change a thing
10:22.28JAMMAN2110Interesting
10:22.51JAMMAN2110Ive taken so many drugs the last few weeks, had surgery, really screwed my sleeping pattern
10:23.19jplc<-- mt dew
10:23.31kaldemarTSM: appendix A in the admin guide
10:24.24JAMMAN2110Oh well, I need to shower and such, take a break for the first time in 7 hours
10:24.29TSMkaldemar: you mean the chord bit?
10:24.30JAMMAN2110Back later ;)
10:25.02gr0mitTSM, what issues are u having?
10:25.27TSMnothin major now, just getting the polys to have UK progression tones
10:25.49gr0mitfor o/g calls?
10:25.51TSMive found a Australian set on VoipInfo site, so i need to change that for uk stuff
10:28.23*** part/#asterisk machoman48 (n=machoman@89.203.164.69)
10:28.29dparkerscore!
10:29.45kaldemarTSM: yes
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10:30.57TSMkaldemar: yup im working though tthat at the moment, its a bit thin on info, hope it all works
10:33.05gr0mitTSM, do u want my UK tones info for polycom?
10:33.20TSMgr0mit: that would be brill :)
10:33.31gr0mit1 sec let me dig it out
10:35.36gr0mitpm me ur email
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10:40.11gr0mitokay, sent
10:41.47ruyoHow can I know what options does ./configure provide? Options like --without-thingies
10:42.22jkroonusually ./configure --help
10:42.49ruyoOk, thanks
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10:52.29TJNIIMan, this grandstream phone sure ARPs a lot.  It is like a confused old man on my network.  "Who am I? Who are you?  Who are all these people?"  Every 15 minutes.
10:52.53Chainsaw:D
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11:05.23JAMMAN2110TJNII - Be happy yours only do it every 15 minutes
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11:20.29JAMMAN2110Well that was painful, silly cables
11:21.22AmorsenDon't worry, wireless is even less reliable
11:24.32JAMMAN2110Oh, I know all about wireless, its half my job :) WIPS FTW
11:24.57JAMMAN2110Whats more fun is running out of stranded cable
11:25.02JAMMAN2110So having solid cable run to your bed
11:25.11JAMMAN2110Deosnt end up so well
11:26.55JAMMAN2110Doesnt*
11:28.21AmorsenHeh
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11:30.27mchouTJNII: It has to arp cause it's ready to fail over :)
11:31.56AmorsenGrandstreams are good for job security
11:31.58JAMMAN2110I doubt the people at Grandstream have ever heard the term "Fail Over"
11:32.07JAMMAN2110I have to agree with Amorsen
11:32.20AmorsenOnce you know all their little cranks you're indispensable, it would take far too long to educate someone else
11:33.15AmorsenLike which firmwares work best with a particular hardware revision
11:33.25mchouJAMMAN2110: I think you're mistaken.  Grandstream is intimately familiar with that
11:33.49JAMMAN2110PoE + Script to reset the ports at midnight and 15 past noon every day ;)
11:34.05JAMMAN2110Solves almost EVERYTHING
11:35.11AmorsenIf you can afford managed PoE switches, you can afford decent phones
11:35.45JAMMAN2110Fair pint
11:35.48JAMMAN2110point*
11:35.58JAMMAN2110My O key has some cat5e under it, please frgive me
11:36.37JAMMAN2110My father is giving me a talk, about how not everyone has the internet
11:36.41JAMMAN2110.......Over facebook
11:37.57JAMMAN2110I dont know whats more concerning, that hes up this late and on the internet, or that hes up this late on the internet talking to me
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11:40.11xrmx__does anyobdy know if didww supports t38?
11:40.48*** join/#asterisk baijum (n=baiju@122.166.46.113)
11:42.58JAMMAN2110xrmx__, http://www.google.com/search?ie=UTF-8&oe=UTF-8&sourceid=navclient&gfns=1&q=didww+t38
11:43.23JAMMAN2110Funny! Your answer is in the FIRST result! You dont even have to leave the google search results to get the answer!
11:43.46xrmx__JAMMAN2110, i cannot find an authoritative answer, the first one is one guy that says try didww, but didww site does not mention t38
11:44.13infernixstruggling with chan_mobile doesn't get me far. is anyone aware of a hardware FXO solution that pairs with cellphones over bluetooth?
11:45.50JAMMAN2110infernix, I was looking at doing that, after a while I gave up and got a GSM Gateway
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11:47.53mchouwtf
11:47.59mchouquassel just crashed
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11:51.14infernixJAMMAN2110: but that makes it impossible to receive calls on the same number in asterisk and on your cellphone. right?
11:51.40infernixeg you have a separate number+subscription in the gateway?
11:52.19JAMMAN2110Erm well
11:52.55JAMMAN2110NZ has 3 Mobile providers: Vodafone (GSM), Telecom (CDMA / a hundred other things except GSM) and 2Degrees (GSM)
11:53.05JAMMAN2110So I have Voda and 2degs in my Gateway
11:53.12JAMMAN2110And have the calls forward to my non GSM cell
11:53.14JAMMAN2110:)
11:53.18JAMMAN2110If Im not home
11:53.41JAMMAN2110Im not sure how well chan_mobile will work if the phone is taken away / brought back
11:53.46JAMMAN2110And you'll be quite limited on hardware
11:54.30infernixwell it works somewhat, but audio has jitter. very constant though so doesn't seem like its lagging
11:54.35infernixmaybe i'll try and file a bug
11:56.07JAMMAN2110I dare say you'd be waiting a while
11:56.12JAMMAN2110Good that you have it working though
11:56.14JAMMAN2110What phone are you using?
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12:11.07infernixAndroid, original G1
12:11.24infernixit worked back when it was running android 1.1 or 1.3
12:11.32infernixbut broke afterwards. now it semi-works on 1.6
12:11.54infernixhm, maybe i'll briefly try it in my Kaiser, should work with winmo
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12:22.21ruyoWhat file holds the configuration I save when I use "make menuselect"?
12:22.36leifmadsenmenuselect.makeopts I think
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12:25.04ruyoThanks, looks like it.
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12:27.11ruyoFunny, looks lie it's inversed.
12:28.46senate1610Hi, I have been battling with this for some time now. Please can someone assist me in advising how I can disable (or change at least to none) the 'default' MOH on an outbound call? When I phone someone, I want to be able to hear their moh (or whatever they have), and not my own.  having problems when call is retrieved from far side, Asterisk does not unhold the call, and I hear hold music indefinatley. Thanks so much !
12:29.41senate1610When the called person puts me on hold that is, I should hear their hold music. Tx
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12:31.42leifmadsenthat's how it should work normally...
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12:31.58leifmadsenis the person you're calling on the same system as you?
12:32.12leifmadsenbecause typically everyone has the same hold music
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12:45.22tzafrircoppice, mISDN as opposed to DAHDI on the same device? (regarding your message to asterisk-users)
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12:47.28coppicedunno how well DADHI works with BRI cards. DADHI isn't perfect with any hardware, so it I guess it won't be perfect with BRI cards. However, mISDN makes it look FANTASTIC :-)
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12:53.38dcmwaihere is more noiser then trixbox
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12:53.46dcmwaitrixbos is totally quite...
12:53.51dcmwaiHello all :)
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12:57.26Omorikaaudiohook.c: Read factory 0xa0c0fa0 was pretty quick last time, waiting for them.
12:57.34Omorika??
12:58.31TSMis their a dialplan command to pause, will the TIMEOUT do this for me?
12:58.47TSMthere woops
12:58.48Omorikadoes anyone have more info about this debug message?
13:06.31TheDavidFactorTSM, you may want to look at the wait command
13:07.33TSMok also is there any way to send sip messages within the dialplan or do i have to execute seperate shell command, i seen references to SIPnotify but it does not work
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13:08.28senate1610I installed Trixbox 1.6 and tested last night, not good at all, had issues with queues. only worked with ringall
13:09.45senate1610the people I am calling are not on the same system as me, but when they put me on hold, yes Asterisk will my my MOH, i want this disabled, because the call never comes back to me, i test on my cell as well, and i retrieve the call, yet I still hear moh on asterisk (SIP phone).  using Dahdi, but same on ZAP
13:10.15senate1610is there a way to disable this 'default' moh? in the kernal or somewhere in extensions.conf?
13:10.46odenkossenate1610: musiconhold.conf ?
