IRC log for #asterisk on 20110208

00:04.23dr00dthanx
00:28.49*** join/#asterisk sourcode (~code@ppp-115-87-239-138.revip4.asianet.co.th)
00:29.52*** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk)
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00:56.59*** part/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com)
01:13.42devmodIs there a way to play sounds to a channel through AMI or cli ?
01:19.09fauxallianceAPI... Background(foo)
01:27.56*** join/#asterisk Kumbang (~kumbang@180.245.137.5)
01:30.18*** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey)
01:32.48*** part/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com)
01:34.59kukuAny reason why an inbound call would go to fast busy after two rings when connected to a TDM400p, even when asterisk is not running ?
01:37.17fauxalliancepay your bill?
01:43.08kukuWhen I disconnect the line from the card, it keeps on ringing.... Other lines are fine.
01:43.18kukuatt said everything is ok on their side ... :)
01:43.22drmessanoHow old is the card?
01:49.14drmessanoAre you carbon dating it?
01:50.00drmessanoMooooonkey?  Mooooooonkey? Mooooooonkey?  BANANA?!
01:50.03*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
01:50.30drmessanoMooooonkey?  Mooooooonkey? Mooooooonkey?  2 BANANAS?!
01:54.47*** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net)
01:54.59drmessanoAnyone here build from 1.8 branch SVN?
01:56.08drmessanoI didn't spend any time debugging, but this morning I tried an SVN update and kept building Zombies.  Had to go back something like 12 days to get a working build.
01:56.24drmessano(Zombie being a technical term, of course)
02:00.11ChannelZtry a bigger gun
02:04.06*** join/#asterisk shapr (~shapr@nat/digium/x-atktgqdglqcywfyq)
02:29.08WIMPyOh, the new format identifiers are there.
02:38.15*** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com)
02:38.31*** join/#asterisk Kumbang (~kumbang@180.245.137.5)
02:55.50luckman212my 1.8.2.3 zombies out every now and then.  1.8 is a bit zombie'ish (also a technical term)
02:57.29shaprluckman212: Any specifics?
02:59.08WIMPyWhat kind of zombies do you get? Or from wha actions?
02:59.24luckman212well I need to recompile with DEBUG_THREADS so I can get proper traces first
02:59.33luckman212but I believe most of my problems are related to MOH at the moment
03:00.49luckman212when my * zombies, I can get to the CLI and run commands but SIP dies, phones can no longer register or make any calls
03:01.11luckman212but the core stays up and just keeps humming along doing nothing in particular
03:02.39WIMPyI usually find that the CLI locks up as well as soon as you press tab.
03:05.20luckman212YES that happens to me often as well
03:05.30luckman212at that point you can't even CTRL+C
03:05.59WIMPyBut you can Ctrl+\
03:06.26luckman212what's Ctrl+\ ? don't know that one
03:07.08WIMPySIGQUIT
03:07.22luckman212hm.  nice to know.
03:07.56luckman212I can't even type Ctrl+\ on my keyboard because I've mapped that key combo to launch my screen grab program
03:08.17luckman212guess I'll make a macro for my terminal emu
03:08.39WIMPyIt might already have a menu entry for that.
03:20.16dr00di have another question re skypeforasterisk for anyone thats interested - ive installed skypeforasterisk and it works - how do i call my asterisk server from skype ? what do i call ? a skype user name that i have configured in chan_skype.conf ?
03:20.33dr00dblinks
03:27.36luckman212sorry I have no clue about that.
03:29.12dr00dthats ok im getting there ill post the answer in a min
03:29.39dr00di can call skype users from asterisk ! it so exciting
03:42.59luckman212that is pretty cool actually
03:43.28luckman212whats the interface for that?  I mean where do you enter someone's skype username on your SIP phone?  or do you have to make custom extensions in your dialplan?
03:45.42shaprI see some discussions from 2008 or so that suggest turning off hyperthreading for an asterisk PBX, is that still recommended? If yes, why?
03:45.47WIMPySwitch your phone to alpa mode if you can, yous its web interface or put short codes in to your dialplan.
03:46.51luckman212yes my phone (polycom) can dial alpha (for sip uri's)  so do you dial like "megan2524534@skype.com" or something?
03:47.00dr00dto call a skype user from asterisk you have to install skypeforasterisk, then create an entry in the chan_skype.conf file, then add an incoming route / did and you put the user name you want to call in the dial field and set the destination
03:48.07dr00dsoory wrong - not a did you have to create an extension
03:48.42WIMPyA prefix might be an idea.
03:49.09shaprdr00d: Or you could use Originate from the CLI to make a call and connect it to an extension
03:49.31dr00dok - i havevnt done it that way yet
03:50.16dr00dyou create an extension with an associtated skype name and then calling that extension calls the skype user - for outgoing
03:50.34dr00dfor incoming i havent figured it out yet
03:51.05luckman212what codecs does the skype channel driver support?
03:51.22luckman212can you do g722?
03:51.58dr00dit only supports g722 and g729
03:52.30dr00dand the installation guide for skypeforasterisk says to use g729
03:52.45dr00dwhich you need a license for - but one comes with sfa
03:53.46*** join/#asterisk ryanlin (~ryanlin@Your.Dad.Blamed.Me)
03:53.50luckman212hmm so is SFA a free product?
03:53.55ryanlini set up a sip trunk with a provider
03:54.09ryanlinand the provider routed a DID to me
03:54.22dr00dno it costs au$66 per concurrent call
03:54.25ryanlinwhenever i dial that number, i get "all circuit are busy"
03:54.25dr00dits pretty good
03:54.31ryanlincan anyone help?
03:54.38ryanlini can post the config
03:56.04luckman212ryanlin: check your dial patterns
03:56.30luckman212does your provider require you to dial numbers in a specific format (e.g. 1NXXNXXXXXX ?
03:56.45WIMPyTurn up verbose and debug and see what happens.
03:57.07ryanlinhmm.no
03:57.26ryanlini will not be performing dialing functions from the asterisk
03:57.38ryanlinthe asterisk will only receive calls..so therefore, dial patterns is not needed
03:57.41ryanlinright?
03:57.54ryanlinall i did was route the call to an auto attendednt
03:57.56luckman212oh I see
03:58.00ryanlinone second
03:58.02ryanlinlet me post my config
03:58.07luckman212what does your asterisk -rvvvvv say?
03:58.28ryanlinit didn't say anything..i am really puzzled
03:58.38ryanlinit shows as a regular sip output
03:58.39luckman212so the call never even hits your pbx?
03:58.44ryanlinit hits the pbx
03:59.09ryanlinbut i did not see any errosr in the output by performing sip set debug on with the asterisk -vvvvvvvv
03:59.12luckman212so if you run asterisk -rvvvvv and then call your DID, what do you see on the screen?
03:59.24ryanlini will paste it right now
03:59.28ryanlinhold on
04:00.43dr00domg it works ! now i can call from skype to asterisk and vice versa omg omg
04:00.58ryanlinhttp://pastebin.com/hTRDfpn1
04:01.27ryanlinand hold on..let me paste the config
04:01.52WIMPyryanlin: There was no call.
04:02.14WIMPyDo you have to register? Did you do so?
04:02.16luckman212ryanlin: turn off SIP debug for now
04:02.29luckman212lets just see how your dialplan execution is looking
04:03.27ryanlinah..wtf!
04:03.31ryanlinno call is coming in?
04:03.36ryanlinno wonder i can't see anything
04:03.40ryanlin]sure..hold on
04:03.44luckman212those are just SIP OPTIONS packets
04:03.52ryanlinoops..i am so sorry
04:03.56luckman212kind of like pings
04:04.02ryanlinah..i see..
04:04.03ryanlinokay
04:04.32ryanlinhttp://pastebin.com/vs1U7Ahp
04:05.20luckman212you have no register line in there?
04:05.26WIMPyAs long as you're not getting a call, the configuration doesn't matter.
04:05.31WIMPyDo you have to register? Did you do so?
04:06.06luckman212ryanlin: who is the trunk provider?
04:06.15ryanlinpac-west
04:06.29ryanlinyea..no registration requried
04:06.32luckman212usually they have a config/interconnect page that details how to properly register
04:06.32ryanlinrequired*
04:06.49luckman212so how do they authenticate, via static ip?
04:07.11ryanlinyep
04:07.21WIMPyAuthentication is a different matter.
04:10.29luckman212I just tried to dial 8052213335@208.57.239.33 and I get this ''Got SIP response 603 "Declined" back from 208.57.239.33:5060-- SIP/208.57.239.33-00000047 is busy"
04:11.21luckman212whats in the "test_mainmenu" context?
04:11.59luckman212try changing qualify=NO
04:13.26ryanlinok
04:13.37ryanlinone second
04:13.42luckman212one other thing I noticed, I get this error in my * logs when I called you "No SRTP module loaded, can't setup SRTP session"
04:13.51luckman212are you only allowing secureRTP calls?
04:16.29ryanlinno
04:16.32ryanlini don't think so
04:16.49ryanlinthe default asterisk config allows non-secure rtp calls as well right?
04:17.45luckman212yes, SRTP requires some effort to set up so if you didn't do it, then you don't have it
04:19.52ryanlinok
04:20.10ryanlinone sec let me restart the config
04:20.15ryanlini just made the changes for qualify
04:21.03ryanlinnope..same shit
04:21.09ryanlinhold on..let me get on the phone with pac-west
04:22.22luckman212ok, be sure to let us know what it was if you figure it out
04:22.36ryanlinPac-West RTP Destination IP 66.42.123.4 or 66.42.123.5
04:22.48ryanlincan i specify this in sip.cofn??
04:23.33luckman212asterisk will get the RTP source & destination IPs & port number via the SIP INVITE
04:23.41luckman212you dont have to specify those
04:24.16ryanlinok thx
04:24.16luckman212as long as SIP traffic can freely flow between you & your trunk (can it?) is there any firewall (iptables, pf, etc) in place here?
04:24.34luckman212the default for SIP is UDP port 5060
04:25.12luckman212but pacwest may use something different, e.g. 5080
04:28.36ryanlinah
04:28.38ryanlinthe provider's fault
04:29.34luckman212yes?
04:48.00p3nguin
04:54.44*** join/#asterisk Alborracho (~chatzilla@200.26.174.218)
04:55.33Alborrachohi, im having a lot of problems witrh noise in a T1 telco line, i tried to adjust rxgain, txgain with no result, i changed my card from digium to sangoma with no result
04:55.37Alborrachopls any ideas?
04:57.10WIMPyDon't use gain on digital interfaces.
04:57.58WIMPyDo you also experience other issues like HDLC aborts?
04:59.48leifmadsenthe noise is not likely on the digital line itself
05:00.00WIMPyHave you tried putting the card in to another slot? Or even another board?
05:00.08leifmadsengoes to sleep.... again.
05:00.29WIMPyAgain? Sleep chatting?
05:02.00WIMPyWell, I occasionally take a peek when I get up for a wee, but I wouldn;t have thought others are as crazy as me :-)
05:02.36*** join/#asterisk timahvo1 (~rogue@41.72.215.94)
05:06.16Alborrachowell i have both cards atm
05:06.21Alborrachoin different slots ofc
05:06.39Alborrachoonly one card i enabled
05:08.14Alborrachoi have two servers one with digium and other with digium and sangoma
05:08.19Alborrachoboth have noise
05:08.39Alborrachodo you think it could be the T1 cable?
05:09.19WIMPyCan you describe the noise?
05:09.48Alborrachowhen there is silence, well it behaves "normal" nothing happens
05:09.48WIMPyIs it contant or does it vary?
05:09.50Alborrachobut when
05:10.02Alborrachoi playback an mp3, wav or gsm
05:10.11Alborrachois like heavy rain
05:10.16Alborrachoand a lot of saturation
05:10.41WIMPyAre they just too loud?
05:10.41Alborrachoi tried to make a SIP to telcos network phone call
05:10.50Alborrachoyes, really loud
05:11.05Alborrachothe voice is like talking from a walkie talkie
05:11.13Alborrachokind of robotic
05:11.41Alborrachoit is constant
05:11.51Alborrachoduring the playback of any sound
05:12.03WIMPyI usually save my moh files at -12db so they don't overdrive.
05:12.29WIMPy(is that the correct term for clipping signals?)
05:12.56Alborrachoi tried that but when i play them, the important sound is low and the noise keeps the same
05:13.22ketemaanyone know how to force a Cisco 7970 to register over TCP ?  been following https://issues.asterisk.org/view.php?id=13996 and other articles but the phone just wont even try to connect over anything but UDP .... using sip70-8-5-4S  thanks in advance
05:13.22Alborrachoi recorded the tx channel with dahd_monitor
05:13.28Alborrachoand it looks really nice
05:18.56WIMPyOk, so the suspects would either be jour line or your line coding configuration.
