IRC log for #asterisk on 20121022

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01:05.39dijibanybody experiences with asterisk11 & dahdi?
01:05.41dijibalso fax.
01:06.06WIMPy~ask
01:06.06infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
01:06.11dijiboi
01:06.16dijibinfobot how are you tonight?
01:06.17infobotdijib: I think you lost me on that one
01:06.17dijib:)
01:07.34dijibi guess what im trying to say is same => n,MeetMe(${EXTEN},dsFMcorx);; is crashing.
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01:08.18WIMPyWhy do you still use MeetMe?
01:08.42dijibwell then what is dahdi's application?
01:08.56dijibBridge? or something man i guess thats root cause. wrong app.
01:09.15WIMPyBut a crash is never good. You should open an issue on jira for that.
01:09.30WIMPydahdi is for some telephony hardware.
01:09.49WIMPyJust as tha name says.
01:09.54WIMPy~dahdi
01:09.54infobot[~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel (more info at http://www.asterisk.org/dahdi ) and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav
01:10.30*** join/#asterisk Orii (~user1@pool-98-111-117-48.hrbgpa.fios.verizon.net)
01:10.34dijibthen what was the bridge dependent on the dahdi drivers?
01:10.37*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
01:10.51WIMPyThat was MeetMe.
01:11.21dijibno it didnt crash sorry call failed... im so melodramatic... THIS IS NOT A PRODUCTION ENVIRONMENT. lower severity.
01:11.38WIMPyBut since Asterisk 10 ConfBridge might be a better solution.
01:11.39Oriihey has asterisk been working better on freebsd? last time i heard it was still a little iffy
01:13.02dijibthanks wimpy you saved the day. just a little extensions-fu here and i will br fixed right up
01:13.15dijibbut then i have a fax issue. but let me fix this first.
01:14.15Oriialso is there a new version of the O'reilly Asterisk book?
01:14.26WIMPy~book
01:14.26infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
01:15.30Oriithank you
01:15.51Oriianyone able to answer my question about asterisk support on freebsd?
01:16.46WIMPycan't, but unless you want to use hardware interfaces, it shouldn't make a difference.
01:18.28Oriicool
01:18.35Oriiwhat if i do?
01:19.57WIMPyThere is a dahdi-freebsd. That's all I can say.
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01:24.12dijibcore show modules
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01:28.59dijibok core show applications does not show Confbridge
01:29.46WIMPymodule load app_confbridge
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01:30.06WIMPyIt might just lack a valid config file.
01:30.11dijibunable
01:31.36dijibls -asl /usr/lib/asterisk/modules/ | grep app_confbridge.so
01:31.37dijib1752 -rwxr-xr-x  1 root     root     1792071 Oct 16 22:13 app_confbridge.so
01:32.27WIMPyTurn up verbose and debug. It might tell you why it doesn;t load.
01:33.27dijibnewp
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01:42.44dijibWIMPy: for some reason i cant unload it or load it, core show applications shows ConfBridge and im pussled
01:42.47dijibpuzzled
01:43.41dijibapp_confbridge.so              Conference Bridge Application            0
01:43.41dijib1 modules loaded
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01:45.28parasitodelsurdijib!!!!!!!
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01:46.29WIMPyHmm. Try to noload it in modules.conf, restart, turn up verbose and debug and then try to load it manually again.
01:46.39parasitodelsurdijib: how you been?
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01:47.52dijibive been alright thanks parasitodelsur is that a new handle?
01:50.32dijibafter noload
01:50.32dijibUnable to load module app_confbridge.so
01:50.33dijibCommand 'module load app_confbridge.so' failed.
01:50.51dijibsame without .so
01:50.56dijibsrry mistake
01:51.41parasitodelsurdijib: nope is a secondari
01:51.44parasitodelsuris me SeRi
01:51.52parasitodelsurI have a ghost
01:51.54parasitodelsurfixing now
01:51.55dijibyooooo hommie
01:52.02dijibcrazy it has been forever
01:52.19dijibi need to get my asterisk 11 fax working.
01:52.37parasitodelsurhahaha nice seeing you again!
01:53.03dijibdefinitly... i was just looking through my dailplan and thought of the conferences we had... ive got a confbridge issue
01:53.31parasitodelsurwhat is going on?
01:55.00SeRidijib: you still have your bridge upo?
01:56.02WIMPyNo output with debug and verbose? Do you have debung enabled in logger.conf?
01:56.39dijibno thats the problem...  i need a bridge for conerence calls and MeetMe and ConfBridge have both died in my asterisk 11-rci upgrade
01:57.39dijibthe problem is app_confbridge.so is not working. the call dies when initiated. i also cannot load reload or unload it, yet it does get loaded on a core restart
01:57.52dijibthe file exists
01:57.54WIMPyThere's an rc2.
01:57.59dijibsecurities are correct
01:58.03dijibthere is?!!?
01:58.06dijiboml
01:58.36dijibwait no i must be running rc2, installed it last week
01:58.51dijibcrap i have rc1
01:59.01dijibthats core show version
01:59.25SeRidijib: upgrade
02:00.31WIMPyMaybe I shoudl do that as well...