13:12.22senate1610unfortunatley not, there i can create another class (nomusic) and point to /dev/null, but cant seem to set my outbound trunk to use this. if I change the [default] to /dev/null then works fine, but all other dont hear moh (obviously, lol))
13:14.30senate1610so basically just need to change /disable this on outbound channels / trunks so that I hear the dialed numbers music on hold (or whatever they have)
13:14.52senate1610and keep [default] class for internal and incomming
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13:16.28odenkosI'm not very experienced with dahdi/zaptel but cant you just disable musiconhold in the channel?
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13:18.22senate1610you can, via freePBX and in configs, but doesnt seem to use that, keeps on using default class, I can tell this by checking within the asterisk console viewing debug / call logs, Enabling Music on Hold 'default', even after I have set it to something else. Ill keep digging, but thanks for replying anyway :)
13:18.27senate1610have to run!
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13:26.21TheDavidFactorTSM, I don't anything about sending sip messages from the dialplan. I've used commands to modify the sip headers in the dialplan but never to send a sip message. Hopefully someone else can help you. I can suggest that you pastebin the part of your dialplan where you're trying to send the sip message and then ask for some help. Having something people can look at will be more likely to get...
13:26.23TheDavidFactor...you good help
13:29.21DNDguys, is there any other way to detect hang-ups aside from changing tonezones?
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13:32.03gr0mitDND, using analog lines?
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13:39.58[TK]D-Fender~cds
13:39.59infobot[~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up.  This is typically done either by a momentary battery cut, or by a polarity reversal on the line.
13:40.01[TK]D-FenderDND: ^^^^^^^^
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14:03.16eppigyhello
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14:06.35michael-iI have an analog phone connected to my asterisk system. It routes calls out through another analog port to my telecom provider. Should I have echo cancellation on both ports? Is one or the other FXO/FXS preferred?
14:07.13ManxPower-workmichael-i: the place you need to echo cancel is where the call is converted to VoIP
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14:09.34michael-iManxPower-work: In a mixed environment (voip+analog, trunks+phones) I guess there's not a good best-practice.
14:10.20michael-iOr better said...no optimal solution. I'm going to just be echo cancelling extra channels
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14:11.28Dovidhi
14:11.35Dovidcan anyone help me with this ?
14:11.35Dovidhttp://pastebin.com/m7110e75d
14:11.51ManxPower-workmichael-i: not at all.  The answer is pretty clear.  echo cancel where the call is converted to IP.
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14:13.10ManxPower-workIf the call is analog/analog then you don't need echo cancel.  If the call is voip/voip then you don't need echo cancel.
14:13.15wwalkerI have app_meetme loaded and the dahdi and dahdi_dummy modules loaded, but meetme is still puking with :WARNING[4588]:  app_meetme.c:800 build_conf: Unable to open pseudo device
14:13.30Katty:<
14:13.39Kattycold paws :<
14:13.43Dovidy the sad face ?
14:13.44Dovidlol
14:13.45Kattyforgot mittens at home :<
14:13.53ManxPower-workIf the call is something like PSTN -> VoIP Provider -> Your Asterisk -> IP Phone then guess where the echo cancel needs to happen.  That's right at the PSTN/VoIP conversion at the provider.
14:13.57Dovidu';ll live
14:14.04KattyHOW DO YOU KNOW
14:14.09KattyI MIGHT DIE FROM COLD PAWS
14:14.16michael-iManxPower-work: but sometimes a call will go from analog trunk -> ip phone. Other times it will go analog trunk-> analog phone. In one case I need EC, the other not. Without reconfiguration per call I'll be ECing unnecessarily to cover my bases.
14:14.19ManxPower-workwwalker: that means asterisk does no thave permission
14:14.40michael-iManxPower-work: unless I can configure it on a per-call basis via extensions.conf?
14:14.59ManxPower-workmichael-i: that's why Asterisk does not echo cancel analog/analog calls by default.  "echocancelwhentdmbridged=yes will override that
14:15.13Dovidchances r u wont die. but yes, u may die. i stand corrected
14:15.21Katty:>
14:15.27Kattyi turned my little space heater on :>
14:15.33michael-iManxPower-work: aha! That's probably what I was fishing for.
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14:15.39KattyI MAY SURVIVE AFTERALL!!!
14:15.44ManxPower-workHmm..The power is flickering everytime the blow something up at the army base nearby.
14:15.56ManxPower-work(the army likes to blow stuff up)
14:16.07Kattythey are required to.
14:16.09michael-iManxPower-work: thanks for the tip...I'll look into that
14:16.11Kattyonce a month.
14:17.08Kattyi feel bad for whoever's in charge of checking the ammo in and out. that job is a royal pain in the snickerdoodles.
14:17.36Dovidits fun to blow shit up
14:18.14Kattynot when you have to return all the empty shells.
14:19.06Dovidanyone from Digium here up early ?
14:19.27leifmadsenDovid: what is the actual question
14:20.46Dovidliefmadsen: www.asterisk.org/downloads LibPRI on there is 1.4.10.1 and 1.4.10.2 is out
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14:21.49ManxPower-workDovid: That is typical.  The people that do releases are not the people that put the files on the Digium download site.  Give it a day or two.  Someone will realize the problem.
14:21.58leifmadsenDovid: I haven't seen the libpri 1.4.10.2 announcement, but perhaps it is out :)
14:22.21leifmadsenManxPower-work: actually, the people who put the releases out ARE the people who should be putting out the anouncements, updating the website, and putting it on the downloads server
14:22.35leifmadsenManxPower-work: for example, I do all that for the Asterisk releases, but I don't do DAHDI or libpri
14:22.50ManxPower-workleifmadsen: do you put files up on downloads.digium.com?
14:22.54leifmadsenyes
14:22.58ManxPower-workIs that new?
14:22.59leifmadsenit's part of the release process
14:23.06leifmadsenno, I've been doing that for nearly a year
14:23.15ManxPower-workleifmadsen: that is "recent"
14:23.26leifmadsenin Asterisk time, that is hardly recent
14:23.41leifmadsenI consider the last couple of months as "recent"
14:23.42ManxPower-workI've been using Asterisk for 8 years.  1 year is "recent".
14:24.07leifmadsenwhatever
14:24.14leifmadsenthis is a stupid debate
14:24.18ManxPower-workleifmadsen: Anyway, it's good to know that Digium tried to address the issue.
14:24.32leifmadsenI'm not sure what "issue" you're talking about
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14:25.05ManxPower-workleifmadsen: the issue (in early 1.4, IIRC) was that it would take a day or two for releases to show up for download.
14:25.27ManxPower-workBTW, I'm very impressed with 1.6.  Just started using it.
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14:27.15leifmadsenManxPower-work: well, the process of putting things on the downloads site at release time seems to be sometime prior to June 5th, 2008 (that's when the documentation was created it seems)
14:28.08ManxPower-workleifmadsen: *nod*  I've not really kept up with releases in the last year or so.  Never used 1.4 much, went from 1.2 directly to 1.6
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14:31.29ChainsawManxPower-work: That's a proper upgrade, yeah. I've done that.
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14:33.55nyrangerI am having a weird issue.  when trying to make a bunch of calls using agi and the Async: true flag.  Basically I can reprodue it trying to make 3 phone calls at a time.  THe error I get is:
14:34.11nyranger[Nov  5 01:00:26] ERROR[2583] pbx.c: Received invalid event that had no device IE
14:34.40Dovidcan anyone help me with this ?
14:34.44Dovidhttp://pastebin.com/m7110e75d
14:35.26superbeefT1 troubles?
14:35.51nyrangersuperbeef: no, i am using just IAX2 trunks to my provider
14:36.04Dovidsuperbeef: Could it be the card ?
14:36.48superbeefDavid: hard to tell.... I'm dealing with similar woes right now.. you should enable PRI debugging on your spans
14:38.44Dovidguess i goto do it when i have the issue
14:38.46Dovidnow its wokring
14:38.51Dovidcomes and goes every so often
14:39.06leifmadsenDovid: libpri link updated on asterisk.org
14:39.09Anth8708hey guys, I'm trying to figure out how to have a queue answer without starting with MoH.  I'd prefer the queue answer with announcement, then go to MoH.  Right now, I get 1-2 seconds of MoH and then announce.  Here's my relevant config: http://pastebin.com/m62a37bc3
14:41.52[TK]D-FenderAnth8708: It's not going to.
