IRC log for #asterisk on 20110515

04:20.21*** join/#asterisk infobot (~infobot@rikers.org)
04:20.21*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.4 (2011/05/10), 1.6.2.18 (2011/04/26), 1.4.41 (2011/04/26), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/
04:47.00_pllFreeaqingme: Manager doesn't see/detect the dtmf.
04:47.49Freeaqingmedid it register other dtmf events?
04:47.49_pllNo dtmf events are generated during ringing.
04:48.12Freeaqingmewhat kind of channel is it?
04:48.28_pllIt's a sip channel.
04:49.04_pllThere is an analog conversor using gsm chips.
04:49.19_pllThe setup is... not normal.
04:49.26FreeaqingmeI'd guess that there's your problem
04:49.36_pllI will try through the E1 line.
04:50.07Freeaqingmehow do you know btw there's dtmf tones?
04:50.13Freeaqingmewouldnt be a first time specs were off
04:50.15Freeaqingmeor outdated
04:52.07_pllE1 line doesn't detect them either.
04:53.05_pllIn fact, I don't see any dtmf event.
04:53.39_pllWhen was the dtmf event option added? This is a Asterisk 1.6 instalation.
04:54.44Freeaqingme_pll: easiest is to just check using a softphone & sip
04:55.26_pllThat's the issue, I don't know if they are dtmf tones. If you hear this recording http://www.mediafire.com/file/p3mbv96g561c5cj/tones.wav Their volume is way below the ringing tone.
04:56.16Freeaqingmedunno
05:23.08_pllHmm, it doesn't send dtmf events. I tried it with pjsua
05:23.48_pllThis is all I allow in the manager read = call,dtmf
05:24.05_pllI can see call events but not dtmf events.
05:25.35_pllI send dtmf events through pjsua and they aren't broadcasted in the manager.
05:27.10_pllFreeaqingme: Do you think it's feasible to write a module to detect those 3 tones using audiohooks?
05:27.51_pllI mean, how hard can it be? It would look better in the dialplan too.
05:41.01Freeaqingme_pll: I dunno, I've only been using asterisk for a few weeks
05:41.04Freeaqingmeand nothing related to this
05:41.37_pllAh.
06:10.59_pllGood night.
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06:16.28WiretapMacAMI is far too much fun to play with :P
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08:10.56lgbrI have a T1 PRI coming in that I know is TDM. I'm told I need a PRI. How can I get a PRI card to give me 23 POTS lines for modem use? And can Asterisk help with this?
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08:36.28ectospasmlgbr: Asterisk wouldn't be involved in that, afaik
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09:02.06Dovidis there any way to have asterisk listen on two IP's or is it one or all ?
09:03.35puzzledDovid: iirc it's one or all
09:03.38puzzledat least in 1.4
09:04.15Dovidpuzzled: I am wondering in 1.8.x
09:05.17puzzledDovid: no idea then :)
09:06.40Dovidlooks like all or nothing ;(
09:06.41Dovid:(
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09:08.44puzzledtoo bad, it would be a nice feature
09:08.55puzzledand iirc it has been requested for eons
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10:15.33nicola_pavhello. I have asterisk installed as a VM on proxmox. while the hosting node has fail2ban, the VM does not have iptables and so it does not block hits.
10:16.01nicola_pavthis VM receives lots of registration hits. how can i block those hits on the VM. any hints please?
10:17.28Guggenicola_pav: just like if it was a normal server
10:17.43Guggeenable a firewall on the VM, or place a firewall infront of it
10:18.03nicola_pavGugge: how do i enable firewall on the VM?
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10:37.17puzzlednicola_pav: did you try asking proxmox?
10:38.02nicola_pavpuzzled: i already submitted a question there. still waiting for answer. i thought someone here in asterisk community faced same issue
10:38.18Guggenicola_pav: just like you do on a normal server
10:38.38Guggei usually just enable pf on my servers
10:38.53nicola_pavGugge: but VMs they dont have iptables and fail2ban
10:39.14puzzlednicola_pav: google is your friend: 2nd hit when googling for proxmox & iptables: http://forum.proxmox.com/threads/833-Firewall-via-IPTABLES
10:39.24nicola_pavI receive lots of registration hits with wrong password and in really need to block them
10:39.34Guggenicola_pav: my vmware guests does.
10:39.53nicola_pavproxmox guests do not :(
10:39.57Guggenicola_pav: i have no idea how proxmox works, and i dont really care either :)
10:40.08puzzledwonders why people do not google first
10:40.15nicola_pavpuzzled: i already came across the link. did not find how to enable on the guest though
10:40.27nicola_pavpuzzled: i did google first :)
10:40.42nicola_pavGugge: thanks anyway
10:41.49Guggebut if you want to solve it in asterisk you could setup some acl on your accounts
10:42.21puzzledcan't imagine they don't offer iptables. if that really were the case I would avoid them like the plague
10:43.25nicola_pavpuzzled: u better do
10:44.27puzzledI don't use VoIP on virtual machines anyway
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13:00.54peterthesingis there someone still using asterisk 1.4?