02:01.32dijibim upgrading now... :S
02:01.38WIMPyUgh. Tons of changes.
02:01.43dijibare their?
02:01.54dijiblots to break
02:02.22WIMPySince my last update, yes.
02:02.35WIMPyThat was a fourtnight ago.
02:02.47dijibsee that
02:03.29SeRiI am still put with 1.8.14.1 :)
02:04.52WIMPywonders why there's a substantial pause after Installing modules from addons...
02:05.30dijibi need FAXing
02:05.57SeRifaxing is also not working?
02:07.00dijibyes
02:07.09dijibhas not worked for a long while
02:07.10dijiblol
02:08.40SeRilol
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02:09.42dijibnow i have a pjproject error on install
02:10.18dijibthis:
02:10.20dijibhttp://forums.asterisk.org/viewtopic.php?p=179485
02:10.26newtonrWIMPy: you at Astricon?
02:10.44WIMPyNope. Several thousands of kilometes too far away.
02:10.45SeRiman CC is undergoing a massive DDoS
02:10.52SeRiWIMPy: lol
02:11.26newtonrWIMPy: ahhh
02:23.38SeRiuggghhh been working on rwiring my network
02:24.08SeRiI hate making cat6 cables
02:26.51dijibwhy?
02:28.45dijibmake[5]: ccar: Command not found
02:28.46dijibmake[5]: *** [../lib/libpj-x86_64-unknown-linux-gnu.a] Error 127
02:30.55SeRidijib: beacuse they are a bitch to make unlike cat5 cat6 uses this stupid inserts for RJ-45
02:31.42WIMPyDon't use them if you don't like them.
02:35.14dijibGIG-E is worth it
02:36.31WIMPyloves distcc
02:38.45dijibim not so much in love with it
02:39.13dijibthere we go... past that
02:39.33dijibhad to do a make distclean in res/pjproject
02:39.36dijibthen reconfigure
02:40.26WIMPyrc2 doesn't look good on my netbook so far.
02:40.36dijibthen reconfigure
02:40.48dijibnetbook why?
02:40.57WIMPyplay
02:41.06dijibAsterisk 11.0.0-rc2 built by root @ swissarms on a x86_64 running Linux on 2012-10-22 02:32:46 UTC
02:42.06WIMPyLooks like the pjlib stuff didn't like distcc :-(
02:42.29SeRiWIMPy: mehhhh I am just bitching
02:42.32SeRi:)
02:43.10SeRiGigE POE are expensive
02:44.26dijibok i still have the same app_confbridge.so issue
02:44.51WIMPyHmm. sip still needs websockets?
02:45.15dijib?
02:45.27WIMPyI thought that was changed.
02:45.36dijibdependency of asterisk?
02:45.57WIMPyinside Asterisk
02:46.02dijibya
02:47.14WIMPySo technically sip needs to be in extended support now since it depends websockes which are in extended support or what?
02:51.24dijibyour telling me.
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02:54.42dijibi cant for the life of me fix this.
03:02.58dijibinstalling 10.9.0
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03:39.23dijibanybody a 10.9.0 fax expert?
03:44.07josefigis there an API to handle the regular features of asterisk ? something like
03:44.19josefigall the features in an abstraction layer ?
03:48.55dijibno clue
03:49.00dijibwhat are you trying to do
03:49.56josefigdijib: an api to handle some users in the cloud
03:50.10dijibno idea
03:56.32DuPietheres AMI
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03:58.26josefigDuPie: oh Manager Interface
03:59.19DuPiethats really what you're looking for
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04:02.41josefigDuPie: yes, maybe I can extend the AMI in order to support such a several things from my customers.
04:03.54josefigthanks
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04:57.54oyugikHey guys what could cause an incoming call rule not to be obeyed
04:58.04oyugikin the case of a ring group
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05:05.23dijibCannot reserve FAX session - session limit exceeded
05:06.50kaldemaroyugik: #freepbx
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05:26.37oyugik@kaldemar I am not using freepbx
05:27.59Wiretapoyugik: what are you using then
05:28.07Wiretapasterisk doesn't have 'ring groups' or 'incoming call rules'
05:28.35oyugikI am asterisk
05:28.41oyugikand dahdi
05:28.51oyugikwith principles from asterisk gui
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05:29.15oyugikI have installed asterisk-gui to manage
05:30.35oyugiksorry for the confusion I am using asterisk gui
05:32.28oyugikthis is how my extension.conf looks like
05:32.47oyugikand this is the dump from the debugger
05:33.10oyugikhttp://pastebin.com/zrGaRpsp --> extensions.conf
05:33.45oyugikhttp://pastebin.com/P6kqB5XC --> asterisk debug dump
05:41.47ChannelZDoesn't look like any of your devices are actually working
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05:44.24oyugikeh, really
05:44.33[TK]D-FenderNone registered
05:45.01[TK]D-Fenderhelps to actually look at the status of what you think you should be calling...