14:42.19[TK]D-FenderAnth8708: If you want to change that, then "vi app_queue.c"
14:42.23ManxPower-workAnth8708: playback(the-message) before you send the call into a Queue
14:42.37leifmadsenthat's what I do
14:42.50Anth8708[TK]D-Fender:  gotcha.  I might try that and thanks ManxP.  I'll try that first.
14:42.50leifmadsenOr create your MOH so it has the announcement message in it
14:42.53superbeefleifmadsen: oh man.. that libpri update my fix my crazy d-channel dropping problem
14:43.11leifmadsensuperbeef: hawtness :)
14:43.30Anth8708liefmadsen:  * doesn't start a new MoH thread for every call though, does it?  Just feeds the current stream?
14:43.31superbeefyeah too bad I can't change it until tonight
14:43.36Dovidliefmadeson: Thanks
14:43.45superbeefI'll have to recompile libpri then recompile asterisk, right?
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14:44.30leifmadsenI believe so... but I'm not sure
14:44.42leifmadsenAnth8708: I think you can configure that
14:45.00Anth8708leifmadsen:  rgr.  I can look it up.  Thanks man
14:45.02Dovidsuperbeef: http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.10.2
14:45.02leifmadsenAnth8708: but I usually just do the Playback() thing like ManxPower-work said
14:45.08Dovidseems to be the issue I am having
14:45.32LemensTSHey Dovid, what itsp do you work at?
14:46.38Kattymaybe he doesn't want the whole channel to knwo wher ehe works.
14:47.18superbeefman this is a potentially relief, i've been fighting this problem for like 3 weeks.
14:47.19*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
14:47.31Kattysuperbeef: it wasn't the flu was it.
14:47.33DovidLemenTS: Who do we work with or who is my boss ?
14:47.45LemensTSWell he could message me than ;)
14:47.49superbeefKatty: not quite... http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.10.2
14:48.20LemensTSDovid: i talked to you on msn a long time ago, im customgt. I was wanting to add another itsp to my ssystem i remember you had good rates but dont remember who it was you worked for
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14:48.24Dovidsuperbeef: I am wondering why it took so long for this issue to come up
14:48.38DovidLemesTS: PM me
14:49.30superbeefDavid: It might just happen under certain vendors.. I've got 2 other boxes deployed in different parts of the US runnning same hardware, and 1.4.10.1 and they never had d-channel drops,
14:49.33leifmadsenDovid: actually, it's been fixed for quite a while, just the announcement and such were delayed while people were performing testing on an issue in the bug tracker
14:49.58Kattysuperbeef: ohisee :<
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14:51.09Dovidsuperbeef: This PRI by me has been up for a while. It's only recently that there is an issue
14:51.21*** join/#asterisk psilikon (n=joel@static-173-65-4-24.tampfl.fios.verizon.net)
14:51.29Dovidalso i have several other Boxes with this provider with no issues
14:51.34Dovidwhat kind of card r u using ?
14:52.00superbeefDavid: http://pastebin.ca/1638165
14:52.16superbeefDavid: Sangoma A102 PCIe with Hardware Echo Cancelation
14:53.32Dovidsuperbeef: You seem to be having an issue where it's going up and down.
14:54.07Dovidu can call sangoma - u should have support from them - if the line comes up and down it seems that its the carrier's issue
14:55.18superbeefDavid: I put in my identical spare card and the same thing occurs... I've had my carrier do several tests.. The never see the D-Channels drop, and are blaming my equipment
14:55.20Kattyhas done conference calls with herself, sangoma folk, and isp.
14:55.34Dovidsuperbeef: they always will
14:56.16Kattysuperbeef: time for a conference call
14:56.50superbeefI should probably try the lib-pri update first
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15:02.28Katty:>>>
15:02.31Kattyhugs jaytee
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15:04.42tuxcrafterhi all
15:04.59jayteemorning Katty *hugs*
15:05.31Jumpiesup fellas
15:05.36Jumpieanybody heard of vicidial
15:06.55leifmadsenJumpie: many people have
15:06.57leifmadsen~vicidial
15:06.58infobot[vicidial] a predictive dialer available from http://astguiclient.sourceforge.net/vicidial.html .
15:07.03*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
15:07.39Kattyjaytee: any Deliveries this morning?
15:07.41tuxcrafteri have got a call forwarding setup in my asterisk but i am getting failures when i am calling does somebody see someting out of order in my logs:
15:07.45tuxcrafterhttp://debian.pastebin.com/d390bd401
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15:10.54Kattyhi gary
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15:12.07grharryHi, some help required with the echo problem ISDN BRI - HFC-4S and dahdi
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15:15.04Kattyman iam draggin gtoday
15:16.52*** join/#asterisk xuser (n=xuser@unaffiliated/xuser)
15:20.39Kattypaging mister black to the front desk.
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15:24.55p3nguinhmm
15:25.00Kattyhi p3nguin
15:25.02Kattyhugs p3nguin
15:25.05p3nguinmornin'
15:25.57p3nguinHow're things so far today?
15:26.17Kattycold.
15:26.32p3nguinI know!  It's terrible.
15:26.33Kattybrain's a bit fuzzy still..and unresponsive.
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15:26.38p3nguinlow 40s here.
15:26.55Katty:<
15:27.10Kattyit's just gonna get worse too, you know
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15:30.02p3nguinkatty: Yeah, I've been concerned about that.  I might have to get out some long pants pretty soon.
15:33.22Jumpieim not lookin forward to winter :(
15:33.24Jumpiecept for snowboarding
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15:50.44*** join/#asterisk MWE (n=michel@nl06sr01.targetmedia.nl)
15:50.46MWEhi there
15:51.02MWEis there a parameter to stop a meetme after x-minutes?
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15:52.38leifmadsenMWE: I believe you can use schedule=yes if doing it from realtime, or you could use the time limit on Dial() and dial the MeetMe() via the Local channel
15:52.53leifmadsen(the latter option is probably the route you'll need to go)
15:53.04MWEif will use it in realtime
15:53.23MWEschedule= yes but how do I setup the minutes? (where can I say break up after 40 minutes)
15:54.04leifmadsenMWE: I've not used that option, but if you can stop the conference at a certain time, then you can just update the database via func_odbc and calculate what the meeting end time is 40 minutes from now
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15:55.09MWEty leifmadsen
16:00.26*** join/#asterisk fofware (n=chatzill@190.7.25.160)
16:03.44*** join/#asterisk mumtazah (n=anees@203.82.91.103)
16:04.18*** join/#asterisk Faustov (i=user@gentoo/user/faustov)
16:04.52Faustovhi, did anyone manage to solve such situation with wanpipe? http://pastebin.com/m6b4c9f07
16:10.22*** join/#asterisk e4 (n=e4@rrcs-76-79-48-214.west.biz.rr.com)
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16:12.10MWEleifmadsen, do you thing it wil work on ast. 1.4.23.1 (yes I know I saw 1.6.x)
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16:34.17ppcyo
16:36.45*** join/#asterisk HenrikJott (n=info@d83-183-226-89.cust.tele2.se)
16:37.31HenrikJottHi everyone! I have a problem with asterisk and srv_lookup:
16:38.53HenrikJotti have to use srv_lookup because the service provider uses two different servers, and it works almost all the time, but some incoming calls just get an errormessage and never gets to my server. Does anyone have experience of this?
16:39.36HenrikJottAnd it seems to me that the problem appears every time our server does a new lookup..
16:39.57mchouit's not out of the realm of possibilities
16:40.16mchouI've seen several instances of this
16:40.46mchouespecially when backup servers are in different countries
16:41.22mchouI dont think it's an issue with asterisk as much as the itsps
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16:42.11HenrikJottmchou: okay, you mean that the fault is on the other side?
16:42.17mchouyeah
16:42.39mchouitsp screw up their server setups all the time
16:42.55mchounot properly t=synced
16:43.27mchoudifferent dial plans becuse of different countries etc.
16:43.39mchoubecause*
16:44.38mchoulike sometimes they dont even pass the caller ID properly (because srv puts me on a different server)
16:44.50mchoujust lameness all around
16:45.49HenrikJottthat sounds really crappy =) i use swedish itsp "tele2" and i doesn´t have any previous experience with them... but if it´s their servers screwing up i have to switch...
16:46.40HenrikJottand the times it doesn´t work my asterisk server doesn´t say anything in the log at all... i interpret that as it´s not getting any signal...