05:20.04Alborrachotomorrow i will try to connect both * boxes with a crosed T1
05:20.14Alborrachoand make a call between then and record it
05:21.12WIMPyIs it a pri or some inband signalling?
05:21.26WIMPyrbs
05:22.50Alborrachopri with ss7
05:23.02Alborrachopri cpe
05:23.49WIMPyOk, so wrong lie coding should give you more trouble. Do you have the right codec?
05:24.05WIMPys/lie/line/
05:24.31WIMPyI've never tried what it sounds like, when you mix them up.
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06:36.13*** join/#asterisk Bloudermilk (~Bloudermi@cpe-76-90-0-11.socal.res.rr.com)
06:36.42BloudermilkHey all. I'm wondering if anyone has a page where I can find a nice sip.conf setup to allow one of my asterisk boxes to Dial() another on a the local network
06:38.03shaprBloudermilk: make an iax2 trunk between 'em?
06:38.09shaprthen you can dial from one to another
06:38.16shapror do you want them to be sip peers?
06:39.45BloudermilkHmm. I've never messed around with IAX2 before. I'm down for suggestions as to what the easiest/most efficient way to do it would be, but my only requirement is that I be able to Dial() potentially multiple boxes on my local network for a given channel
06:39.53BloudermilkSo it would Dial() one, then another, etc until hangup
06:40.12BloudermilkI figured a sip peer would be the easiest, but I hadn't considered IAX2
06:40.17BloudermilkNor do I know what it is :)
06:40.34shaprinter-asterisk xchange protocol, version 2
06:40.43BloudermilkOhh interesting
06:40.44*** join/#asterisk darkskiez_ (~dz@62-50-193-152.client.stsn.net)
06:42.35BloudermilkWell on that note, do you have any information on setting up IAX to allow me to do what I explained above?
06:45.22shaprBloudermilk: http://www.asteriskdocs.org/html/ch04s12.html ?
06:45.39Bloudermilkshapr: Thanks very much!
06:45.48WIMPyThe setup is very similar.
06:47.47BloudermilkWIMPy: It seems so
06:47.57*** join/#asterisk SkoZombie (~quassel@60-240-15-173.tpgi.com.au)
06:48.07*** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114)
06:48.54SkoZombieI'm recently having a problem that calls are passing through the local handset extension number rather than the username for our SIP provider
06:49.29SkoZombiein the "From" field
06:49.56WIMPyDid you use Dial() with @ instead of / ?
06:52.36SkoZombieWIMPy: using DIAL(SIP/123456/${EXTEN}) in our dialplan where 123456 is our account number with our provider
06:53.09*** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110)
06:53.29WIMPyOk, that's no possible cause.
06:54.16SkoZombieI'm getting 403 forbidden from my provider for what it's worth
06:54.38SkoZombieand I've just compiled and installed the very latest sourcecode to ensure it's not some ubuntu-package type bug
06:56.15*** join/#asterisk manji (~manjiki@ppp-94-65-254-67.home.otenet.gr)
06:57.49shaprSkoZombie: checked that the username and secret are correct? verified with a softphone?
06:57.59SkoZombieshapr: works fine with a soft phone
06:58.13SkoZombiethats why my provider is getting pissed at me, they test it, and it works fine
06:59.19SkoZombiethe INVITE seems to come directly from the phone, not asterisk, but it hasn't REGISTERed, so the provider is responding 403
06:59.58WIMPyDo you have fromuser set?
07:00.17WIMPyRegistering is usually not related.
07:00.26shapror canreinvite=no if you're using NAT?
07:01.14shaprer, that would in your sip.conf
07:02.17shaprSkoZombie: In the asterisk CLI try "sip set debug on" and then "originate SIP/yourcellphonenumber application Playback demo-congrats"
07:02.53shapryou may want to do "core set debug 10" and "core set verbose 10" before that to get a deafening amount of information
07:04.38shaprSkoZombie: works?
07:04.48SkoZombiejust testing now
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07:06.33kaldemarSkoZombie: set fromuser for the peer in sip.conf
07:06.42SkoZombiekaldemar: ok I'll try that
07:06.48SkoZombieshapr: no joy
07:06.55shaprwhat errors?
07:07.00*** join/#asterisk Russ (foobar@ip70-176-251-1.ph.ph.cox.net)
07:07.30SkoZombieshapr: hitting enter after  "originate SIP/yourcellphonenumber application Playback demo-congrats" just gives me a new line on the console
07:07.56shaprdid you turn on sip debug and debug & verbose?
07:07.56kaldemarSkoZombie: defaultuser and remotesecret may also be necessary, if your asterisk version has those options.
07:08.14SkoZombieshapr: yeah, started getting all sorts of stuff that tcpdump was giving me
07:09.34*** join/#asterisk Kumbang (~kumbang@180.245.137.5)
07:09.58SkoZombiekaldemar: where does defaultuser/remotesecret belong?
07:10.42kaldemarSkoZombie: in the peer definition.
07:11.13SkoZombieshapr: yes
07:11.28SkoZombiekaldemar: still no joy
07:11.42SkoZombiethe error I'm seeing on the console is:  WARNING[24868]: chan_sip.c:19082 handle_response_invite: Received response: "Forbidden" from '"Andrew Whalan" <sip:101@59.167.136.212>;tag=as0f0d2c0d'
07:12.08SkoZombie59.167.136.212 being my IP
07:12.25SkoZombiei have canreinvite=no in the peer definition
07:13.51SkoZombieit's hitting the dialplan rule of: exten => _0.,1,NoOp(Outgoing call on ${EXTEN})
07:13.51SkoZombieexten => _0.,n,DIAL(SIP/123456/${EXTEN})
07:14.54kaldemarand 123456 is the peer in sip.conf?
07:15.16SkoZombieyeah, defining my SIP provider
07:15.50SkoZombieI just don't get why asterisk isn't placing the call as 123456@provider and passing it on to the internal extension 101
07:16.39ChannelZseems more like the callING peer is being rejected
07:16.43ChannelZsip:101
07:17.01SkoZombieI can call other extensions fine
07:17.28SkoZombieI don't know why the phone is the caller not the asterisk server
07:17.31ChannelZwithin your own system
07:17.36*** join/#asterisk Iamnacho (~Iamnacho@ip174-70-138-161.ks.ks.cox.net)
07:19.03ChannelZIf your ITSP doesn't only matching you by IP, chances are it hates your From
07:19.30SkoZombieChannelZ: that sounds like the issue
07:19.41ChannelZdo a Set(CALLERID(num)=1111) before you Dial(), 1111 being your DID they recognize you as
07:19.41SkoZombiei'd just like to pass valid auth info to my provider when I place a call
07:20.32ChannelZsee if that does anything.  I have to do that for my particular provider otherwise it sees whatever extension my softphone or whatnot is programmed with and barfs
07:21.13shaprSkoZombie: If you're getting 403 instead of 401, I'd check the user or secret set for your SIP provider.
07:21.22SkoZombieChannelZ: that worked!
07:21.24SkoZombieffs
07:21.33SkoZombieone month they took frigging around with it
07:21.40SkoZombieand you fixed it in a few minutes
07:22.48shaprHey, we're just awesome here.
07:23.00ChannelZyou might be able to get around it also with 'fromuser' in your sip.conf for that peer, but I don't remember
07:23.18SkoZombieyeah, and if any of you (shapr, ChannelZ, kaldemar) are in newcastle, nsw, au ever, I'll buy you all the beer you can drink
07:23.34SkoZombiethis has been such a freaking headache
07:23.56ChannelZwell glad you got it sorted
07:23.57shapr<PROTECTED>
07:24.22SkoZombiewhere are you based?
07:24.49shaprI'm based in Huntsville Alabama at the moment, though I've previously been based in Stockholm.
07:25.20Dovidshapr: Isn't it in middle of the night by u ?
07:25.34shaprYup, it's 01:25 here
07:25.40shaprDovid: Where are you based?
07:25.47DovidIsrael
07:26.00Dovidfrom Jersey so I know all of the TZ's in the US. I go back and forth
07:26.51WIMPy's got two pieces of israeli hardware on my desk that is not too talkative.
07:27.40Dovidlol
07:28.17DovidWIMPy: Xorcom ?
07:28.51WIMPyradvision
07:28.55*** join/#asterisk DJClean (~djclean@unaffiliated/djclean)
07:29.08shaprWIMPy: Galil?
07:29.32shaprWhat's the default behaviour for asterisk.. do config files or runtime settings win?
07:29.55WIMPyWell, I might have use for that as well, but that's another story.
07:30.26shaprI'm specifically interested in ecm=yes in res_fax.conf and then Set(FAXOPT(ecm)=no)
07:31.02kaldemarshapr: runtime settings.
07:31.13shaprthanks
07:31.29kaldemarconfig files are just read upon module loads.
07:33.22SkoZombieERROR[25199]: chan_sip.c:27972 setup_srtp: No SRTP module loaded, can't setup SRTP session.
07:33.22SkoZombie<PROTECTED>
07:33.24SkoZombiewhat does that mean?
07:34.13ChannelZno secure (encrypted) RTP
07:34.21SkoZombieah ok
07:34.35SkoZombieI might uninstall this manually commpiled version and go back to the packaged version
07:35.36*** join/#asterisk hehol (~hehol@2001:1438:1009:200:e17f:bb65:bc44:1ee1)
07:35.37drmessanoYou dont hear that every day
07:36.36shaprEh, I wish the packaged versions were often updated, would make life easier.
07:36.57shaprSkoZombie: What version of Asterisk? oh, and what distro?
07:37.11shaprPersonally, I wish Ubuntu were the standard Asterisk distro instead of RedHat derivatives.
07:39.35SkoZombieshapr: 1.6.2.7 now
07:39.44SkoZombiewas using the latest source from the website
07:39.51SkoZombie1.8.2.3
07:40.01SkoZombierunning ubuntu
07:42.16shaprI wonder if I could persuade someone here to make ubuntu 10.10 packages for the latest release of Asterisk...
07:43.29SkoZombieshapr: there's a few PPA's
07:43.45SkoZombieproblem is that if it's not in universe/multiverse, most people won't find it
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07:48.24schmidtsgood morning
07:52.06shaprgrussgott schmidts
07:53.27schmidts;)
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07:53.44BloudermilkHey guys. I'm looking at iax.conf right now (using the example on http://www.asteriskdocs.org/html/ch04s12.html) and I can't figure out how to set it up for only one direction dialing
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07:53.59Bloudermilkie. box A needs to Dial() box B
07:54.14BloudermilkShould box A register() and box B define a type=peer?
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07:56.13shaprBloudermilk: if you only need one direction, setting up a SIP peer is probably simpler
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07:59.28Bloudermilkshapr: Thanks :)
07:59.48BloudermilkIn that case though, which end does the register go on, and which end does the definition of the peer go on?
08:00.45nix8n82I have a problem with a TDM400p card not getting the caller id all the time. When the caller id does not come in it chan_zap show Ring but no Ring/Answer and callerid.c says it fails checksum. There are 4 lines that roll over but it happens most on line 3
08:01.09nix8n82What could cause this?
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08:02.40schmidtsbloudermilk you will have the same problem with sip then you have with iax ;)
08:03.19schmidtsbloudermilk you have to define a user an box a and a peer on box b and on box a you must set the register user to box b, or you can just use friend on both sides ;)
08:04.33BloudermilkThanks schmidts and shapr :)
08:04.45BloudermilkI figured they were more or less the same
08:04.48BloudermilkJust different protocol
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08:05.09BloudermilkNight guy
08:05.10Bloudermilks
08:05.38drmessanoI would have loved to help him, but I am lactose intolerant
08:05.41drmessanoNo milk for me
08:06.11schmidtsdrmessano Mr black coffee ?
08:07.19drmessanoEwww
08:07.24drmessanoNon-dairy creamer :)
08:08.15WIMPytmi
08:10.52shaprnix8n82: Has the tdm400p gotten callerid correctly before?
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08:27.58drmessanoAny thoughts on getting Alison to read back a string as a time?  Ex.  0740 as Seven-Fourty
08:29.23drmessanoI could split it, which would handle 1 thru 12 just fine.. Not sure about the second set of digits.. 40 being fourty and 05 being oh-five vs five
08:29.24shaprdrmessano: If you know it's a time, hack it up a bit and play "7" and then "40" ?
08:29.34drmessano^^
08:29.51shaprdrmessano: You could turn read/show the number to get rid of the leading zero
08:30.15drmessanoThe leading zero isn't a problem.. it's the last two digits, I guess.