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13:28.46e911_indianaso ive been googling, and i cant seem to find any good guides on how to convert a 7960 from SIP to SCCP for use with asterisk/freepbx, would anyone have a suggestion?
13:29.08sxperte911_indiana: from sip to sccp ?
13:29.12e911_indianayes
13:29.23sxpertdetect either mistake or masochism
13:29.34e911_indianano.. i have it right..
13:29.39sxpertok
13:29.43e911_indianathe 7914 doesnt work with asterisk when the phone is flashed as sip
13:29.55e911_indianaso id like to convert it over to sccp so i can use the 7914
13:30.05sxpertpossibly just like for sip... feed it the firmware via bootp / tftp or something
13:31.04e911_indianai downloaded the latest image from cisco cmterm-7940-7960-sccp.8-1-2SR1.cop.sgn and cisco cmterm-7940-7960-sccp.8-1-2SR1.cop
13:31.21e911_indiana<PROTECTED>
13:31.31sxpert.sgn is probably some signature file to check it's all ok
13:33.59e911_indianahmm.. alrighty.. last time i tried this i botched something.. buggered the darn thing all to bits
13:36.40e911_indianabugger.. guess the update script hasnt been running right.. 238 updates..
13:36.43e911_indiana*grumbles*
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13:45.41e911_indianahmm its still loading the sip stuff
13:46.51e911_indianaops.. another reboot..
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13:56.01e911_indianaok so that didnt work
13:56.34e911_indianait still only wants to load the sip
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13:58.07e911_indianathis bugger phone, i swear..
13:58.12e911_indianagives me more headaches than what its worth..
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14:00.55leifmadsenm4xx: if you're talking about typos / mistakes in the book, please submit them as errata through O'Reilly
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14:10.58e911_indianawell bollocks! it refuses to load sccp *grumbles*
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18:45.40_pllHello. I am trying to detect/read/identify the three tones at the beginning of ringing. It can be heard in this wav file http://www.mediafire.com/file/p3mbv96g561c5cj/tones.wav Is there a way?
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20:32.15iscariohi! does anybody is familiar with the use of asterisk with OpenBSD ? i would have several questions... thx
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20:34.04WIMPy~ask
20:34.04infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
20:36.15iscariook sorry. So : i would like to set Asterisk with mysql support under OpenBSD. I checked, there is no packages with mysql support. So what is the better way ? recompile from source, or use ODBC ? (is there is a mysql driver for mysql easily compilable ?)
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21:34.46Yudaisrael1984hi guys long time havent been here but today i started working on spandsp and im getting stuck after i install it and move the app_fax.so to the modules folder and load it in i dont see it on core show applications and i am using 1.4 for compatibility issues with other things i use. can someone please guide me on what i need to look up that im doing wrong?
21:37.36Yudaisrael1984hello anyone here?
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21:39.36Yudaisrael1984tzafrir can u go to the other channel for a minute?
21:39.37WiretapWorkYudaisrael1984, wait longer and fix your grammar
21:40.07Yudaisrael1984i apologize for my grammer but if what i wrote is understood then i would love some help
21:40.20Yudaisrael1984if it isnt understood i can rephrase
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21:43.42Yudaisrael1984basicly i need to know what im doing wrong, i am trying to add fax for asterisk using spandsp. i installed the files from the site that had all the source files and i added the app_fax to the asterisk modules folder. when i type show applications the app_fax does NOT show up and for some reason it seems that asterisk doesnt want to load the application i need help in debugging it to see what i did wrong
21:44.15tzafrirWhat version of asterisk?
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22:01.22_pllHello. I am trying to detect/read/identify the three tones at the beginning of ringing. It can be heard in this wav file http://www.mediafire.com/file/p3mbv96g561c5cj/tones.wav Is there a way? I am investigating audiohooks to create a module and set a variable if these tones are detected, am I in the right track?
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22:36.26ghostmediaprohave multiple gv account setup but want to force 3 to use ext 701-703 and not ring other phones, i'm lost as to what do with inbound routes
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22:52.00Linux4EricGood evening all
22:53.12ghostmediaprohelp with inbound routes
22:53.51WiretapWorkghostmediapro, asterisk doesn't have inbound routes
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22:54.43WiretapWork(you have no idea how much of a good thing it is that he left)
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22:56.32blitzrageKatty, ping?
22:56.34Linux4EricI am hoping for a little direction/advice.  I would like to figure out a way to have asterisk act as a gateway providing trunks to PBX's with dynamic IP's
22:57.09WiretapWorkLinux4Eric, that's extremely easy in my mind
22:57.14Linux4EricIt's a bit cumbersome to do by hand, though
22:57.23WiretapWorkreally?