16:46.47p3nguinYou could submit a ticket and see if they will work on it.
16:47.03HenrikJotta ticket?
16:47.10*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
16:47.18p3nguinsubmit a report to them.
16:47.21*** join/#asterisk Hikah (n=jfoote@83-244-221-226.cust-83.exponential-e.net)
16:47.31HenrikJottah, okay =) well i will!
16:47.32p3nguina complaint
16:47.56p3nguinMaybe they will solve it; maybe they didn't know there was a problem.
16:49.04HenrikJottbut is there any possibility that it´s asterisk who doesn´t really support srv_lookup? because the itsp uses two servers and they clearly state in thier "manual" that we HAVE to use both because some of the incoming calls come from one, and some from the other...
16:49.21mchoulol
16:49.31mchouthat's a bad sign
16:50.17HenrikJotthehe, okay =) i thought it sounded like a good thing at first though...
16:50.26mchouyou only register to one sever at a time :)
16:50.35mchousrv is a good thing
16:50.51HenrikJottbut whats srv_lookup for then?
16:50.53mchouprovides reduancy (anongst other things)
16:51.03TSMim trying to use the PHO
16:51.05TSMwoops
16:51.06mchouamongst*
16:51.24HenrikJottah, okay, so if they change the server it will change for me automatically? right?
16:51.44mchouyup
16:52.47*** join/#asterisk hardwire (n=spencers@216-67-98-253.static.acsalaska.net)
16:52.58HenrikJottalright, but it does not mean that we can use both their servers.... can i use two "register"-strings instead?
16:53.23HenrikJottthat way i can cover them both, they only have two different servers
16:53.27mchouHenrikJott: nono, you need to be very clear on this
16:54.35*** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net)
16:54.39mchouyou only register to one server (at any given time), but calls may come from any one of their servers
16:55.45mchouyou _can_ register to all thier servers but whether you should is a different question
16:55.56TSMim trying to use the PHPAGI script to setup a call between two end points using originate, is it possable to show differnt CID to the caller and callee, at the moment i can only specify a standard cid
16:56.47mchouHenrikJott: some ITSPS dont behave well if you perform simultaneous registrations with same credentials
16:57.32HenrikJottokay... so i should keep just one registration and talk to them about the errors basically....
16:58.30HenrikJottbut if i register to one of their servers and a call comes from another server, will my asterisk even recieve the call?
16:59.00mchouHenrikJott: yes
16:59.23mchouprovided you allow access from other ip addresses
16:59.37mchoufrom the same ITSP
17:00.12HenrikJottas it is right now i allow access to all ip adresses on certain ports... so that should not be a problem..
17:00.34mchouHenrikJott: no, that's a sure way to get hacked
17:00.44mchouhappens all the time
17:00.44HenrikJottbut how does asterisk know that it´s a call coming from the correct itsp and not some other ip?
17:01.06mchouHenrikJott: bingo ! :)
17:01.13HenrikJottyes, i know and i will turn on my firewall as soon as i get things working =)
17:03.04mchouthe key in addition to the firewall rules, is to specify in asterisk to let only certain ips/ports connect
17:03.11HenrikJottmchou: so asterisk will know which calls to recieve?
17:03.41mchouHenrikJott: I dont understand the context of your question
17:04.03HenrikJotthaha, no, it was a bit wierd question i realize now =)
17:04.04mchousuppose your ITSP has 2 servers....
17:04.09HenrikJottyes
17:04.43mchouyou specify in asterisk those are the only two ip addresses allowed to connect
17:04.56mchoufor best security
17:05.11mchoujust an example
17:05.35mchouand you also need other settings too
17:05.36HenrikJottah, okay... where do i specify that?
17:06.06mchouin sip.conf
17:06.20HenrikJottaha!
17:06.55HenrikJottbindaddr?
17:07.03mchousigh
17:07.14mchouread sip.conf.sample
17:07.23mchouit's all explained in there
17:07.30*** join/#asterisk Ad-Hoc (n=nimbus@ppp85-137.adsl.forthnet.gr)
17:08.14Ad-Hochi
17:08.38HenrikJotti will! thanks for all the help!
17:09.45*** part/#asterisk HenrikJott (n=info@d83-183-226-89.cust.tele2.se)
17:10.49*** join/#asterisk IBC_jkenney (n=jkenney@ip65-44-169-66.z169-44-65.customer.algx.net)
17:11.28IBC_jkenneyHas anyone in here played with the SPA 962 and the LDAP directory feature if so.  Did they get it working and if there was a howto for open ldap could you kindly point me to it?
17:16.29*** join/#asterisk ariel_ (n=chatzill@63.214.236.169)
17:24.11Anth8708Guys, is there a variable that stores how long a caller has been on the current call?
17:25.17ryduhYou could what time the call was started in a variabler
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17:27.18[TK]D-Fenderryduh: Talks does funny Yoda hmmmMMMM!?!?!?
17:27.33Anth8708Think I found it.  testing: ${CDR(duration)}
17:28.11Corydon76-digAnth8708: only on 1.6.1 and higher, I think
17:28.40Anth8708funny, i look for 10 minutes before i ask, and then find it within a few seconds of asking
17:28.44Anth8708thanks Cory
17:28.52ryduhoh god. I missed a few words there
17:28.54Corydon76-digOh, it's in 1.6.0, too
17:35.44GuggeI have an asterisk 1.6.1.9 on a debian 5 system, around 50% if all invites sent to it is just ignored, and doesnt even show up in the console with sip debug enabled
17:35.52Guggetcpdump show the incoming invite
17:35.57Guggebut no answer from asterisk
17:36.01Guggewhat could cause that?
17:38.59leifmadsenGugge: ACL setup incorrectly?
17:39.44Guggetheres no ACL's .. and all the calls are from the same IP ....
17:40.08Guggeshould asterisk show blocked calls when debug and verbose is set high ?
17:40.15Guggecause nothing is shown on the console
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18:03.11TSMjust been reading up about software EC, is there any point in using mg2 if you have hardware EC?
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18:10.57Kattyhi!
18:13.49jblackRequiring nickserv id to join #asterisk is annoying.
18:13.59[TK]D-Fenderits AWESOME
18:14.12Kattyyes. yes it is awesome.
18:14.19Kattyi am forced to agree with the awesomeness of it.
18:14.24Kattyhowever it can be annoying.
18:14.33Kattyit's definately worth it
18:15.10jblackOk, guess I"m outvoted.
18:15.27Kattyjblack: think of all the annoying people that would randomly pop in
18:17.42Kattymore like think of all the additional annoying people we're preventing ;)
18:24.03ariel_annoying popin saying Hi everyone....hehehe
18:24.16jblackSome fences are a larger impediment to the locals than the ones being kept out.
18:24.44ariel_that is why they make wire cutters...
18:28.57*** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de)
18:29.09Kattyhehehe
18:29.11Kattyhugs ariel_
18:29.54ariel_hug Katty back, & says t/y
18:30.20Corydon76-digIf people have to learn something before they can ask a question, that puts them in the right mindset for this channel
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18:32.41ariel_is yelling at the stupid spread sheet as it was given to him with the right mac's but they did not seperate the numbers with the ":
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18:33.41LemensTSmacro's :)
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18:33.53Kattyscowls
18:34.04ariel_yes trying to create on the debian box to import them to the dhcpd server
18:34.07Kattywhy do people call me about installing stupid software for them?
18:34.19Kattythe reason YOU are not an admin is so that software does NOT get on my network
18:34.21ariel_because your the best
18:34.31jefimenkodoes the lastest version of asterisk support SIP session timers? i can't find docs on it
18:34.45Kattyi don't /care/ if you want to transfer pictures from your blackberry to the computer. do it at home!
18:34.58Kattygrumbles.
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18:35.50IBC_jkenneyKatty can you help me copy company sensitive data to my android phone?  PLEASE =)
18:35.56Nuggethttp://icanhascheezburger.com/2009/11/05/funny-pictures-why-do-you-ask-2/
18:35.58*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
18:36.01KattyIBC_jkenney: :P
18:36.18IBC_jkenneyyou know how i fixed that question
18:36.32Kattythe sad part is if i tell them no, they go whine to the owner of the company about it. turns into lots of Drama(tm)
18:36.35IBC_jkenneyi "accidently deleted the stuff off their phone"
18:36.38IBC_jkenney:)
18:36.45IBC_jkenneyfor some reason nobody asks now
18:37.03KattyNugget: that's cute.