08:30.29shaprWhat's wrong with forty?
08:30.34drmessano40 is fine
08:30.35shapror, what would you rather have?
08:30.45drmessano05 is oh-five not five
08:30.55drmessanoWell, it NEEDS to be oh-five
08:31.58drmessanoI guess I need at least an extra line to break out 00-09 to SayDigits and 10-59 to SayNumber
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08:37.45drmessanoTurning out to be like 6 lines of dialplan to read "your call has been scheduled for seven-fourty" lol
08:40.26shaprPython could do it handily
08:40.45drmessanoI am allergic to snakes
08:40.55shaprHaskell?
08:41.40drmessanoDoes Asterisk accept Commodore 64 BASIC?
08:41.53drmessanoI could whip this out in like 5 mins in BASIC
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08:42.13drmessanoor at least put multicolored dots on the screen
08:42.32drmessanoActually, I THINK I found a way to do it
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08:50.26nix8n82shapr, yes it does get the callerid alot but if fails once in a while..like 300  times out of the past 1500 calls  since the begging of the year
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09:44.14Diffen2Good morning, I have asked this before but whats the syntax for getting the extensions out from the asterisk cli to a text file.
09:44.28shaprgodmorgon Diffen2
09:45.06Diffen2shapr :)
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09:45.16shaprDiffen2: "dialplan show" funger för mig
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09:45.22shaprfungera*
09:45.54plundra(And to put it in a file, 'asterisk -rx "dialplan show" > yourfile.txt)
09:46.16shaprjo, faktist
09:46.42shaprarbetar för Digium och gör den svensk tech support
09:46.56shaprSadly, nobody ever calls for Swedish tech support :-/
09:47.15Diffen2:D
09:48.12shaprIs there lots of Asterisk use in Sweden?
09:48.28kaldemarshapr: do you do swedish support in alabama?
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09:50.47WIMPydrmessano: Have you tried brandy?
09:51.20shaprkaldemar: Sure, why not?
09:51.48*** part/#asterisk TobSnyder (~schneider@dslb-088-073-184-171.pools.arcor-ip.net)
09:51.55shaprkaldemar: But seriously, my knowledge of Swedish is incidental to my job here.
09:51.58kaldemarshapr: thinking about working hours.
09:52.30shaprAt the moment I do night shift, but I still don't get any Swedish callers.
09:53.17schmidtsshapr do you want to take away oej?s customers?
09:53.33kaldemarshapr: DIY people those swedes. :)
09:55.22shaprLast I lived in Sweden it was covered up in Microsoft software except for a few outposts of open source.
09:55.44Diffen2Hmm isnt it possible to get the peers from the cli ?
09:55.58shaprsip show peers
09:56.19schmidtsdiffen2 why dont you just look at the config files? or is this not your server?
09:56.51Diffen2yes its my server
09:57.48schmidtsdiffen2 just asked ;) with sip show users you can also see the passworts
10:00.10Diffen2hmm dont want that but i found what i was looking for :) thanks
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10:04.27pifhi, when using MeetMe(1234,M) (with musiconhold) the music sounds terrible (when using MusicOnHold is sounds fine), any idea?
10:05.55shaprpif: Do you have DAHDI installed?
10:06.01pifyes,
10:06.27pifand * 1.6.2.9
10:08.17shaprpif: What do you get from "dahdi show channels" in the CLI?
10:08.29shaprIs the MOH being transcoded?
10:08.58pif<PROTECTED>
10:09.02pifetc,
10:09.35pifI don't think so, it's in .al format (alaw)
10:09.37shaprDo you see the pseudo channel?
10:11.32pifpseudo            default                    default                         In Service
10:13.37shaprpif: check the system load during meetme use?
10:13.43pifok
10:14.46shaprIf it only happens during meetme, I'd guess the dahdi_dummy module isn't loaded to give timing
10:15.11shapror that perhaps transcoding and mixing at the same time is too much load for your system
10:16.28pifunlikely as I have a Digium E1 card with dahdi module loaded, and the server is not loaded
10:16.50pifI'll keep looking :)
10:17.34shaprYah, sounds unlikely in that case.. you definitely have timing available.
10:17.56shaprpif: looked at core show translation to see if you have any crazy large values?
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10:19.17pifwhat would be a crazy value? gsm/speex is the highest at 8001
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10:19.50shaprThat's still only 8ms, so...
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10:27.34schmidtsshapr 8 ms is a lot IMHO
10:28.38schmidtsok not really, i have taken a look at my server ;)
10:31.40pifwhat's your number ?
10:37.27schmidtspif if you mean me, i will show you in a moment ;)
10:37.41pifyes
10:40.44schmidtshttp://pastebin.com/VXCEVgnB thats the values from three different servers but dont ask me why i have these strange high values on the svn trunk version
10:42.32schmidtsand again asterisk 1.2 wins :D
10:43.25schmidtsoh moment, i know why i have these strange values, on the second server the card isnt loaded correctly, just dahdi dummy again ;)
10:45.40schmidtspif maybe thats your problem too?
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10:48.28pifI don't need dahdi_dummy as I have real dahdi hardware :)
10:48.44pifwith a non-dummy dahdi module
10:48.55schmidtspif thats what i mean, maybe your hardware doesnt work rigt
10:49.20pifoh it's working else 100 users would be screaming in my ear
10:50.45schmidts:D
10:51.22shaprpif: Can you describe how the MOH "sounds terrible" ?
10:52.18piflike "too slow", prolly a codec or transcode problem, no biggie, I  disabled MOH in the MeetMe for now
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10:53.29schmidtspif AFAIK there was a bug with a wrong translation path between two codecs but i am not sure if this was fixed in 1.6.2.latest
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10:55.36pifI'll check that
10:56.45schmidtspif http://lists.digium.com/pipermail/asterisk-dev/2010-September/046401.html and following mails
10:56.53pifthx
10:56.57schmidtspif but this was only for 1.8 but maybe you have the same problem
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10:59.12andylockranhey guys, anyone using VoIPtalk in the UK?
10:59.47andylockrancalling into my IVR, and DTMF works fine - trying to use 'Confirm Calls' -when dialing to my o2 mobile, and DTMF not recognised.  Any suggestions?
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11:11.13johanneslWhat could be the cause for the "Background" application to not return control to the next row in execution order?
11:11.51kaldemarjohannesl: a keypress
11:12.27johannesljohannes, Ah, but well I'm just silent and the step below doesn't get run
11:13.21kaldemarpastebin your extension and we'll have a look.
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11:16.55shaprjohannesl: Do you have an error handler extension?
11:16.58johanneslkaldemar, It's a Background followed by a NoOp. Replacing Background with Playtones does keep execution going.
11:17.18johanneslAnd the sound file is playing just fine, but execution halts
11:17.42johannesl1.4.21.2~dfsg-3+lenny1
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11:20.26kaldemarjohannesl: pastebin the extension and CLI output for a call with verbosity enabled.
11:29.03shaprMay I encourage everyone to upgrade to Asterisk 1.8? Digium drops commercial support for earlier versions in a few months.
11:29.46schmidtsshapr isnt commercial support only for the business edition available?
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11:34.47shaprschmidts: Nah, you can purchase support from Digium for the open source edition.
11:34.58shaprAlso, as far as I know BE is no longer sold.
11:35.09schmidtsshapr good to know ;)
11:45.35johanneslkaldemar, Sorry for my late reply, tight schedule here but thanks for you help. I actually found a solution for my original problem without using Background.. Doing workarounds for broken SIP providers is not the most fun task
11:46.24johanneslI wonder it must be so hard to get ring tones right :-|
12:05.14*** join/#asterisk DelphiWorld (~VoIpGuy@41.200.24.251)
12:05.16DelphiWorldhi guys
12:05.26DelphiWorldhow can i manualy dial a number through FXO, from console?
12:05.48DelphiWorldi configured my dahdi span
12:06.03DelphiWorldusing dahdi_genconf but if i dial ththrough my FXO isn"'t working
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12:10.43kaldemarDelphiWorld: "channel originate" or "console dial".
12:12.49DelphiWorldkaldemar: can you give me full command please?
12:13.05DelphiWorldkaldemar: let say i want to dial 911 and pass it to sip softphone 101
12:13.58schmidtschannel originate SIP/101 extension 911 or channel originate SIP/101 Dial g0/911
12:14.12kaldemarchannel originate DAHDI/1/12345 application Dial SIP/101
12:14.24schmidts:D
12:14.47kaldemar^ assuming your FXO channel in DAHDI configs is 1 and the number you want to dial is 12345.
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12:15.34shapror perhaps: originate SIP/yourcellphonenumber application Playback demo-congrats
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12:16.16DelphiWorldkaldemar: [Feb  8 12:16:01] NOTICE[3350]: channel.c:5316 __ast_request_and_dial: Unable to
12:16.17DelphiWorldrequest channel DAHDI/1/1544
12:16.44kaldemaris chan_dahdi.so loaded?
12:17.37DelphiWorldkaldemar: yes... http://www.dpaste.de/Nzwz/ is dahdi_channel.so
12:17.43DelphiWorldkaldemar: yes... http://www.dpaste.de/Nzwz/ is dahdi_channel.conf... grrr
12:18.33kaldemardoesn't look like a config file.
12:19.17kaldemarand the appropriate file would be chan_dahdi.conf.
12:19.24DelphiWorldlol kaldemar sory :-P
12:20.24DelphiWorldkaldemar: http://www.dpaste.de/z4zS/
12:20.58DelphiWorldohh kaldemar i didn't edit chan_dahdi.conf...
12:21.15DelphiWorldkaldemar: this dahdi_channel.conf is generated by dahdi_genconf
12:21.51kaldemari suppose "dahdi show channels" in CLI lists those?
12:22.56DelphiWorldkaldemar: nothing listed in dahdi show channels
12:23.48kaldemaris dahdi_channels.conf included in chan_dahdi.conf, i.e. #include dahdi_channels.conf ?
12:25.15DelphiWorldkaldemar: no is not included
12:25.31kaldemarthere's issue #1.
12:26.05DelphiWorldkaldemar: :)
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12:29.45Dovidj #opensips
12:31.26DelphiWorldkaldemar: worked! channel originate DAHDI/1/1544 application Dial SIP/101
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12:43.50Secret_Hamsteranyone able to recommend a dect sip phone that will work with the gsm codec and asterisk?
12:44.58WIMPyHaven't senn any with GSM. If you need low bandwidth you can get G.729.
12:46.00Secret_Hamstercurrently using the gsm codec across the board, not wanting to really have to alter and retest everything just to replace an old system
12:46.03Secret_Hamsterhum
12:46.48Secret_Hamstercurrently we have a digium iax device that connects to a analogue dect phone but it appears the iax device is dead
12:46.51WIMPyOr you stay with G.726.
12:48.37Secret_Hamsteras I said means retesting the system to use another codec
12:50.06Secret_Hamsterlooks like the business will have to dig into their pockets and allow me some testing time
12:51.17WIMPyMaybe some of the more specialised ones have gsm, like Snom M3 or some Kirk ones, but I can't remember having seen that anywhere.
12:52.33DelphiWorld:=
12:52.34DelphiWorld:)
12:52.41DelphiWorldkaldemar: how do i handle FXO incomming call?
12:52.53Secret_Hamsterjust been looking at the snom m3, having difficulty tracking down which  codecs it can handle
12:53.21p3nguinRetest?  It's quite trivial to change codecs.
12:53.32Secret_Hamsterin essence I suppose dect is not really what I need, just a portable device. So the guys can wander away from the frontdesk
12:54.00Secret_Hamsterp3nguin, business is somewhat paranoid about the phone system. Somewhat critical to the business
12:54.18WIMPyYes, dect is what you want.
12:54.29WIMPyOr CAT-iq.
12:54.41p3nguinWhat kind of load is on the box now, in the way of concurrent calls and how much transcoding is going on?
12:54.50Secret_Hamsterwell a "wireless" system is what we need. The actual air side technology is unimportant
12:55.20p3nguinWiFi VoIP phones are usually not desirable.
12:55.22WIMPyWifi is shit for voice.
12:55.51kaldemarDelphiWorld: for example by setting immediate=yes for the channel, which makes asterisk pick the call up immediately and start extension s in the configured context.
12:56.13DelphiWorldkaldemar: but i will bill the caller emediatly :-)
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13:29.12BlackBishopschmidts: hey
13:29.19BlackBishopfigured out what was wrong with the prohib thing ...
13:29.23BlackBishop( hiddin' my caller id )
13:29.38schmidtsblackbishop hi, i remeber. What was the problem?