22:57.58WiretapWorkadd a new peer for each incoming trunk, add a peer for each dynamic PBX, have the dynamic PBX register to asterisk, route calls from relevant trunks to dynamic PBX
22:58.23blitzrageright, the other end needs to register to your system
22:58.33blitzrageit's the only way for asterisk to know where the other end is
22:58.34Linux4EricI have trouble getting the machine to register though
22:58.45blitzragethen you should resolve that :_)\
22:58.56WiretapWorkLinux4Eric, so fix your register strings?
23:00.55Linux4Ericthese trunks aren't used to connect other pbxes to each other, but rather provide origination and termination from a single provider
23:01.21Linux4Ericso do I need to provide DID in my register string?
23:01.30WiretapWorkLinux4Eric, we're aware of that
23:01.34WiretapWorkno
23:01.55WiretapWorkyou register <USER>:<SECRET>@<HOST>:<PORT>
23:02.55Linux4Ericok, I think that's what I have.  I probably have peer info incorrect
23:03.06WiretapWorkwhat does debug say/
23:03.08WiretapWork?
23:04.10Linux4EricLet me go get back to my machine and if I have some messages in console i will provide a pastebin.
23:04.42WiretapWorkyou might wanna turn debug _on_ first
23:04.58Linux4Ericok will do that
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23:13.11WiretapWorkLinux4Eric, if you're using FreePBX things are a bit different
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23:14.06Linux4EricSome of my PBX's do.  Others are plain asterisk. That's where I might get confused.
23:14.34WiretapWorkLinux4Eric, well the register strings are the same either way, but what does the 'core' run/
23:14.35WiretapWork?
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23:15.44Linux4EricI have to split my sip.conf up in freepbx i believe.  The core is just plain asterisk, the pbx's are mostly freepbx
23:16.06Linux4Ericall are 1.6 under the hood
23:16.09WiretapWorkLinux4Eric, with freepbx you don't need to touch ANY .confs
23:16.13WiretapWorknot for your setup
23:16.17WiretapWorkwhich is seriously simple
23:17.09Linux4EricOk, but the gateway machine is plain asterisk unless Freepbx would make that a lot more simple too
23:18.20WiretapWorkdepends, with freepbx it would be very simple to do it simply
23:19.26WiretapWorkbut if you wanted to do any complex stuff, it migth come down to hand-editing
23:20.09Linux4EricI sometimes have to tap into the extensions_custom.conf to do some custom stuff anyway so whatever works best I am going to do
23:21.35Linux4EricI always get confused what to put where in the freepbx sip trunk screen.  It might be easier for me to just put the block of code in sip_custom instead
23:23.35WiretapWorkwhat do you mean' what to put where'?
23:24.02Linux4Ericit has incoming and outgoing section.  (Peer and User)
23:24.24WiretapWorkignore the USER section
23:24.36WiretapWorkwell... that's what I've foudn anyway
23:24.49WiretapWorkyou're right baout it not being terribly well documented, but I've never needed anything in there
23:25.54Linux4EricOk well that might be where I am running into trouble.  I will try just it with just peer section
23:26.42WiretapWorknot sure about the 'type' you want
23:32.06WiretapWorkas some trunk providers prefer peer and some friend
23:33.12GnewtI had an old Debian box with asterisk 1.6 compiled from source which took inbound DTMF just fine. I transferred the config files to a new box running the Debian precompiled Asterisk 1.6, same config and it's working and all, but it won't recognize inbound DTMF and as such I can't make my DISA work. Any ideas?
23:34.53WiretapWorkGnewt, is your incoming trunk not using RFC2833 DTMF?
23:35.34GnewtI think it should be. I'm using IPKall which gets routed to CallWithUs. My box registers with CallWithUs for inbound.
23:35.44GnewtBut it should be fine seeing as the exact same configs worked on the old box?
23:36.14WiretapWorknot sure, but I know people have issues with inband dtmf if they don't have a timing source IIRC
23:36.50GnewtHow do I make sure I have a timing source?
23:36.52Gnewt:/
23:37.03WiretapWorkdo you have a DAHDI card?
23:37.05GnewtI noticed dahdi_dummy wasn't loaded
23:37.06Gnewtno
23:37.10Gnewtbut I modprobed dahdi_dummy
23:37.45Gnewtalthough now I see it isn't listed in lsmod
23:37.46WiretapWorkI would recommend configuring your trunk for AVT/RFC2833 DTMF anyway, I know little more beyond the basics I'm afraid
23:37.59WiretapWorkit would be autoloaded if it was in use IIRC
23:38.34*** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net)
23:39.24GnewtMaybe I should set IPKall to go straight to my Asterisk box rather than through a trunk
23:40.15Linux4Erici am no expert but precompiled asterisk doesn't include the DAHDI software it's a separate install
23:41.57Linux4EricI think a timing source is only needed for IAX2 and meetme, for what it's worth
23:49.06Gnewtdahdi is a dependency though
23:53.27Linux4EricGnewt are you running this on a local box, not something like a cloud or VPS provider
23:54.13*** join/#asterisk tzafrir (~tzafrir@212.179.75.202)

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