18:37.16KattyIBC_jkenney: that would be a whole crapload of drama really quick here.
18:37.21KattyIBC_jkenney: it's like the admin, but i'm not.
18:37.38[TK]D-Fender"We can make it look like an accident..."
18:37.42IBC_jkenneyhug katty
18:37.53IBC_jkenneyright on Tk thats what i am saying
18:37.59Kattyi'm on the admin if something is broken.
18:38.07Kattys/on/only/
18:38.27Kattyotherwise, i'm just here to help you do your personal crap all over the network.
18:38.41IBC_jkenneyHas anyone in here played with the LDAP feature on the SPA962's
18:38.56IBC_jkenneythat lets you hook to a SPA to ldap for the directory
18:39.03ariel_ohh that reminds me I need to start working on Ldap for our asterisk setups....
18:39.20Kattyi have ldap work to do too :/
18:39.23jefimenkohello
18:39.42IBC_jkenney<=== Has ldap running and working
18:39.55IBC_jkenneyi use mysql for the phones though
18:40.00Kattyi just have to setup our new email filter providor with our ldap users.
18:40.37ariel_argh, I have to setup a complete LDAP for our ships/users and clients systems....argh
18:40.56Katty:<
18:44.59jefimenkodoes the latest asterisk release support session timers? i see mixed information about it on search engines
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18:47.46*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
18:48.11[TK]D-FenderGah
18:48.52russellbjefimenko: Asterisk 1.6.0 and later does, yes.
18:49.56*** join/#asterisk Rogier21 (n=rogierve@rogierversluis.xs4all.nl)
18:49.59PanamanI'm doing infosec for a company and am trying to help out the Asterisk admin who has been charged with encrypting conversations. Can someone point me in the right direction for implementing SRTP (or whatever gets the job done) in Asterisk please? He's running 1.6 iirc...
18:50.00eppigyKatty: on the admin
18:50.26Rogier21evening all
18:50.42Rogier21does somebody have a complete SQL file for mysql for all tables for 1.6.1?
18:50.52Rogier21or atleast where i can find the latest version
18:51.32jefimenkorussellb: are there docs for it? i can't find any
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18:58.36alexshellhi all
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19:01.06*** join/#asterisk Rogier121 (n=rogierve@rogierversluis.xs4all.nl)
19:01.56alexshellmay I have multiple networks in "localnet" option in sip_nat.conf file?
19:02.56alexshellthe problem is that I have two branch offices VPN connected
19:03.33*** join/#asterisk Tim_Toady (n=moi@adsl58-161.kln.forthnet.gr)
19:04.33alexshellwhen I defined externip and only one localnet, it appear that branch offices have problems with calls
19:04.41[TK]D-Fenderalexshell: Yes, and that file is not specifically associated with *
19:05.05alexshellIf I remove everything in sip_nat.conf everything works perfectly
19:05.10*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
19:07.58alexshellty [TK]D-Fender found the answer http://www.voip-info.org/wiki/view/Asterisk+SIP+localnet
19:08.56Rogier121does anyone else get this error?
19:09.05Rogier121res_config_mysql.c:1169 require_mysql: Table sipfriends requires a column 'useragent' of size '20', but no such column exists.
19:09.32Rogier121i copied the SQL code from the docs
19:11.12*** join/#asterisk |Cybex| (n=John@atwork-20.r-212.178.82.atwork.nl)
19:11.14Naikrovekwhat's everyone's opinion on bandwidth.com
19:15.21ariel_I don't use bandwidth.com or really know much about them.  So I can't really say much other then just nothing....;-)
19:16.06*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:17.21umaybandwidth recently stopped monthly commission checks for new orders and fired everybody in that department
19:17.44*** join/#asterisk Anth8708 (n=Anth8708@client105.jdcc.edu)
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19:19.26ariel_wow
19:22.06umaythey also install black box edgemarc routers in your network to bring in the SIP trunks but leave default root passwords set and dont let you change it
19:23.07*** join/#asterisk fofware (n=chatzill@190.7.25.160)
19:27.50Naikrovekreally
19:30.54Naikrovekwhat about for bandwidth from them, say a t3
19:32.41*** join/#asterisk ruben23 (n=RPL@122.55.48.243)
19:34.38Naikrovekstill an edge router they put in?
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19:45.04Panamank. thx.. bye
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19:47.10ChaytanHey guys.. please can someone tell me where I can get the Asterisk Tunnel Module (chan_tunnel.so) from ?
19:47.51leifmadsennever heard of it
19:48.03leifmadsenit probably doesn't exist
19:48.08Chaytan:( dammit
19:48.15Chaytanhttp://docs.voipswitch.com:8080/display/VDoc/Asterisk+Tunnel+Module+-+ATCOM+IP04+PBX+Installation+and+Configuration+Guide
19:48.33*** join/#asterisk sjobeck (n=Adium@valdisere.sjobeck.com)
19:48.43ChaytanI have this link, and we recently installed Asterisk, and i needed to do such a config.
19:48.46netpro25_Hey I am getting voice distortion on some incoming calls and a strange buzzing sound on some voice mails. I just upgraded to 1.6.2. Any ideas?
19:48.58Chaytanexcept i've been googling everywhere but can't find anything at all
19:48.59ChannelZSquirrels
19:49.19leifmadsenthat is just a login page
19:49.53leifmadsenChaytan: I did a search for chan_tunnel and only 2 pages show up, and they are just a discussion about such a channel; I'm pretty sure it doesn't exist
19:50.06umayNaikrovek: have only dealt with bandwidth.com's sip trunks, sorry
19:50.14Chaytanleif, sorry try login: reader | pass: voipswitch1
19:50.17Naikrovekthat's cool
19:50.33Chaytanleif yes, that's my impression as well.
19:51.50leifmadsenChaytan: have you emailed the guys who made that page? It looks to be something either proprietary, or not publicly released
19:52.05netpro25_I am getting voice distortion on some incoming calls and a strange buzzing sound on some voice mails. I just upgraded to 1.6.2. Anyone?
19:52.18leifmadsennetpro25_: what version of 1.6.2?
19:52.30netpro25_1.6.2.0~rc2-0ubuntu1
19:52.38leifmadsentry 1.6.2 branch
19:52.45Chaytanyes i did, except they're taking their sweet time to reply... so i thought i'd take a chance here
19:52.46leifmadsen1.6.2.0-rc2 is kinda old now
19:52.58kaldemarand it's a release candidate...
19:53.01netpro25_okay
19:53.05leifmadsenand that :)
19:53.20leifmadsenbut still, if its happening in 1.6.2 branch, we should have an issue filed so it can be fixed
19:53.21kaldemarjust because it's in ubuntu's repo, doesn't mean you should use it. :)
19:53.31leifmadsenRCs are still useful for people to test though
19:53.36kaldemarsure
19:53.42leifmadseneveryone gets up in arms when a release has a bug, but no one tested the RCs :)
19:54.23kaldemarit's the mentality of "let other people do the testing" :)
19:56.27*** join/#asterisk moy (n=moy@74.12.130.190)
19:56.29ruben23hi are there any startup script for dahdi on slackware..?
19:58.04*** join/#asterisk Methose (n=Methose@38.101.237.250)
19:58.09sjobeckleifmadsen is in the hizzouse
19:58.13sjobeckleif, how you ?
19:59.07[TK]D-Fenderruben23: /etc/rc.d/rc.local
19:59.47leifmadsensjobeck: zup zup! just workin' away
19:59.50leifmadsenkeepin' bizzay
20:00.25sjobeck<PROTECTED>
20:00.42sjobeckvancouver or toronto or elsewhere today?
20:01.01sjobeckor whereabouts  = unknown
20:01.08sjobeckyeah, lets go with that
20:01.12sjobeckok, back to work
20:01.29leifmadsensjobeck: heh, I'm in Toronto :)
20:01.47*** join/#asterisk trentcreek (n=trentcre@129.113.43.116)
20:02.17MethoseCan someone recommend a hardware appliance for an IP PBX based on Asterisk, that can be purchased with a 24/7/365 support?