13:29.41BlackBishopI set my phone to ignore/reject calls with a hidden id .. and forgot about it :)
13:29.47schmidtsLOL :D
13:30.12BlackBishopyeah, and I was wondering why does it ring for ~ < 1 sec then it says line busy and my phone doesn't even sing anything
13:30.13schmidtsBlackbishop the typical problem ;) its like the DND button ;)
13:30.16BlackBishop:)
13:30.50BlackBishopyeah, it came to me at about 3 AM .. I was sleeping .. I just woke up .. "I have 0000 on my block list", removed it .. tested it .. and got back to sleep
13:31.30schmidtssleep is the best debugger ;) i also know this situations
13:34.01BlackBishopnow only if the chan_datacard would send 160char messages :|
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13:34.36mducharmegood morning
13:34.56mducharmecan anybody tell me, is there a tool that will update phone numbers in LDAP with peoples phone numbers in asterisk?
13:35.39mducharmeasterisk-ldap add ons I seem to be finding are specifically geared to switching asterisk so it pulls the list of extensions etc from ldap, which I do not want
13:36.00schmidtsblackbishop i have tried to get chan_datacard running too but we only have the wrong type of huawei modems, and i didnt want to buy one just for playing around
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13:48.31BlackBishopI get K3520 kinda' cheap here 32$ .. voice unlocked and carrier unlocked ...
13:51.05schmidtsBlackbishop thats really cheap ;) we use the 1823 but without voice and carrier locked to t-mobile
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14:03.56phretorhello
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14:31.13wdoekes2BlackBishop: the datachan show device shows Use UCS-2 encoding: Yes, right?
14:31.32BlackBishopyup
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14:33.56BlackBishopwdoekes2: should it be disabled ?
14:34.13wdoekes2you could try that
14:34.47BlackBishopwdoekes2: any idea how to disable it ?
14:34.59wdoekes2no yet
14:35.00wdoekes2*not
14:35.33BlackBishop:|
14:36.29wdoekes2something like datacard cmd datacard0 AT+CSCS=0 ? perhaps..
14:36.51wdoekes2disclaimer applies
14:37.27BlackBishopAT+CSCS="GSM"
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14:38.25BlackBishopthe thing is that I get timedout while waiting 'OK' in response to 'SMSTEXT'
14:38.30BlackBishop( using pdu mode )
14:38.38BlackBishopwhen it should be using AT+CMGW for sending messages !
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14:41.42Kattyhello my asterisk does not work at all how to fix plz????
14:41.53schmidtsKatty first start with coffee
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14:42.29Kattyi'll start your coffee in a minute
14:42.36IsUpthen install XP Service Pack 2
14:43.08schmidtsIsup you forgot to restart your system first
14:43.39IsUpah, i should try CTRL + ALT + DEL
14:45.42Kattyi have a sinus infection
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14:46.38BlackBishopKatty: is your monitor plugged in ?
14:47.52schmidtsKatty ouch you should better go to a doctor ;)
14:48.41IsUpdamn, i had 2 sinus surgeries
14:48.51IsUpwhen i was 9 and 11
14:49.00IsUpright now, i am 12
14:49.32*** part/#asterisk porche (~kursad@91.188.208.204)
14:49.46IsUproot@smurfs:~#
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14:52.29KattyBlackBishop: huh?
14:53.13BlackBishopm00
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14:56.10Kattymooer.
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15:03.58schmidtssounds like asterville the new game only on irc. building cluster farms, grow peers and mow jitter :D
15:04.05IsUphahahaha
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15:18.26BlackBishopwdoekes2: well, after looking a lil' bit more at the verbose debug .. I noticed the pdu send was using "+" infront of the destination number .. and I didn't type the number in an international format ..
15:18.41BlackBishopas soon as I corrected and entered it in international format without a + .. worked like a charm !
15:19.15IsUpgreat
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15:21.39BlackBishopawesome even ! :)
15:21.43andylockranis it possible to use the asterisk management interface to initiate a call to a ringgroup?
15:22.46schmidtsandylockran by ringroup you mean ? SIP/a&SIP/b&DAHDI/1234 ?
15:23.15phretorI need to make automated periodic calls to a list of numbers. Whenever someone picks up, he or she will hear a welcome message such as "Hi, here is Bob" and record what the other party says (if anything is said). What's the fastest way to do this in Asterisk?
15:23.50IsUpphretor: use Originate on manager interface
15:24.11IsUpbut if you want to call multiple numbers, (like Voice SMS
15:24.12phretorand I can run this on a regular (say, daily) basis?
15:24.18IsUpuse call files
15:24.28IsUpyes
15:24.33IsUpuse crontab :p
15:25.04phretorIsUp: I guess that calling multiple numbers implies I have multiple lines, right?
15:25.07andylockranschmidts: not sure I follow.  I have http://dpaste.com/397454/.
15:25.51IsUpphretor: actually, forget manager interface, use call files.
15:26.08phretorok
15:26.13IsUpyou have to create call file for every number
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15:26.42andylockranYeah, call files are quite simple.. I like them.
15:26.58phretorIsUp: as long as it's text I can leverage a template and have a Python script that does the whole job.
15:27.20IsUpphretor: exactly, ive coded same thing with PHP
15:27.26phretorIsUp: great.
15:27.46IsUpi have numbers array, and its creating call files for every number and i am using crontab for make it daily basis
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15:28.25phretorIsUp: another question. Is there any good phone number provider that you know where I can buy phone numbers and set up the IP of my Asterisk box to handle the incoming calls to that number?
15:29.49Tozz_depends which country you want the numbers from
15:30.15phretorTozz_: any country would work. Although having one number per country would be easier.
15:30.23phretors/easier/better
15:30.26IsUpphretor: my example PHP script, context and crontab, hope it helps: http://pastebin.com/2mnA8S8P
15:30.35phretorIsUp: hey thanks!
15:30.43Tozz_yes but there are none or very few providers that can offer DIDs from ALL countries
15:30.50IsUpphretor: http://www.ukddi.com/ for faster sign up and testing
15:31.11phretorIsUp: do you also record the conversations?
15:31.12IsUpit provides +44 (UK) DIDs from different cities
15:31.34IsUpphretor: i am manually putting recording there, its kinda static on me
15:32.14phretorIsUp: what do you mean "manually"?
15:32.38IsUpi mean, i am recording on my Windows box, converting it to ulaw and uploading to my PBX
15:32.46IsUpi am not using Record() on PBX
15:33.09phretoris this list reliable? http://www.voip-info.org/wiki/view/DID+Service+Providers
15:34.00IsUpphretor: some providers are not working on this list, first, decide the country for DDI and google it
15:35.51phretoralright, it looks like I've everythin I need, except for speech2text, which is the last step in the chain.
15:36.14IsUpgood
15:36.18BlackBishopwdoekes2, schmidts now I wonder if I could send messages with delivery confirmation .. and somehow store it in a mysql table
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15:37.04IsUpBlackBishop: what you are using for sending SMS?
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15:39.50BlackBishopIsUp: datacard sms datacard1 <number> <message>
15:41.28BlackBishopIsUp: any idea ?
15:43.16IsUpno, sorry
15:44.33schmidtsblackbishop: http://www.smssolutions.net/tutorials/gsm/sendsmsat/
15:44.51schmidts+CSMP: 1,169,0,0OK Bit 5 SSR Set this bit to request a delivery report
15:45.59schmidtsbut it looks like CSMP isnt implementet in chan_datacard
15:46.35kaiii have problems understanding the concept of call redirection / transfer in asterisk 1.6/1.8 .. in asterisk 1.4, when a SIP peer redirected the call to another extension, asterisk said "Now forwarding 'XX/YYYY' to Local/ZZZ@context" ...
15:47.11kaiinow with 1.8, it says "Now forwarding 'SIP/YYYY' to 'SIP/ZZZ::::UDP@mydomain'
15:47.24BlackBishopschmidts: yeah, that's what I was going to say :)
15:47.25kaiiand then fails with "Not accepting call completion offers from call-forward recipient SIP/mydomain"
15:48.00phretorwhat's the best option for speech2text of the recorded audio?
15:48.01schmidtsblackbishop you have the code, go and fix it :D :D :D
15:48.13BlackBishopwill do ! :)
15:48.18BlackBishopfixed one problem for today :)
15:48.35schmidtsblackbishop problems solved while sleeping doesnt count
15:49.16BlackBishopschmidts: not that one
15:49.24BlackBishopthe problem of > 70 chars in pdu mode
15:49.34BlackBishopthe pdu generated was adding a + infront of my number
15:49.37BlackBishopwhich made it invalid
15:49.57kaiiokay it was just "promiscredir=on"
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15:54.26BlackBishopwhat does "allowed_passed_screen" the passed screen mean ?
15:55.00WIMPyBlackBishop: Maybe it would be a good idea to use SM software?
15:55.17WIMPyBlackBishop: Official version: User providede, verified and passed.
15:55.32*** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr)
15:55.41WIMPyAnd presentation allowed.
15:56.47BlackBishopWIMPy: and I guess not all providers allow this :|
15:56.47*** part/#asterisk wwalker (~wwalker@208.92.232.27)
15:57.00WIMPyWhat?
15:57.15BlackBishopto specify your own name/number that should appear on the other guy's phone...
15:57.38WIMPyProbably more like none.
15:57.50WIMPyAt least from a MT.
15:58.06BlackBishopdang :)
15:58.11BlackBishopworth a shot :))
15:58.17WIMPyVia IP or whatever, usually, yes.
15:58.35BlackBishopwell, I guess I can't use that with my datacard :)
15:58.48WIMPyBut that would not be verified then, obviousely.
15:59.03BlackBishopany way I could try to send SMS with a different number/name ?
15:59.11BlackBishopaltough I doubt they'd allow that either
15:59.15*** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl)
15:59.48WIMPyVia IP/http?
16:00.02WIMPyTry to google for bulk sms or something.
16:00.58BlackBishopneah, I was thinking about sending with my datacard
16:00.59BlackBishop:)
16:01.21*** join/#asterisk StaRetji (~BigAll@91.143.222.166)
16:01.30BlackBishophow could I send via IP/http if I don't have a voice gateway or something and agreements with them
16:01.41BlackBishopthe only way I can reach my provider is via a datacard
16:01.42BlackBishop:)
16:03.56kaiiBlackBishop: using sms webservices it is possible to set the sender number of your datacard in the sms
16:05.16IsUpBlackBishop: pswin.com
16:05.21*** part/#asterisk StaRetji (~BigAll@91.143.222.166)
16:05.57BlackBishopkaii: care to detail ? :| I don't get it ...
16:08.53BlackBishopkaii: I don't see anything in the pdu header that would allow me to change the sender number ..
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16:35.36DelphiWorldhi guys
16:35.48DelphiWorldcan please someone help me configuring asterisk 1.8 for GV?
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16:42.53WIMPyBlackBishop: Not via your catacard, but via IP instead. It's usually a lot cheaper as well.
16:43.24WIMPyAnd some even allow you to set any string as sendor.
16:43.49andylockranhey guys. Is there a way of getting asterisk to return whether or not the user hungup, or the originate timed out - when originating a call from the AMI?
16:44.48*** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr)
16:44.55BlackBishopWIMPy: well I'm not interested in other providers !
16:47.21WIMPyI've never heard of unscreened mobile access.
16:47.24DelphiWorldno one use google voice?
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16:51.31blitzrageDelphiWorld: I just follow the instructions I wrote in the book :)
16:52.04DelphiWorldblitzrage: :)
16:52.46*** join/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com)
16:52.54blitzragehttp://ofps.oreilly.com/titles/9780596517342/ch18.html#ExternalServices_id285030
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16:54.37DelphiWorldlol
16:56.39sarelonI am looking for a good comparison of asterisknow, trixbox, and elastix
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16:59.34Qwellsarelon: Nobody uses trixbox anymore.  I can't speak about elastix, because I know nothing of it.  FreePBX endorses AsteriskNOW, and it's from Digium, so it actually works.
17:01.26pifis 1.8 in debian? (even experimental)
17:01.59DelphiWorldtrixbox dead :-)
17:02.18sarelonstill uses trixboxes...but is seriously considering other alternatives
17:02.44Qwellgood decision
17:03.38*** join/#asterisk hexanol (~etienne@modemcable094.94-70-69.static.videotron.ca)
17:03.41Qa|im3r0use xivo
17:03.42DelphiWorldchan_gtalk can't by reloaded... :(
17:03.50sarelonanyone put asterisknow on a phonebochs? - any hardware problems?
17:03.52Qa|im3r0http://wiki.xivo.fr
17:04.07anonymouz666what happened to trixbox? people got tired of trying to understand why there was a dialparties agi to dial
17:05.43anonymouz666blitzrage: chapter 22 is the most interesting one :D
17:05.44*** join/#asterisk drift-_ (ae3001a4@gateway/web/freenode/ip.174.48.1.164)
17:05.49Qwellsarelon: if CentOS 5 works on it, so will AsteriskNOW
17:05.59sarelonI could rant for an hour about what's wrong with TB pro - CE always seemed like a good system to me
17:06.15*** join/#asterisk m_tadeu (~quassel@bl13-140-42.dsl.telepac.pt)
17:06.15sarelonI've had trouble in the past with things like sound cards, etc...
17:06.16russellbdoes anyone work on it anymore?
17:06.21blitzrageanonymouz666: :)
17:06.31*** join/#asterisk lanning (~lanning@208.87.233.137)
17:06.34russellbblitzrage: which chapter is that
17:06.35blitzrageanonymouz666: that's basically my astricon presentation expanded with lots of text :)
17:06.37sarelonGarrison is still championing them I think
17:06.41blitzragerussellb: Clustering
17:06.41russellbah yes, that one
17:06.48Qwellsarelon: because he wrote a book on it
17:06.56Qwellhe kinda has to..
17:07.00blitzrage:)
17:07.26*** join/#asterisk timahvo1 (~rogue@41.223.57.76)
17:08.43blitzragethat's the only reason I still champion asterisk :)
17:09.10chazzamheh
17:10.54brainiacHow can I have b chans on 1 - 9 instead of 9 - 23?
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17:12.07DelphiWorldwhy if i call myself through FXO i see 101 in my ip phone, while 101 is my ip phone callerid?
17:12.19DelphiWorldi want to show my fxo from dialplan... any idea?
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17:21.48DelphiWorldmoy!
17:22.05*** join/#asterisk metiu (~chatzilla@85-18-228-185.ip.fastwebnet.it)
17:23.01metiuhi all, does meetme need chan_dahdi.so? I can use it just with dahdi.ko loaded, but I'm having some problems, so I was wondering if chan_dahdi.so is needed at all
17:23.33drift-foo
17:23.34citywoknot sure, use confbridge instead :)
17:24.07metiuyeah I'd like to, but it was segfaulting on me in my setup (1.6.14.1), and I need Page() anyway
17:24.26metiu(1.6.2.14.1 really)
17:24.47citywokodd.  report it as a bug if you can reproduce it.
17:25.10_Corey_metiu: I'd make sure you have at least the dummy driver with chan_dahdi loaded
17:26.05metiu_Corey_: yeah, that is loaded and working, however when I play MOH from linphone, the rtp from linphone is ok, while the one from A* has random (and rare) silence packets
17:27.15_Corey_metiu: hmmm, it does sound like something is off if you're actually seeing silence packets
17:28.45_Corey_I'd check your box for timing issues, interrupt problems and the like
17:28.48metiuA* is not complaining, so I thought dahdi may be the culprit... I'd like to switch to confbridge, but Page() insists on using meetme anyway. And confbridge has no way I know of to kick users from a conference
17:29.30metiudahdi_test is seeing accurate timings, I'll try to run it while on a conference
17:30.36metiuI wonder why this happens, even if I set huge jitterbuffers on a*
17:31.12DelphiWorldmy FXO line is answering callers
17:31.19DelphiWorldand bridging it to sip phone
17:31.39DelphiWorldbut i don't want to answer it if the ip phone don't answer... any idea?
17:33.12*** join/#asterisk Cydd (~Cyd@unaffiliated/cydd)
17:36.11DelphiWorldany idea about callerid?
17:36.14DelphiWorldkaldemar: :)
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17:45.27paulcDelphiworld: Are you using Answer() before your Dial() ?
17:45.45DelphiWorldpaulc: if i use it, thay go through
17:45.54DelphiWorldpaulc: but if i don't answer, is going to demo :(
17:46.42anonymouz666DelphiWorld: you need to answer the analog line
17:46.44paulcDelphiWorld: Sounds like an issue in your extensions.conf then - you should be able to send calls from your FXO to a SIP end point using Dial(), without using Answer(), and your FXO line will only get answered if the SIP device picks up (or if something else does - like voicemail for example)
17:47.18DelphiWorldanonymouz666: i don't need to wast caller's money
17:47.54anonymouz666you need to if it is analog.
17:48.22anonymouz666if SIP you could playback something with early media (option noanswer)
17:50.27metiu_Corey_:  anyway I had filed this bug https://issues.asterisk.org/view.php?id=18314 which I think is very related, only now I'm playing a file from linphone and not from A*. I wonder why the real time microphone sound is not affected
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18:01.48De_MonI'm running asterisk 1.4 and having problems with some new SPA504G staying reachable behind NAT
18:02.12De_Mondon't have any problems with any of our other SPA's or polycom phones though...
18:03.57De_Monfor testing I tried a working phone on a working sip account without issues for years and added the new phone to the working sip account and the phone goes amber once or twice a day and can't make or receive calls until it's rebooted.
18:04.56De_Monanyone else having problems with the SPA504G series over NAT to asterisk on a public interface?
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18:10.02p3nguinde_mon: Does the phone's peer entry have qualify enabled?
18:10.53anil-limis it possible to read data transmitted over D-Channel using dahdi pseudo interface?
18:12.10De_Monp3nguin ya qualify and nat
18:12.50*** join/#asterisk rrb3942 (~rbullock@208.34.105.161)
18:12.54p3nguinUsually qualify will keep the nat open.  That's pretty unusual that the new phones don't cooperate.
18:18.09anil-limDelphiWorld, ideally whatever callerid is received from the network should be passed on via Dial(), there might be some problem in callerid identification
18:19.06anil-limyou can try Set(CALLERID(num)=xxxx) before Dial()
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18:21.18DelphiWorldmigh by
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18:38.53DelphiWorldSparFux: go out !
18:39.03DelphiWorldhow do i use meetme?
18:39.17SparFuxHi all. In DAHDI, is pri_enslave used only when multiple PRI/BRI lines are there? Or even on one single BRI, i.e. is it enslave whole PRI to another, or the B-channels to the D-channels somehow?
18:39.24SparFuxDelphiWorld: :-)
18:42.01SparFuxDelphiWorld: wtf are you doing with ASterisk?
18:42.47p3nguindelphiworld: MeetMe()
18:43.24DelphiWorldp3nguin: i should meet you... no issue!
18:43.33DelphiWorldSparFux: setting it up in my alix
18:43.53SparFuxDelphiWorld: why this?
18:44.13DelphiWorldSparFux: cause :)
18:44.15p3nguindelphiworld: Say what?  You asked how to use MeetMe, so I told you.  The answer was "MeetMe()".
18:44.29citywoklol
18:44.31DelphiWorldp3nguin: was a funy joke
18:45.09p3nguindelphiworld: Yeah, it did make me laugh, but only because I thought your response was unbelievable.  :)
18:45.18DelphiWorldp3nguin: :)
18:45.50SparFuxDelphiWorld: or.....: Bridge(FS(mod_conference)) :-P
18:45.58DelphiWorldSparFux: iax
18:46.06SparFuxyeah,ok.
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18:50.39DelphiWorldIsUp! IsUp! IsUp!
18:55.53IsUpah hey DelphiWorld
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19:19.08DelphiWorldIsUp: should by down!
19:20.02IsUp^^
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19:24.51as001Hi is it possible to change music class for agents logged in queue on so they does not need to logout and login again to listen new music class ? I tried with CHANNEL(musicclass) without success.It seems it controls just moh for calls placed on hold.
19:25.07m_tadeuI have 2 sip clients outside the server nat, and when the call is done from client 1 to client 2 everything goes fine, but if the call is from client 2 to client 1, no sound goes through after answering the call
19:25.29citywoksounds like a nat/firewall issue
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19:26.03m_tadeuclient nat or the server nat?
19:29.55citywokpotentially on client1's end.  sounds like client1 only works if it initiates the call.
19:29.59anonymouz666~sipnat
19:29.59infobot[~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions .  Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet.
19:33.11DelphiWorldanonymouz666: strangest things
19:33.30DelphiWorldanonymouz666: my asterisk chan_gtalk is behin nat and no issue both in and out!
19:34.21anonymouz666your asterisk is powered by Delphi and coca-cola
19:34.53DelphiWorldanonymouz666: no joke... seriously why i don't have nat issue?
19:36.32anonymouz666gtalk isn't SIP
19:37.21DelphiWorldanonymouz666: but rtp
19:37.47*** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net)
19:38.54*** join/#asterisk m_tadeu (~quassel@89-181-68-4.net.novis.pt)
19:39.28*** join/#asterisk awclin (~alinford@g0962184.demon.co.uk)
19:49.24*** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net)
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19:50.53MrTelephoneCouple questions. Asterisk 1.4.33 still wants to authenticate with type=friend,host=***.***.***.***,insecure=very. How do we disable auth between asterisk machines? I use 1.4.24.1 and it accepts calls from 1.4.33 but not the other way around.
19:52.28*** join/#asterisk atan2 (~atan@unaffiliated/atan)
19:54.31*** join/#asterisk nny (~Scott@174.107.201.103)
19:55.49nnyresearching an issue, wondering if it's something that could be handled by vanilla asterisk. I have a queue that I want to failover after 5 minutes to another queue. This is if the agents are logged in but on the phone. Is this an option in the queue app or handled by dialplan directly?
19:57.47fauxalliancehttp://www.freepbx.org/support/documentation/module-documentation/queues @ nny  |grep overflow
19:58.36m_tadeudo I have to set the STUN hostname in the client directing to my asterisk server?
19:58.46fauxalliancem_tadeu, NEVER
19:58.47nnyfauxalliance: i'll take it back to freepbx
19:58.52m_tadeuor any STUN server can do
20:00.20m_tadeufauxalliance: so I leave it set to some other server?
20:01.16fauxalliancem_tadeu, what exactly is the relationship between asterisk and STUN?
20:03.50fauxalliancethere is none... STUN belongs outside of Asterisk....
20:04.10p3nguinIt's a rocky relationship.  STUN often cheats on Asterisk with other clients.
20:04.36m_tadeufauxalliance: and asterisk doesn't use for anything?
20:04.46fauxalliancem_tadeu, better not...
20:05.29m_tadeufauxalliance: I thought it was required when a sip client was behind another nat
20:05.29wizard171Arg, I am STUNned ... Set Phasers to ... Fabulous ... instead :)
20:06.02Kobazwhat settings should i use for a sip user that match the defaults for an iax user (ie jitter settings and stuff)
20:06.18Kobazover a crappy network, the default iax settings work 100x better than the default sip settings
20:06.47MrTelephoneAnyone know why asterisk 1.4.33 forces digest authentication?
20:06.53Kobazor maybe there's just better jitter buffering in the iax channel driver?
20:09.03fauxalliancem_tadeu, or just a properly configured NAT
20:09.12*** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au)
20:09.27m_tadeuso I have client 1 with a public ip that is unable to call to client 2 behind a nat...client 1 has nat=no and client 2 has nat=yes and qualify=yes
20:10.07*** join/#asterisk Defraz (~Defraz@63.226.95.152)
20:10.15p3nguinm_tadeu: I have Asterisk behind NAT and SIP devices outside, behind another NAT... without STUN.
20:10.25fauxalliancep3nguin, we all do.
20:10.31p3nguinexcept for m_tadeu
20:10.55fauxalliancehttp://www.asteriskguru.com/natut.php @ m_tadeu
20:11.02p3nguinm_tadeu: Is your Asterisk behind NAT or on public IP address?
20:11.08lanningcanreinvite=no?
20:11.13fauxalliancep3nguin, it has pretty diagrams...
20:11.34fauxalliancem_tadeu, "Without STUN support, you will also need NAT=yes"
20:12.01fauxalliancethen Asterisk will do all the magic for you... (whilst configured)
20:12.01*** join/#asterisk species86 (~Species@c053880-02-v1031-static.nvrninaa.cinergymetronet.net)
20:12.06m_tadeup3nguin: behind a nat...note that I don't have STUN...was just checking the documentation and they say I should use STUN
20:12.28species86anyone here have any experience connecting a mitel phone system to asterisk
20:12.34fauxalliancem_tadeu, no STUN...ever.
20:12.37wizard171m_tadeu, is client2 behind something doing SPI (stateful packet inspection)? If so, can you turn it off? You issue sounds like you can not "initiate" a conversation with the client from "outside" ...
20:12.56fauxalliancem_tadeu, no SonicWall nonsense...
20:13.09fauxallianceno call log either...
20:13.46_Corey_species86: yeah, there are a couple ways to tackle that
20:14.25_Corey_species86: I guess it really boils down to what kind of Mitel are we talking about? SIP enabled?  What kind of integration are you looking to achieve?