20:02.46leifmadsenMethose: I think Switchvox fits that bill
20:03.34*** join/#asterisk johnakabean (n=john@static-173-50-101-13.nrflva.east.verizon.net)
20:03.42Kattyscreams
20:03.44Kattythrows things
20:03.45Kattycries
20:03.52MethoseSIP trunking is the only real need, so no analog cards needed
20:04.38Methoseleifmadsen: thank you
20:05.08leifmadsenMethose: no problem! I'd check with Digium about the support stuff. I know they have the support center, but not sure how that works in with Switchvox, which might have a separate support structure
20:05.10leifmadsenhttp://www.digium.com/en/supportcenter/asterisk.php
20:06.07[TK]D-FenderMethose: Now when you say "appliance" are you expecting to have control over the back-end of your server?
20:08.47Methose[TK]D-Fender: by appliance I mean a hardware device, that is specifically designed for and dedicated to the functionality of the device
20:09.21*** join/#asterisk jelly-bean1 (n=jelly@75-148-103-190-Utah.hfc.comcastbusiness.net)
20:09.31[TK]D-FenderMethose: Well I'm pretty sure you'll void your support if you playing around outside the scope of their GUI, etc
20:09.42[TK]D-FenderMethose: As in "if you want to do it yourself"
20:09.57johnakabeanfender, how can I set a variable to the last number called by an extension?
20:10.20[TK]D-Fenderjohnakabean: You need to set that value whenever they place a call in the first palce
20:10.29[TK]D-Fenderjohnakabean: and I'd recommend AstDB for this
20:10.33johnakabeanyes, i'm aware of when to set it
20:10.43Methose[TK]D-Fender: I'm fine with not voiding my warranty and letting the vendor fixing it as long as the support agreement is acceptable
20:10.44johnakabeanbut what variable?
20:10.44[TK]D-Fenderjohnakabean: "core show function DB"
20:10.50[TK]D-Fenderjohnakabean: NOT A VARIABLE
20:11.04[TK]D-Fenderjohnakabean: use AstDB so its persistent
20:11.24[TK]D-FenderMethose: And as long as the product suits your needs
20:12.24jelly-bean1why are the queues not aware of each other? if i have an agent logged into three queues: Sales Eng, Sales Spanish, Tech Support--it does not work well. They can get a call on A, then get a call on B--and B will not realize the agent is already on A. system will log them out of B for missing a call. whats up with that?
20:12.32eppigyholds Katty close
20:12.38Methose[TK]D-Fender: the needs are extremelt simple. IP PBX that does SIP trunking, takes calls in via 729 or 711 and transfers them to an internal extension
20:12.53Methoses/extremelt /extremely
20:13.15MethoseSIP phones internally, so no analog needed at all
20:15.48Kattyeppigy: i are having bad day :<
20:16.12Kattyeppigy: it may require i go home early, and unwind :<
20:16.14*** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net)
20:16.25Kattyeppigy: or possibly hack some people up with a chainsaw.
20:16.29Methoseoh I guess one more requirement: works in London, and the vendor has a POC there or is able to support it in the UK
20:16.32Kattyeppigy: but that'd be messy.
20:16.38*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:17.00ayesoanyone know of a way to see how many calls are active and what context they are active in?
20:17.12Kattysip show channels
20:17.14Kattyor sip show users
20:17.19Kattyi think
20:17.23ayesoKatty: doesnt show the context tho
20:17.53Katty^_-
20:18.14Kattymine sure does.
20:18.42ayesoKatty: sorry, i mean what context in extensions.conf the active call is currently in.
20:19.07Methose[TK]D-Fender: do you know if Digium has presence in the UK?
20:19.47Kattyhas a presence?
20:19.56Kattyas in, haunting?
20:20.09Kattyi can see it now, digium with a sheet over their head.
20:20.13filehaunts Katty
20:20.14Kattyfrolicing about the streets of the UK
20:20.22Kattyhugs file
20:20.23[TK]D-FenderMethose: They might have sub-contractors...
20:20.25[TK]D-Fenderat most
20:20.31filehugs Katty
20:20.44Kattysurely they sell internationally
20:20.53MethoseKatty: as in an office or a place that they hang their hat while they work on peoples broke shit in the UK?
20:20.57[TK]D-FenderKatty: SUPPORT
20:21.04Katty[TK]D-Fender: that's what i meant.
20:21.18[TK]D-FenderKatty: He's expecting 24x7x365.25 w/ physical presence
20:21.24Kattyoh.
20:21.28[TK]D-FenderMethose: Good luck with that
20:21.30Kattygood luck with that
20:21.38Katty[TK]D-Fender: get off my brain wave.
20:21.40[TK]D-Fenderhugz on teh Katty
20:21.47[TK]D-FenderKatty: I was here first!
20:21.51[TK]D-FenderKatty: I'll SUE!
20:21.53Kattydon't make me throw things
20:21.54[TK]D-Fenderlitigates
20:21.57Methoseheh
20:22.03Kattyprosecutes.
20:22.11Kattythrows popcorn at [TK]D-Fender
20:22.24Kattyso. has anyone heard of MXLogic?
20:22.39[TK]D-Fenderpulls out his water-gun and newspaper
20:22.43Methoseso, no go for any Asterisk based IP PBX appliance with 24/7 support in UK, huh? :(
20:22.44Katty:<
20:22.59Nuggetit's all got 24/7 support if you count irc.  :)
20:23.02KattyMethose: oh i'm sure there's some digium partner out there.
20:23.06Nuggetand I'm only half-joking
20:23.11arekmhi, when someone dials number then he have silence in earphone for some time and then dialing signal appears. Is it possible to put somthing other than silence in the beginning? (people think that nothing is happening while it takes some time)
20:24.01KattyNugget: depends on the type of support eh? ;)
20:24.06QwellKatty: I'd totally haunt you.
20:24.18Kattywhat? where did this come from
20:24.28Qwell<Katty> has a presence?
20:24.28Qwell<Katty> as in, haunting?
20:24.37Kattyoh, that. that was forever ago.
20:24.45QwellI was outside
20:24.46Methoseindeed
20:24.52QwellI can't be here 100% of the time!
20:24.56Kattyhaunting the streets of UK with a sheet over their head?!
20:25.07Katty<qwell> holy ghosttttttttt holy ghosttttt!!!!
20:25.18KattyQwell: this is not an episode of scooby doo
20:26.42*** join/#asterisk ddickenson_ (n=chatzill@67-198-0-5.static.grandenetworks.net)
20:27.11jayteemaybe the Hokey Pokey IS what really is all about
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20:27.40Kattythat's only fun if you're on skates tho.
20:28.36ddickenson_can anyone explain or point to a tutorial that tells the best practice for setting up BLF on an asterisk 1.6 system?  Everything I find on the net has to do with 1.2 or 1.4 and they all talk about patched methods or workarounds.  Also where does devstate figure into it? Do you need that along with the hints to make it work?
20:29.16TheDavidFactorWhat are you trying to do with BLF? You don't normally need devstate
20:29.37ruben23how do i check if dahdi is running upon after my reboot, using slackware distro
20:29.39*** join/#asterisk thehar (i=thehar@thehar.xmission.com)
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20:30.01ddickenson_trying to set up a semi key system functionality for administrators and their secretaries, I however do not need them to be actual SLA trunks (although a directed call pickup would be nice)
20:30.11ddickenson_so they can see who's on phone etc
20:30.14*** part/#asterisk sjobeck (n=Adium@valdisere.sjobeck.com)
20:30.55Kattyruben23: does slackware have a ps aux?
20:31.38ruben23<PROTECTED>
20:31.46ddickenson_TheDavidFactor: also endpoints I'm using are either going to be Polycom ip650's for secr. and cisco 7960's for admins or polycoms all around
20:32.01ruben23Katty: are there any command to run it up
20:32.35Kattymodprobe mayhaps?
20:32.37Kattynot familiar with slackware.
20:33.45Kattydahdi has some tools
20:34.11ruben23<PROTECTED>
20:34.12TheDavidFactorddickenson_, basic BLF functionality is handled by the subscribecontext= in sip.conf; and the 'hint' priority of extensions in extensions.conf Read http://www.voip-info.org/wiki/view/Asterisk+presence for a decent overview
20:34.34Kattyuhh
20:34.37Kattydahdi-tools
20:34.42Kattyyou ./configure it
20:34.47Kattymake, make install, make config
20:34.50Kattyyou know. tools.