20:16.13drift-hrm
20:19.24*** join/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com)
20:36.37*** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave)
20:36.48*** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com)
20:37.28*** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com)
20:37.39m_tadeuchecked everything you guys suggested....still no luck
20:38.49hobodavehey guys, I'm using AGI with 1.6. How do I get a Playback to work when the absolute TIMEOUT is reached? I have this currently, but I never hear the goodbye: https://gist.github.com/817177
20:39.42hobodaveI get this error: [Feb  8 12:39:19] WARNING[18459]: res_agi.c:2971 run_agi: No channel, no fd?
20:40.12hobodaveI've googled this of course, and found references to using DeadAGI, but then when I consulted those docs it appears it is obsolete in 1.6, and AGI should somehow do what I need
20:41.08*** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr)
20:41.58MrTelephoneis there a workaround when you sip trunk two asterisk boxes together that have the same accounts on each side? The remote asterisk box always wants to authenticate by username even though I have an entry at the bottom of the file for unauthenticated connections.
20:42.55MrTelephoneIf I use fromuser than the callerid(num) gets wiped out from the sip headers..
20:43.20m_tadeuI think I got it....in the database I was setting qualify=null instead of qualify=no...the default in sip.conf was qualify=yes....damn...too many working hours
20:43.56m_tadeuthanx a lot guys
20:45.54MrTelephoneI guess inserting deny statements in all the sip users works.
20:46.48MrTelephoneor does it make more sense to make usernames with mac addresses and then do fromuser=NUMBER
20:47.02m_tadeunot in all....only works well if I deny the ones with a public ip
20:47.40citywokMrTelephone: what?  why not just have a sip peer for the other box in each sip.conf?
20:48.58wizard171hobodave, your post came across with "#" not "t" ... ergo, what exactly made you believe it would go there ? (maybe I am confused?)
20:49.15hobodavewizard171: there is a t there
20:49.19hobodaveline 8
20:49.22MrTelephoneThe second asterisk box wants to authenticate the sip client no matter what...
20:50.05MrTelephoneI have this entry which is a no brainer;
20:50.07MrTelephonecontext=from-ast
20:50.07MrTelephonefromuser=asterisk1
20:50.07MrTelephonetype=friend
20:50.07MrTelephonehost=192.168.90.5
20:50.07MrTelephonedtmf=rfc2833
20:50.07MrTelephonecanreinvite=no
20:50.39MrTelephonebut because the call comes from a device that has a username that is also in the sip.conf it wants to authenticate..
20:50.40citywokdon't paste lots of stuff in here
20:50.41citywok~pb
20:50.42infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude.
20:50.49MrTelephonesorry, was a small paste
20:51.48MrTelephoneIt makes sense that it tries to authenticate. Maybe I should just try moving that line to the top of sip.conf. You think it will matter?
20:52.17citywokare you attempting to load balance peers or something? why do they exist in both servers?
20:52.41MrTelephoneyeah load balancing
20:53.14MrTelephoneUsername matching takes presidence over IP matching
20:53.31*** join/#asterisk Ad-Hoc (~nimbus@62.1.165.6.dsl.dyn.forthnet.gr)
20:53.33wizard171hobodave, ah, the infamous "goto" ... :) I missed it ... is this dialplan snippet in a "macro" or "gosub" ?
20:53.39citywokare you just using round robin dns matching to register?
20:54.11MrTelephoneWill be using SRV records. Even though you set it up as failover you are going to end up getting some devices connecting to both
20:54.36hobodavewizard171: I don't know what those are. This is our [default] dialplan
20:54.37MrTelephoneI'm working on a mesh dialplan but matching peers/users is my first real roadblock
20:55.11hobodaveI've tried it without the Goto fwiw
20:55.17citywokwell, something is wrong if the other * box is trying to authenticate the original calling device.  the call should be coming from the authenticated * server, and it shouldn't ever really "see" the calling phone.
20:55.17hobodaveI copied the Goto from the [demo]
20:55.28MrTelephonedeny=IP_OF_ASTERISK/255.255.255.255 prevents username matching and it works.
20:56.14citywokwork on getting the link between your * servers working properly, you shouldn't be denying your * servers from talking to each other
20:56.20MrTelephoneI have canreinvite=no and the sip dialog shows From: JOHN SMITH <2929129@192.168.90.10>
20:57.04citywokcanreinvite controls whether or not * will remove itself from being in the media path if possible.  won't have an impact here.
20:57.15m_tadeunow both clients can make direct calls to each other....but if client 1 calls and is sent to a queue, the same thing happens...it gets moh but no sound after client 2 answers
20:57.16wizard171hobodave, ah, is "autofallthrough=no" by chance ?  if so, when the "500" runs out of things to do, it gets "hangup" instead of "timeout" ...
20:57.47hobodavewizard171: yes that is the case
20:57.53citywokpb your sip.conf
20:58.11hobodavewizard171: wait maybe not
20:58.11MrTelephoneI denied the peer from authenticating to my asterisk server because I want the asterisk server to authenticate to the peer i pasted in the channel earlier
20:58.24hobodavewizard171: I take that back, it is commented out
20:58.37citywokare you hard coding the ip of every phone?
20:59.19MrTelephoneno, but is there a setting where you can force priority of a peer match?
21:00.12MrTelephoneIf you block caller presentation from a sip ata device then asterisk using the registry information to peer match
21:00.20citywokhttp://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer
21:00.42MrTelephonemaybe I should just try and have asterisk register with each other
21:01.52citywoki'm pretty sure what you are doing is wrong. lol
21:01.53MrTelephonei'll give it a try
21:02.02MrTelephoneset to peer instead of friend
21:02.26wizard171hobodave, duplicate your "t" entry with "h" ... try it again ... or, put exten => 500,n,waitexten(10) after the AGI ... test it and see ... eh?
21:02.41*** join/#asterisk darkskiez_ (~dz@62.50.249.253)
21:04.18hobodavewizard171: I added an "h" entry and it _tries_ to play that but it goes unheard and this warning is logged
21:04.21hobodave[Feb  8 13:03:42] WARNING[21805]: file.c:753 ast_readaudio_callback: Failed to write frame
21:05.12wizard171hobodave, yeah, the channel is gone ... it got hung up ... use the "waitexten" after the AGI instead ...
21:05.49MrTelephonecitywok, what would you suggest for a sip ata endpoint with 2 ports. They use the same ip and source port but line2 screens their identification so asterisk matches to line1 with the wrong credentials :(
21:06.30citywoki'd suggest adding it to your * box as a sip peer. if the ata can have 2 sip peers to differentiate the ports... then use 2 different sip peers.
21:08.00hobodavewizard171: still hanging up :(
21:09.01citywokMrTelephone: how do you plan on figuring out if the peer is registered on the other server?
21:09.24citywokjust send the call to the local peer, and the other server and let them deal with it?  not the most efficient way to do things :)
21:09.27hobodaveaccording to the docs you can't Playback from an h
21:09.44citywokhobodave: yea, it's the hangup extension.  once you are in h the call is being torn down.
21:09.53wizard171hobodave, can you set "autofallthru=yes" ... explicitly ...
21:10.11hobodavewizard171: no because I don't understand what it's doing, but it seems like it will affect my other extensions
21:10.35hobodavethe h thing already was running on my other extensions, it thankfully just wasn't doing anything
21:11.01hobodaveunfortunately I have no staging environment besides a different extension on our production box
21:13.09hobodaveI'm aboutto give up and just let it kill the call without saying goodbye
21:13.48wizard171hobodave, ah, just for laughs, what happens if you put the "goodbye" just after the AGI call ?
21:14.09_Corey_I was just going to say the same thing...
21:14.11_Corey_:)
21:14.23hobodaveI tried that before, but I will do so again to be thorough
21:14.28wizard171Great ... somethings ... minds ... maybe ?
21:14.49hobodavecall terminates, this logged:
21:14.50hobodave[Feb  8 13:14:38] WARNING[23104]: res_agi.c:2971 run_agi: No channel, no fd?
21:16.04wizard171hobodave, that sounds like the AGI is not "running" ... and when it "errors" it "hangsup" ...
21:16.15hobodavethe AGI is running
21:16.35hobodaveunless you meant "something that doesn't mean running at all" by quoting the word running
21:17.22hobodaveall I'm trying to do here is add an absolute call timeout
21:17.31MrTelephonecitywok, just by CHANUNAVAIL, check asterisk1(local), if not there, check asterisk2(remote), if not available call cannot be completed.
21:17.31hobodaveI've done that, but it rather ungracefully kills the call
21:17.44*** join/#asterisk vinhdizzo (~vinh@75.85.182.165)
21:17.51MrTelephoneand vise versa
21:19.31wizard171hobodave, sorry, perhaps I was not clear ... to me its possible the AGI was not working at all ... or ... when the 500 extension runs out of things to do, because "autofallthru=no", the channel gets "hangup" ... or ... something in the AGI is causing the channel to get "hangup" ...
21:20.30citywokautofallthrough=yes means after dial() it goes to h.  autofallthrough=no means after dial it is supposed to continue on in the dialplan.
21:20.31wizard171hobodave, the warning you are getting sounds like the AGI is getting SIGHUP ... like the channel is being torn down while its working ...
21:20.39hobodavewizard171: the AGI is definitely running, and working, but it's quite likely it _is_ hanging up the call
21:20.59hobodaveas far as I know the AGI _does_ hang up the call, just glancing at the code
21:21.01hobodaveshould it not be?
21:21.26wizard171hobodave, perhaps it could "goodbye" just before it does that ... eh ?
21:22.15hobodavewizard171: well, it does that actually
21:22.21citywokhobodave: if you hangup the call in the agi, how do you plan on using dialplan for the call to play something?
21:22.27hobodavewizard171: the problem is the TIEMOUT(absolute) is hardcore pulling the plug on the AGI
21:22.34citywokthe call has already been hung up, so you can't do anything
21:22.42hobodavecitywok: I don't need to
21:22.48_Corey_um, ok
21:22.50_Corey_i have a solution
21:23.03_Corey_use dial with the timeout flag and call the agi via a local channel
21:23.09_Corey_the dial will play the goodbye for you
21:23.12wizard171hobodave, if you tell asterisk in the AGI to tear down the call , the dialplan can not "play" something after that point ...
21:24.06_Corey_the Dial argument you want is: L(x[:y[:z]])
21:24.24hobodaveI'm going to try to noop the AGI's "hangup" function first
21:24.29hobodavebecause that's something I understand
21:25.03_Corey_um, if the absolute timeout is killing the agi it cannot do media any further when it gets the signal
21:25.25hobodaveif the diaplan can play the sound that's fine
21:25.44hobodavemy understanding of what wizard171 is telling me is that I don't need to hangup the line in the AGI
21:25.50_Corey_well, you can't play it after the agi
21:25.58hobodaveeven if the AGI doesn't hang up the line?
21:26.38_Corey_well, if I'm understanding what you're trying to do correctly, the only to accomplish this with a call limit is to use something like the Dial's L flag
21:26.44wizard171hobodave, then the AGI can not do "hangup" and the dialplan, afterward, play the file ... the AGI can not use all of TIMEOUT(absolute)=10 to end either ...
21:26.55_Corey_so you could call the AGI from behind that via a Local channel and achieve what you want
21:27.09hobodaveI'm having a really tough time understanding you wizard171
21:27.43wizard171hobodave, the AGI = no hangup, and it must finish what it does *INSIDE* the 10 seconds you give it to work ...
21:28.02hobodavewizard171: ok?
21:28.13wizard171hobodave, and your dialplan then needs the "goodbye" after the AGI line ...
21:28.29*** join/#asterisk darkskiez_ (~dz@62.50.248.245)
21:28.46hobodavetesting
21:28.59wizard171hobodave, no offense meant, sometimes my mind "sees" what my fingers ... just can not seem to type ... eh ?  :)
21:29.36wizard171but then, I have been know to quote ... bad poetry ... in public as well ... :)
21:29.36hobodavecall still hangs up, still logs: WARNING[24916]: res_agi.c:2971 run_agi: No channel, no fd?
21:31.12hobodavemy AGI is not hanging up the line
21:31.15hobodaveit just exit()'s
21:32.12hobodavescrew it
21:32.20hobodavetelling my boss to hire a consultant if she wants a goodbye message
21:32.21_Corey_lol
21:32.44hobodavethanks for trying guys
21:32.48*** part/#asterisk superbeef (~Lane.Jenn@74-120-209-234-rev.redanvil.net)
21:33.24hobodaveis there a way to have the TIMEOUT apply globally without needing ot add it to each extension?
21:34.56*** join/#asterisk fofware (~Fabian@host45.190-31-21.telecom.net.ar)
21:36.46_Corey_hobodave: What is the objective?
21:37.23MrTelephonecitywok: Max depth is 1 machine so it shouldn't be too bad. Most peers should be the primary server. peer doesn't work on endpoints with multple accounts same ip/port.
21:37.26hobodave_Corey_: I have a dozen "extensions" that are really full 10 digit US phone numbers - e.g. exten => 7035551212,1,Agi(asterisk.php)
21:37.37MrTelephoneso I still need the deny statement
21:37.40hobodave_Corey_: I'd like to apply Set(TIMEOUT(absolute)=120) across all of them
21:38.53hobodaveWith my current set of knowledge I know I can simply add a Set(TIMEOUT ... in front of each extension
21:39.02hobodavebut I don't like to needlessly duplicate work
21:39.24citywokis this a failover based plan, or a load balancing based plan?
21:39.40hobodaveI don't know what that means :-\
21:39.54citywokthen you probably shouldn't be doing what you are...
21:39.56hobodavethis is a phone interface to our API, each number is for a different one of our customers
21:40.04hobodavecitywok: probably not, but alas I was tasked to do so
21:40.14citywokwhy are you using 2 servers?
21:40.28hobodaveI'm not, where do you get that I'm using 2 servers?
21:40.34_Corey_hobodave, so a XXXXX,1,Set(TIMEOUT... and XXXXX,2,AGI() wouldn't work?
21:40.46wizard171hobodave, citywok is talking to MrTelephone ...
21:40.53*** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164)
21:41.00hobodaveblah
21:41.08citywok_Corey_: oh you are right, crap i got the 2 people confused. my bad.
21:41.22_Corey_;)
21:41.30hobodave_Corey_: yes that would work, but the issue I have is that I would prefer not to have N copies of the Set(TIMEOUT for N lines
21:41.38hobodaveIs it not possible to set a "global" timeout?
21:41.45hobodaveif not, I'll just suck it up and go forward
21:42.56wizard171hobodave, maybe I don't really understand ... you want to forcefully limit ALL calls on your server to 2 minutes?
21:43.02_Corey_hobodave: it depends on what else is in the dial plan I guess.  If these numbers are similar, I'm suggesting you pattern match
21:43.02hobodavewizard171: yes
21:43.28_Corey_hobodave: Why not just have _X.,1,etc  then?
21:43.37_Corey_you could have a two line dial plan if you want
21:43.38hobodave_Corey_: :-\
21:43.43hobodaveI don't know what this stuff means
21:43.50hobodavethe "front end" was outsourced
21:44.04hobodavee.g. asterisk configuration and the PHP AGI
21:44.05_Corey_oh, ok  :) sorry
21:44.22hobodaveI wrote the API backend, there is no documentation or maintainer for this stuff
21:44.33hobodavejust me, so I'm learning on the fly mostly by just looking at what was already there
21:44.45_Corey_Well, I would start by reading the chapter on the dial plan in the book
21:44.56_Corey_You're going to save yourself a lot of time
21:46.09MrTelephonelive failover. I'm ordering an adtran atlas 550 so I can oversubscribe a DS1 and give my asterisk boxes each a DS1 that shares timeslots on the telco side. I'll be able to run 2 asterisk boxes and if I unplug one of them the other should automatically failover without worrying about switching ip addresses and using heartbeat.
21:46.53*** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net)
21:46.53citywokheartbeat would probably be easier...
21:47.18*** part/#asterisk SparFux (~raoul@rl2-1-pt.tunnel.tserv5.lon1.ipv6.he.net)
21:47.23*** join/#asterisk joobie (~joobie@mx01.anric.com.au)
21:47.28citywokonce you get it configured, it's brilliantly simple as long as you make sure your servers are mirrors of each other, but you have to do that anyways with your current strategy
21:47.48MrTelephonehow long does it take to switchover?
21:47.49*** join/#asterisk vinhdizzo (~vinh@75.85.182.165)
21:48.06MrTelephoneI'm worried about arp timeouts
21:48.09citywokabout half a second, less time than it will take for your phones to figure out the server disappeared, re-query DNS, and get the other IP
21:48.25citywokwhich could be up to a couple minutes
21:48.58MrTelephoneso it takes a couple of minutes for the clients to find the new asterisk box?
21:49.01joobiei stumbled in on this convo mid way - what sort of redundancy are we talkinga bout?
21:49.18hobodave_Corey_: this is what I went with: https://gist.github.com/4b3d3cf11b5fefe0547f
21:49.22hobodaveall 50 lines of it
21:50.32citywokMrTelephone: well, your clients all have to go to your other boxes IP if they are on the one that fails
21:50.48citywokif you use heartbeat w/ a VIP, the IP moves to the other box and the clients don't have to worry about it.
21:51.15*** join/#asterisk [Outcast] (~anonymous@64.202.62.5)
21:51.43MrTelephoneI'm going to consider heartbeat. But I was planning on purchasing another PRI. If one of them fails then I can route calls through the sip trunk out the other PRI.
21:51.48hobodavethe x's there are me obfuscating the real digits
21:52.26citywokyea you're still stuck with the T1 issue, i run pure sip so all i do is move my IP
21:52.32joobieheartbeat w/ a VIP is for a LAN failover type setup right?
21:52.41joobieit doesnt work in WAN scenarios does it?
21:52.55citywoksure, you can have it move the external IP's as well
21:53.11joobiethis is going to be more expensive though
21:53.18joobieyou need to consider BGP, etc to do something like this
21:53.21hobodavesigh it's so ghetto that it just hangs up
21:53.22joobiein a WAN scenario
21:53.25jkroonhi guys, is there an AGC (automatic gain control) application for asterisk?  Specifically for dealing with some analog (DAHDI) lines?
21:53.33citywokoh, you mean if the entire WAN Link fails.
21:53.40joobienod
21:53.53citywokyes, if that's what you are going for you will have to make sure your IP can move between links and your carrier can handle it. :)
21:54.02joobieching ching
21:54.03joobie:P
21:54.05MrTelephoneImagine if the telcos could provide DID's that used both TDM and SIP circuits
21:54.23citywokMrTelephone: some can do failover from SIP to TDM
21:54.30_Corey_hobodave: Yeah, you really could do this with two lines
21:54.31citywokno idea if any provide it the other way around
21:54.32joobieMrTelephone, in AU the DID's are assigned to the telco's switch
21:54.39joobiethe switch routes that DID
21:54.41hobodave_Corey_: I'm reading about _'s now
21:54.52joobiewe had an instance where the datacenter where our ISDN terminates went down last year
21:54.53_Corey_hobodave: and the "goodbye" thing is easy too if you study the Dial command with the call limit
21:54.58joobiewhich dropped our ISDN
21:55.08joobiebut the ISDN provider was able to redirect calls to our DID's to another number
21:55.19joobieif the ISDN providers switching goes down, you are screwed
21:55.24joobieI mean that DID is allocated to a provider
21:55.27MrTelephoneYeah it's hard to find a provider in rural Canada that provides such services.
21:55.31joobieif their core equipment drops, there wont be any failover
21:55.42hobodave_Corey_: does a real extension take precedence over a _XXXX extension?
21:55.54_Corey_hobodave: depends on the sequence in the file
21:56.04hobodavenice
21:56.08_Corey_hobodave: Put the real ones first
21:56.10Qwellit's best match
21:56.15Qwellorder doesn't matter
21:56.23_Corey_oh yeah? lol
21:56.27joobieMrTelephone, you need to define what you want to protect with your failover first
21:56.39p3nguinasterisk-h323 is the official h323 channel driver for debian, right?
21:56.46joobieheartbeat will work well if you want asterisk server protection, in that if your WAN/LAN is still functional - heartbeat is good for this failover
21:56.50_Corey_I guess I just do it for the aesthetic :)
21:57.03joobiebut if you are talkinga bout DID failover, there are many other points of failover before it reaches your asterisk box
21:57.16hobodaveso _Corey_ is this what I need? https://gist.github.com/817bade7f45732999687
21:57.36citywokjoobie: that's what he is going for with 2 seperate servers :)
21:57.38_Corey_yeah, that could work
21:57.47joobieahh
21:57.48_Corey_i didn't count the X's but it looks right
21:57.52hobodave10
21:57.56joobiejust need to be aware it will provide asterisk / server redundancy
21:58.01hobodaveI figured that was better than _X!
21:58.01joobienot true "DID failover"
21:58.07hobodaveok here goes
21:58.10hobodavewatch me break everything
21:58.34MrTelephoneI have a feeling heartbeat wont' work for me without a lot of configuring. What if you lose your t1 card and your ethernet network is still up?
21:58.53hobodavewell that was easy!
21:59.10hobodave_Corey_: you have no idea how much time has been wasted by us not keeping the damn configs in sync between our primary and backup server
21:59.18_Corey_lol
21:59.19hobodaveI could punch whoever we paid to do this
21:59.33citywokhobodave: crontab + rsync
21:59.41MrTelephoneI made a backend database with perl scripts that generate configs. What other way can you keep them sync?
21:59.47hobodavecitywok: the point is it was entirely unnecessary
22:00.42hobodave_Corey_: so for kicks I should lookup this Dial() thing?
22:01.02_Corey_hobodave: if you want to play goodbye :)
22:01.25hobodave_Corey_: I've been readint his page: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
22:01.25_Corey_but you're going to have to also check out the Local channel concept first
22:01.32hobodaveand I'm not sure how that fits my scenario
22:01.36hobodaveok I'll look at that
22:01.48*** part/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com)
22:02.17_Corey_essentially, the AGI is going to be at the other end of a fictional "channel" that you're dialing with the Dial command
22:02.39_Corey_the Dial command has a built-in option to do a call limit with a hangup WAV for calling card applications and such
22:02.49MrTelephonecitywok, have you ever setup a cluster type system?
22:02.51_Corey_we use it for all kinds of things like that
22:03.32citywokMrTelephone: i use heartbeat in my current test lab (which i plan to implement in production soon), and in my production system i use a manual failover system which i wrote.
22:04.05citywoki didn't trust failover at the time and didn't want to run in to any false-failover attempts, but in testing recently heartbeat is very stable.
22:04.30MrTelephoneI just don't want it to crash hard when I'm on holidays
22:04.37hobodave_Corey_: I half-grok what I'm doign here. do I always refer to the extension as _XXXXXXXXXX ? or do I use s ?
22:04.59_Corey_not sure what you mean
22:05.08hobodavelet me throw an example
22:05.10*** join/#asterisk delroy (~delroy@tba.usask.ca)
22:05.14_Corey_s is usually used when you don't have a number being dialed
22:05.28MrTelephonecitywok, we have a lot of customers at home so we can't be down too long or else shit hits the fan
22:06.30p3nguinThe 's' extension is not a catch-all nor a wildcard extension.
22:06.54hobodave_Corey_: https://gist.github.com/833e1cbe9b6c17a111fb
22:07.14_Corey_ah
22:07.16_Corey_ok
22:07.24hobodavethat's what I'm kinda getting from http://www.voip-info.org/wiki/view/Asterisk+local+channels
22:07.39_Corey_You're getting closer
22:07.47delroyQuestion:  I got 2 locations and 1 asterisk server in each
22:08.03_Corey_you would do Dial(Local/1234567890@myagi,whatever)
22:08.19p3nguinhobodave: The underscore indicates that you are going to be using pattern matching for extensions.  The X matches numbers 0-9.
22:08.20delroyAll calls come into location A.  How can I make extensions in Location B appear in  directory on A?
22:08.22_Corey_hobodave: Keep in mind the 120 you're using there is a ring timeout and won't really apply
22:08.32hobodave_Corey_: but how would I know the extension number that if I'm matching it with a pattern?
22:08.40delroyI am using Asterisk Now and FreePBX
22:08.43hobodavegrammar fail
22:08.51_Corey_hobodave: use ${EXTEN}
22:08.57_Corey_it will repeat what was dialed
22:08.57hobodavesigh
22:08.59hobodave:)
22:09.01sled-dogI'm getting this on a slackware box when trying to ./configure:   "configure: OS type  : linux-gnu   |  configure: Host CPU : x86_64   |  configure: build-cpu:vendor:os: x86_64 : unknown : linux-gnu :  |     configure: host-cpu:vendor:os: x86_64 : unknown : linux-gnu :
22:09.05sled-dog"
22:09.11Qwellsled-dog: and?
22:09.16sled-dogit bails
22:09.24Qwellbecause it's done
22:09.36sled-dogfair enough
22:09.42hobodave_Corey_:  https://gist.github.com/833e1cbe9b6c17a111fb - Now I should be able to add the TIMEOUT foo to this?