20:35.21ddickenson_TheDavidFactor: ok, I believe I have those set up already but I'll read over that and see what I'm missing. Thanks
20:35.30Kattythinks
20:35.40Kattyztcfg, ztmonitor, ztscan, ztspeed, uhhh
20:35.49Kattythere's a zap generator
20:35.57Kattyand dahdi_tool of course
20:36.04[TK]D-Fenderruben23: DAHDI isn't a running process
20:36.11[TK]D-FenderKatty: CRAZY-TALK
20:36.12TheDavidFactorThen you point your phone's BLF to that extension and the pixie dust does the rest
20:37.01Kattyguess those were the old names
20:37.16Kattydahdi_cfg, dahdi_monitor, dahdi_scan, dahdi_speed, dahdi_test, dahdi_tool, dahdi_genconf
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20:42.13*** join/#asterisk Nockian (i=hidden-u@unaffiliated/nockian)
20:43.26Nockiani'm using asterisk with voicepulse to dial out, and it's working fine except when i try to dial an 877 number i get a message almost immediately that says "the number you are calling is temporarily unavailable"
20:43.44QwellNockian: Have you asked your provider why?
20:43.46Nockianany ideas on where to start looking to debug it, or is it most likely a problem with voicepulse somehow?
20:44.08NockianQwell: no, i haven't opened a ticket with them yet
20:44.25Nockianso it's most likely on their end, if i'm handing it off properly (and i believe i am)
20:44.41trentcreek~book
20:44.42infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
20:46.56bwcook0i am having a problem with the piaf and the google voice setup.  the inbound route gv-ringback rings my xlite client just fine when i set its destination to ringroup 700.  but when I set the destination to gv custom-park as is needed for this to function, I see asterisk moving it to the gv-incoming route and trying to dial the 700 ringgroup but the client never rings
20:47.08bwcook0i put the asterisk debug out in http://pastebin.com/m1a26e44a
20:49.17eppigyKatty: yes I recommend the first option
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21:02.50trentcreekbwcook0: wrong channel. YOu need to go to the PIAF site
21:03.00trentcreekor GreePBX
21:03.05trentcreekoops.. FreePBX
21:03.20bwcook0k thanks
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21:12.53p3nguinkatty: http://i35.tinypic.com/i39mit.jpg
21:14.36Kattythat's cute.
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21:16.05russellba usb powered pistol sounds wimpy.
21:16.10leifmadsenmeh... Colt .45 ftw
21:16.17ppcIt's USB 3.0
21:16.30leifmadsenas features in Duke Nuken Forever
21:16.46leifmadsenas featured in Duke Nukem Forever
21:17.38*** join/#asterisk bpgoldsb (n=bpgoldsb@gw.teamgleim.com)
21:18.11[TK]D-Fenderleifmadsen: Springfield Armory XD-M
21:19.09[TK]D-Fenderleifmadsen: combines the best features of the Glock and Sig Sauer line with a double-stack .45
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21:28.50[TK]D-Fendercheckout time, later all
21:29.23rdude99Hey guys, simple problem from a noob: I can't get the "invalid" extension to work.
21:29.34rdude99here is my dialplan: http://pastebin.ca/1658749
21:29.48rdude99basic, but I get a 404 from my sip client
21:30.27rdude99The user's context is set correctly 'cause the 123 extension works
21:30.46Qwellrdude99: i exten doesn't work like you think it does.  It's not a catchall
21:32.04rdude99Qwell, what's the correct way to use it? I was going off of this page: http://www.voip-info.org/wiki/view/Asterisk+i+extension
21:32.22QwellYou would think that Asterisk would automatically jump to the 'i' extension if the client dials a number that is not matched by any other extensions. But it doesn't. Take this simple example:
21:32.30Qwellright from the page you linked...
21:33.16*** join/#asterisk p3nguin (i=BuhPjX1J@asterisk-klv5.a2infotech.com)
21:34.45rdude99Qwell, lol; I guess I misread that paragraph. I thought the example was showing the correct way to do it. Sorry...but I warned I was a noob so that's my excuse :)
21:40.34*** join/#asterisk torrancew (n=torrance@btr.macconsultinggroup.com)
21:40.44p3nguinrdude99: Have some sound file play backgrounded or use WaitExten after answering 123.  Enter in some invalid extension (anything, really), and see if 'i' doesn't execute.
21:41.27rdude99p3nguin, ok, will try
21:42.38*** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1)
21:45.11rdude99p3nguin, ahh ok; "i" only works after a call is in progress?
21:45.58p3nguinWell, I don't know how to answer that.  It works when you have entered an invalid exten.
21:49.18rdude99ok, I understand what you're saying. Next question:
21:50.58rdude99I want to execute a program that does some other stuff as a step in the dialplan. I see I can use the System() command. However the documentation does not specify return values with surety.
21:51.09rdude99page in question: http://www.voip-info.org/wiki/view/Asterisk+cmd+System
21:51.53rdude99It lists 3 possible return vals; one is "undocumented" (APPERROR), the other is a "not sure" when it will be returned (FAILURE)
21:52.01*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
21:52.37rdude99Is there a definite resource in processing the return val?
21:52.50*** join/#asterisk gardo (n=gardo@110.55.242.67)
21:53.02p3nguinI think I would rather use an AGI script.
21:53.50kaldemarrdude99: you better learn to read app documentation from asterisk itself. core show application System will print you the documentation for your version. wiki is often filled with outdated and invalid information.
21:53.59p3nguinYou could put it in a GoSub, that way you control what happens when it's done; it would be able to return back into the dialplan or it could do something else.
21:54.06kaldemarp3nguin: absolutely no need for AGI.
21:54.22kaldemarrdude99: with func SHELL you can even save system command output to a variable
21:57.31rdude99kaldemar, where can I get docs for shell? core show application shell yields "Your application(s) is (are) not registered". Saving the output to a variable seems the best solution
21:57.45rdude99vs. messing with the return value
21:58.04kaldemarrdude99: it's a function not an application, hence "func SHELL". core show function SHELL
22:00.00rdude99kaldemar, thanks...got it. also just learned it's case sensitive. I think I have enough to go play around now. thanks kaldemar & p3nguin for the help.
22:02.49*** join/#asterisk [TK]D-Fender (n=joeblow@161.216.171.140)
22:19.13Kattystares at clock
22:19.25theharalso stares at clock
22:19.27Kattypouts
22:19.53theharindeed
22:20.21Kattythehar: what do you wish you were doing right now
22:20.27*** join/#asterisk TimToady_ (n=moi@adsl322-163.kln.forthnet.gr)
22:20.41theharI wish I were drinking.
22:20.47theharNot working.
22:20.50Kattythat sounds pretty fun.
22:20.51ryduhKatty: better question: why aren't you doing what you wish to be doing?
22:20.51*** join/#asterisk voipmonk (n=voipmonk@69.172.70.23)
22:21.13Kattythehar: what kind of drinks do you intend to drink
22:21.35theharKatty: delicious IPAs will be in my belly.
22:21.45Kattywhat's an IPA
22:21.45theharKatty: or belgian beer
22:21.49theharit's beer
22:21.51thehartasty beer
22:21.58Kattyoisee.
22:22.06Kattynot a big fan of Beer(tm)
22:22.13Kattycider's not too shabby tho
22:22.31theharI'll make you some delicious alcoholic cider.
22:22.40Kattythat sounds grand.
22:22.51jefimenkowhat about wine?
22:23.07thehari could also enjoy some wine.
22:23.14Kattyi enjoy the inebriated giggles.
22:23.15theharPerhaps a pinot
22:23.24Kattybleh, wine.
22:23.27jefimenkogrigio or noir?
22:23.28Kattynot sweet enough.
22:23.34theharwhat are we eating with our wine?
22:23.42Kattyi hope italian food
22:23.52jefimenkopizza
22:23.58theharwhat is on our pizza?
22:24.00jefimenkored sauve
22:24.01Kattyyou're supposed to drink beer with pizza
22:24.01jefimenkosauce
22:24.08theharKatty: has it down ;)
22:24.18theharbeer + pizza = heaven
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22:24.19Kattydon't you boys know anything
22:24.31Kattyhi tzafrir
22:24.41jefimenkoonly in italy
22:24.50jefimenkoonly in italy is wine + pizza forbidden
22:24.55tzafrirhi
22:25.09jefimenkobut there are lots of great pairings
22:25.16Kattyeverytime i hear italy i think of Grapes, Olive Branches, and The Boot(tm)
22:25.46thehardas boot
22:25.50Kattypoor sicily :<
22:28.17Kattywith the right device, you can set off a bomb with the speaker of a cellphone wired up.