22:09.44sled-dogthought it was an error
22:09.48*** join/#asterisk elihunter_ (639298d9@gateway/web/freenode/ip.99.146.152.217)
22:10.11draeathThis is perhaps a silly question. Does asterisk care about whitespace? IE, "extn=>1000,1,foo(bar,bar)" vs "extn=>1000, 1, foo(bar, bar)"
22:10.12_Corey_hobodave: Ok, you're close enough...  give me a sec :)
22:10.22draeathspecifically regarding dialplans
22:10.30hobodave_Corey_: appreciate it :)
22:10.37Qwelldraeath: in some versions, yes.  best to not include it.
22:11.00delroy<PROTECTED>
22:11.08draeathDoes support come and go? What's the "current/moving-forward" support?
22:12.06hobodave_Corey_: I'm trying something on my side too, it should work, in my head anway
22:12.33elihunter_Hoping someone can help me.  I'm getting random poor call quality.  I'm using voipmonitor to try to figure out what's going on and am seeing latency gradually increase during some calls
22:13.38_Corey_hobodave: Something like this: Dial(local/${EXTEN}@myagi,5,L(120000))
22:13.58hobodaveah
22:14.05hobodaveI was trying TIMEOUT again, but I dont' even need it
22:14.20_Corey_nope
22:14.27_Corey_you can set this variable: ${LIMIT_TIMEOUT_FILE}
22:14.42_Corey_to a WAV of your choosing but I believe by default it just says 'goodbye'
22:15.28hobodaveplayed nothing by default
22:15.31hobodavetrying the variable thing
22:15.53hobodavehow would that be set? I don't see an example in extensions.conf
22:16.40hobodaveah Set
22:17.22_Corey_Set(LIMIT_TIMEOUT_FILE=whatever)
22:17.26_Corey_before the Dial
22:17.28hobodave_Corey_: badass!
22:18.18_Corey_:)
22:18.33hobodavethanks so much :)
22:19.37_Corey_hobodave: Just /msg me if you guys need some consulting :)
22:19.51hobodave_Corey_: sure thing :)
22:21.47p3nguinWhat h323 package(s) do I need to install on Debian in order to get chan_h323 support in Asterisk 1.4 compiled from source?
22:29.11*** join/#asterisk Nugget (nugget@carrera.macnugget.org)
22:29.19*** join/#asterisk rrb3942 (~rbullock@208.34.105.161)
22:34.25*** join/#asterisk Nugget (nugget@carrera.macnugget.org)
22:39.40elihunter_Can somebody help troubleshoot call quality issues?  I'm running asterisk 1.4.38 with dahdi 2.4.0 and am seeing some weird latency issues and dahdi timing issues
22:39.59*** join/#asterisk DelphiWorld (~VoIpGuy@41.200.9.119)
22:40.03DelphiWorldguys
22:40.15DelphiWorldhow do i specify 100 to... 199 as my local exten?
22:40.36DelphiWorldanonymouz666: any pathern that specify calling from 100... to 199?
22:40.53*** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net)
22:41.03QwellDelphiWorld: Read about patterns in the book.
22:41.04Qwell~book
22:41.04infobot[~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, or http://ofps.oreilly.com/
22:41.30DelphiWorldQwell: if is html then is welcome. no PDF, sory.
22:45.34citywoki don't think he cares what formats you can open, you don't need to apologize to him.
22:45.47chazzamthere are optioms for both of those in that description...
22:46.12citywokgiven the broken english i'm guessing he failed to notice that
22:46.25DelphiWorldQwell:  exten => _1XX,1,Dial(sip/${EXTEN})
22:46.31DelphiWorldQwell: is this right?
22:46.44citywokdid you try it?
22:46.55DelphiWorldsitydo you have eyes?
22:47.14citywoknope, i'm blind and use text to voice software
22:47.26_Corey_lol
22:47.45DelphiWorldcitywok: realy or are you joking? cause if you are realy blind you *SHOULD* not say this
22:47.47citywokwait, you were talking to whoever sitydo is, must not be me
22:47.52*** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox)
22:47.58DelphiWorldcitywok: is you, no other
22:48.01citywokstupid translation software pronounced it about the same
22:48.11DelphiWorldcitywok: here we go... cause is me that use text to speech, not you.
22:48.33citywokwell, you should get new software, yours doesn't appear to work very well.
22:49.04DelphiWorldcitywok: develope it for me?
22:49.10*** join/#asterisk quintana_ (~sylvain@modemcable198.162-82-70.mc.videotron.ca)
22:49.36citywoki don't develope
22:50.00DelphiWorldcitywok: so stop shiting
22:50.12*** part/#asterisk quintana_ (~sylvain@modemcable198.162-82-70.mc.videotron.ca)
22:50.38citywoki was just asking you if you tried it, you were the one that started "shiting"
22:52.09draeath1: spelling. 2: er, ok? simmer down ya'll :)
22:52.37*** join/#asterisk yoda1410 (~yoda@aether.hipocoon.be)
22:52.49p3nguiny'all
22:52.54citywokdraeath: i now huw to spiel gud, aite?
22:53.00draeathyarp
22:53.13DelphiWorldexten => _1XX,1,Dial(sip/${EXTEN})
22:53.41citywok[14:46:56] <citywok> did you try it?
22:54.33*** join/#asterisk CrashHD (~na@209.65.244.26)
22:54.53seanbrightthat looks right
22:54.54CrashHDwhat is the goto asterisk / gui / free pbx these days?
22:55.09citywokCrashHD: AsteriskNOW
22:55.16wizard171DelphiWorld, exten => _1XX,1,Dial(sip/FILTER(0-9,${EXTEN})) ... would be more "best practice" ...
22:55.41DelphiWorldwizard171: while try :-)
22:55.45Qwellwizard171: why?  the match there is known
22:55.47CrashHDcitywok: it displace trixbox ce?
22:56.02citywokCrashHD: no idea, i don't use either. just going based on what other people around here have said.
22:56.36Qwellmy cup o' noodles displaces trixbox ce
22:56.40citywoktrixbox uses freepbx's interface afaik, which asterisknow does too (or their own interface), and asterisknow is made by digium for which i believe you can get paid support if you want
22:56.41CrashHDlol
22:56.46CrashHDQWELL!
22:56.49CrashHDnot a fan by any means
22:57.04CrashHDahh
22:57.15CrashHDsweet
22:57.20CrashHDI've typically done all cli stuff
22:57.30CrashHDbut clients are demanding gui based now
22:57.45CrashHDany in roads to asterisk as a virtual machine?
22:57.47citywokwrite a gui in PHP lol
22:57.49CrashHDon say vmware platform?
22:57.54QwellCrashHD: the super awesome developer at Digium who you should buy a beer maintains AsteriskNOW, so you know its good.
22:57.58citywokCrashHD: i've run it in Hyper-V without much trouble
22:58.13chazzamlol
22:58.17citywokQwell: who maintains *now?
22:58.22Qwellchazzam: too subtle?
22:58.24chazzamQwell
22:58.28CrashHDQwell, I was an idler around here when asterisknow was initially released
22:58.33CrashHDwasn't much then
22:58.42citywokoh hahaha, i didn't know you were the maintaner
22:58.43CrashHDsounds like its maybe come a ways since then?
22:58.44chazzamit has come a long way, and Qwell didn't maintain it then
22:58.45Qwellwe don't talk about AsteriskNOW 1.0
22:58.50CrashHDLOL
22:58.58CrashHDstep 1 :)
22:59.10CrashHDsoo night and day, then and now?
22:59.14chazzamyeah.. we liked 1.5 and later
22:59.17Qwellyes
22:59.24chazzam1.7.1 is just all spiffy and pretty
22:59.30Qwellofficially renamed AsteriskNOW 1.0 to AsteriskTHEN
22:59.32chazzamand works
22:59.34QwellOFFICIALLY.
22:59.42chazzamheh
22:59.43CrashHDsweeeet
23:00.02CrashHDQwell, we typically put a couple different installs of asterisk on a single physical box
23:00.07CrashHDany issues with that?
23:00.20CrashHD(modifying makefile to change the binary as well as dir structure)
23:00.30Qwellwell, AsteriskNOW is a distro, so...
23:00.31citywokCrashHD: does it work well?
23:00.36CrashHDahh
23:00.42Qwellcitywok: lots of people do it.  works fine
23:00.43chazzam1.7.1 lets you choose among Asterisk {1.4, 1.6.2(, maybe 1.8)} and {no GUI, AsteriskGUI, FreePBX}
23:00.46CrashHDworks great
23:00.56CrashHDit's what I jokingly refer to as asterisk virtualization
23:00.58citywoki've never tried, so i was legitimately curious, now "shiting" :P
23:00.58CrashHD:)
23:01.01citywoknot*
23:01.18CrashHDok so it's a full distro, not just an interface
23:01.23CrashHDhmmmm
23:01.29citywokyea, centos + asterisk + gui
23:01.40CrashHDany luck with virtualizing instances of it?
23:01.41Qwellshould work fine
23:01.51chazzamworked fine in vmware player for me
23:01.59citywokyup, but obviously it will have more memory overhead than seperate asterisk installs running on a single box
23:02.02chazzamgranted that was just a test thing, and never saw a call
23:02.20CrashHDthe gui part I would assume, but what about actual real world use of it?
23:02.27citywokchazzam: i've had VM's running on laptops on wifi in vmware workstation, and the call quality was abysmal. lol
23:02.38CrashHDmore in reference to sip channels, calls, core asterisk stuff
23:02.43chazzamlol
23:03.02citywokby laptops i really mean just my one laptop. but it is a nice new laptop, lol
23:03.04CrashHDthe timer move to RTC?
23:03.19CrashHDor we timerless now with sip?
23:04.27chazzamreh? If you are asking about what I think you are asking, then if you go with 1.6.2 (I believe) a new enough kernel/glibc then you can use a kernel timing module
23:04.37chazzamotherwise you would still more than likely want DAHDI installed and loaded
23:05.05CrashHDdahdi_dummy
23:05.13Qwelldahdi_dummy is dead
23:05.18Qwellit's just part of dahdi now
23:05.22citywokthe last thing i virtualized was a month ago with 1.8 and debian-lenny
23:05.28citywokno dahdi, thank god.
23:05.31chazzamDahdi ko dahdi_dummy
23:06.06chazzamtimerfd I think it is?
23:06.32CrashHDah ok
23:07.02chazzamhttp://svn.asterisk.org/svn/asterisk/branches/1.6.2/doc/timing.txt
23:08.01CrashHDah sweet
23:08.03CrashHDsomething to ready
23:08.07CrashHD*read*
23:08.09CrashHDthank you
23:10.04CrashHDmeetme and moh still require timing?
23:10.14citywokmeetme requires dahdi
23:10.17CrashHDI see here it says only IAX2 trunking requires it
23:10.27CrashHDit requires dahdi's timing source?
23:10.31Qwellno
23:10.35CrashHDor will res_timing_timerfd due?
23:10.37Qwellit requires dahdi for mixing
23:10.38citywokmake menuconfig, and go to meetme.  it says requires dahdi.
23:10.38CrashHDd0*?
23:10.48CrashHDok
23:10.57CrashHDhmm
23:11.11CrashHDso if I install dahdi, asterisk will use it as a timing source rather than the kernels timing?
23:11.15CrashHD*per this document*
23:12.08Kobazif the kernel module is loaded, and asterisk loads chan_dahdi, yes
23:12.55CrashHDhmmmmmmm
23:13.24*** join/#asterisk Defraz (~Defraz@63.226.95.152)
23:14.32CrashHDhmm so no production references runing asterisk on vsphere (vmware) 4.0+?
23:18.57p3nguinWhat was the problem?
23:22.59*** part/#asterisk nny (~Scott@174.107.201.103)
23:34.29*** join/#asterisk IsUp (IsUp@unaffiliated/isup)
23:40.28*** join/#asterisk umay (~chris@71-212-199-100.hlrn.qwest.net)
23:41.36*** join/#asterisk f2Knight (~ben@c-71-236-165-148.hsd1.or.comcast.net)
23:42.22*** join/#asterisk ClintGoudie-Nice (~clint@smtp.callware.com)
23:42.47f2KnightQ: How do I use asterisk realtime to register to a sip server?  I want to have my test box make a sip-trunk to my existing asterisk box, (which is all config files)
23:50.33ClintGoudie-Nicelike build the sip trunk and tear it down automatically?
23:51.20ClintGoudie-NiceI think the simplest thing to do would be to build some sort of script that you could either call from a web page, or an authorized ssh key that added the appropriate info to the config files, and then reloaded the config.
23:57.54*** join/#asterisk socomm (~socomm@99-121-209-85.lightspeed.irvnca.sbcglobal.net)

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