22:28.18*** join/#asterisk gmarsh (n=gmarsh@mobile-166-137-132-160.mycingular.net)
22:28.46theharorly
22:29.28Kattynot that i know anything about sodium nitrates and diesel.
22:29.59Kattyor that ammonium nitrate actually works better.
22:31.14Kattyi just like chemistry.
22:34.23Kattyyou gotta try iodine and alluminum :>
22:35.25Kattybut becareful.
22:35.38Kattyit can get a little...uhh...smokey.
22:36.06[TK]D-Fenderis Iron Chef of the Anaarchist's Cookbook
22:37.17jplcmmmm, smokey...
22:37.22jplcthinks of ham
22:37.36Kattythink.... purple fumes and fire
22:37.43*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
22:38.59jplcsaltpeter, used in preservation and flavoring and some sulphur and charcoal and you have a nice evening of fireworks with your dinner ;)
22:39.01*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
22:39.13jplcjust not um on your dinner
22:39.50Kattyidly wonders if aluminum foil would work
22:40.46Kattydo you believe they use sodium phosphate to cure ham?
22:41.57*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
22:42.35Kattyhi _ShrikE
22:43.49*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
22:44.21[TK]D-FenderKatty: Sodium tri-iodide FTMFWY
22:45.16jplcKatty: potassium and sodium salts both i believe
22:45.52_ShrikEHey Katty
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22:46.49*** part/#asterisk gmarsh (n=gmarsh@mobile-166-137-132-160.mycingular.net)
22:48.20[TK]D-Fenderjplc: Nope.  They are both atoms, and metals
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22:48.33jplcgoing the other direction, water + salt + magnesium + a little iron heats a soldier's (or a lazy camper's) food on the go
22:49.08jplc[TK]D-Fender: should i have rephrased that to "the salts of potassium and sodium" ? ;)
22:49.24[TK]D-Fenderjplc: Yes.. a "salf of sodium" is a salt...
22:49.40[TK]D-Fenderjplc: What colour was Napolean's white horse again? :p
22:49.45[TK]D-Fendersalt*
22:50.12outtoluncred?
22:50.23jplcdunno, it was smaller than him so i couldn't see...
22:51.39*** join/#asterisk rdude99 (n=rdude99@adsl-76-192-188-201.dsl.chcgil.sbcglobal.net)
22:52.04jplcwonders how napoleon would have looked on a shetland pony >:)
22:53.49rdude99Quick question: is it possible to monitor an active SIP session to the console through /dev/dsp (so that the call can be heard in realtime)?
22:54.25russellboriginate Console/dsp application Chanspy SIP/somephoneyouwanttohear
22:54.28russellbat the *CLI>
22:54.36russellbsomething like that
22:55.05theharoriginate Console/dsp application ChanSpy SIP/RussellB
22:55.12*** join/#asterisk Malkor (n=marco@hlle-d9ba026a.pool.mediaWays.net)
22:55.32theharoriginate Console/dsp application ChanSpy SIP/RussellB@sexytimes
22:55.37russellbO.O
22:55.44russellbthehar: in yer dreamz!
22:55.47theharhahahahahahaha
22:55.48thehar<3
22:56.12rdude99russellb, thehar: thanks
22:56.20russellb<3
22:56.39theharrussellb: i'm going to gamble and drink this week away tonight
22:57.14russellbyay!
22:57.24russellbI am not, as I have another day of work!
22:57.40thehari'm taking pto
22:57.46russellbcool.
22:57.52russellbtake a shot for me
22:58.09theharyou know it. ;)
22:58.12theharand then some
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23:09.48rdude99Is it possible to have a SIP call in progress but play non-conversation sounds on the console (which are based on the call)?
23:10.10rdude99e.g., like a "Call in progress for 12 minutes"
23:10.32*** part/#asterisk Malkor (n=marco@hlle-d9ba026a.pool.mediaWays.net)
23:13.21leifmadsenhas anyone used GoSub() in the latest 1.6.2 recently?
23:13.42leifmadsenI'm having an issue where Return() doesn't work for me, and I'm trying to see if it's something stupid I'm doing, or if it's a regression
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23:26.09waKKuhi folks..
23:27.02waKKucould someone check this post and try to help me why this don't work (i can't hear/talk - i've already read sip-transmissions.txt): http://pastebin.com/m2baff986
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23:32.49jplccan someone recommend an iax2 friendly provider?
23:33.13[TK]D-Fenderjplc: Do you require IAX2?
23:33.27jplcyep
23:33.48[TK]D-FenderjpWhat for?
23:34.08jplcincoming 800 number and outgoing to usa
23:34.33jplci've been with vitelity since they were sixtel and one of my clients is using them as well.
23:34.43[TK]D-Fenderjplc: No, why IAX2?
23:35.23jplcoh, dynamic ip, the whole sip nat traversal thingy and my client's still using 1.xx
23:37.04waKKu[TK]D-Fender please, a quick view at that post? ... you helped me yesterday with the same problem, but the  changes (nat=no) doesn't solve it totally... i can call, but can't hear/talk ;/
23:37.06johnakabeanfender, how would you define a hangup extension only when internal extensions hangup?
23:39.03*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
23:39.04[TK]D-Fenderjohnakabean: it has nothing to do with the device, but rather what context its in.  So make more contexts and keep things separate
23:39.04[TK]D-Fenderjplc: 1.xx what?
23:39.11jplc* 1.xx
23:39.32[TK]D-Fenderjplc: And * works just fine from behind NAT unless your router is absolute garbage
23:39.41[TK]D-Fenderjplc: ... * I AT 1.xx
23:40.03[TK]D-Fenderis8
23:40.04[TK]D-Fender*
23:40.06[TK]D-Fendergah
23:40.30jplchehe. i get ya. but yep, that box has been working fine for the last 4 or so years
23:41.30jplcrouter might be garbage but i know iax works fine through it
23:42.00jplcrouting an iax  call from my server to that box works.
23:42.40jplcbut vitelity says it's not failing on their end
23:44.19*** join/#asterisk tzafrir__laptop (n=tzafrir@212.179.75.202)
23:45.19[TK]D-Fenderjplc: IAX2 can be incompatible between different versions and have all sorts of fun "hissups"
23:45.25[TK]D-Fenderjplc: Use IAX2 if you NEED to.
23:45.40russellbit's not incompatible between any 2 versions.
23:46.07[TK]D-Fenderrussellb: I recall numerous providers with issues mixing 1.2 & 1.4
23:46.08waKKui've noticed that SIP provider (destiny call) is sending a LOT of packet to my * via RTP but it isn't answering...
23:46.22russellbthere were no protocol changes between 1.2 and 1.4
23:46.25jplcwhich is exactly what this is. i'm setting up a sip test account right now just to prove vitelity wrong
23:46.34russellbthere were iax2 stability problems in early 1.4 versions, but that's all cleared up now
23:46.53russellbif there are 1.2 vs 1.4 issues, nobody has told in me in the years that 1.2 and 1.4 have been out :-)
23:47.03russellbwhich would be very disappointing.
23:47.24jplchehe i've been using 1.1, 1.2 and 1.4
23:47.37[TK]D-Fenderrussellb: Life is full of disappointment :)
23:47.40jplcno show stoppers for me so far
23:47.49russellb1.1?
23:47.53russellb1.0?
23:47.55jplcyeah, long story
23:48.04russellb1.1 didn't exist, heh
23:48.11jplcor 1.0
23:48.28jplchmm, the rpms i compiled say 1.1
23:48.32russellbo.O
23:48.37jplcfrom 20050529
23:48.41russellbwell someone made that up :-)
23:48.50jplcmaybe it was me lol
23:48.51jplcmy bad
23:48.54russellbit's all good
23:48.56jplcdon't remember anymore
23:49.02russellbyou probably had some development tree code between 1.0 and 1.2
23:49.36jplci guess. and when i got that client, it was the working version
23:51.44russellbnods
23:51.51russellba ton of people were running the dev tree code around that time
23:52.21russellbfor features like realtime
23:53.54jplcheh, i can't remember what i compiled into this version...must be in my archives somewhere
23:54.23jplcbut lemme see...it's a calling card setup using a heavily mutated version mark's astcc
23:54.39jplc*of

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