05:45.01oyugikwhat could be causing thus problem?
05:45.10[TK]D-FenderNOT REGISTERED.
05:45.42oyugikwhat the extensions or the device?
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05:45.54oyugikare not registered
05:46.42ChannelZSIP/6006, SIP/6004, SIP/6010...
05:46.47[TK]D-Fender^^^
05:47.08oyugikokay let me have them up
05:48.52ChannelZwanders off to watch Homeland
05:59.14oyugik@ChannelZ now there is no activity on the debug mode
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06:07.33kaldemaroyugik: what debug mode?
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06:08.58oyugikwhen I check the logs at the terminal from #asterisk -r
06:09.34[TK]D-Fender"sip set debug on"
06:09.47oyugikok
06:09.49[TK]D-FenderIf you aren't looking with SIP DEBUG enabled then you aren't really looking...
06:09.57[TK]D-FenderSame goes for any in-depth call debugging
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06:10.26oyugikdone
06:10.33kaldemarif "asterisk -r" is all you're doing, you're not even seeing verbosity. "core set verbose 10"
06:10.45oyugikok
06:11.06oyugikVerbosity was 3 and is now 10
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06:14.04oyugikthe thing that amazes me is that sometimes the calls come in and then they stop mysteriously
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06:46.57mirela666oyugik: bast thing is to sniff traffic and try to catch that situation, and when it happens you can see who sends hangup and why
06:49.32ChannelZI think he meant the calls stop coming
06:50.01ChannelZBut who knows really, with as little information as we're getting
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07:02.29dijibsoup ChannelZ?
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07:08.34mirela666yep lol
07:10.40ChannelZmmmmm soup
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08:40.33oquidaveHello, i've properly installed configured odbc with asterisk and now am now am at the point of configuring res_odbc.conf to allow asterisk to connect to mysql through odbc. i've configure the res_odbc.conf file but when i odbc show from the asterisk CLI, nothing shows...here's my config file and other info. thanks
08:40.38oquidavehttp://pastebin.com/C93SKtX8
08:41.50oquidavehowever,my odbc.ini is not in /etc/odbc.ini but in /etc/unixODBC/odbc.ini could this be the problem?
08:42.00wdoekesoquidave: first make sure isql works
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08:44.18oquidavewdoekes: it does check out http://pastebin.com/PG37B5eP
08:45.46wdoekescore set verbose 20
08:45.49wdoekesmodule reload res_odbc.so
08:48.18oquidavewdoekes: it says res_odbc.c:820 load_odbc_config: Limit should be a number, not a boolean: '0'.  Disabling ODBC class 'asterisk' ...i've checked the limit and it's 0 ...limit => 0 ...so to what figure should i change it?
08:49.03wdoekessee configs/res_odbc.conf for an explanation what limit does
08:49.19oquidavewdoekes: seen it...am changing it to 5
08:50.27oquidavewdoekes: thanks it's now up and running!!!!
08:50.54oquidavewdoekes: " core set verbose 20" why does it have to be 20?
08:51.27wdoekesbecause then you get *all* verbose messages
08:51.36wdoekesbut you don't need it unless you're debugging something
08:51.50oquidavewdoekes: so i can change it back to 3
08:52.01wdoekes(I think 11 is the highest number used, so 20 is safe)
08:52.03wdoekessure
08:52.21oquidavewdoekes: okay thanks
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09:02.06ChannelZI want to say last I looked 5 or 6 was the max ever used
09:04.38oquidaveChannelZ: okay, i'lll go with 5. thanks
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09:05.50ChannelZIt doesn't hurt or anything, just that it's not limitless.  I run normally at 3 which pretty much tells me everything I've ever needed to know
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09:48.01awkHi guys, currently my system writes cdr's to flat file and mysql db. and the only option I can see under cdr.conf is enable = yes / no... Now how can I have it so it doesn't write the flat file but keeps writing to the DB?
09:48.03awkMany thanks!
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09:51.41wdoekesawk: ; To get CDRs to be logged to the plain-jane /var/log/asterisk/cdr-csv/Master.csv
09:51.44wdoekes; file, define the [csv] category in this file. No database necessary. The example
09:51.47wdoekes; config files are set up to provide this kind of output by default.
09:52.03wdoekesthe inverse is most likely true too
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10:03.46awkI found what I had to disable just cdr_custom module :)
10:06.19oyugikhey guys I found a solution for my problems earler
10:07.05oyugiki was using a context "from-pstn" that wasnt explicitly defined in my extensions.conf
10:07.19oyugikhence causing difficulties in the hang up event
10:08.11oyugikhence channels got confused
10:08.22oyugikin the execution of dialplan
10:08.43oyugikI probably should blog about it. But what the heck
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10:23.29oquidavehello guys am just realling on asterisk realtime now and i've discovered there's a db schema/structure that you've to follow which i've read. However, to make my development faster, are there some .sql files with this schema that i can just import into my mysql db and simply fill in the values? thanks
10:25.58oquidavesomething similar to the sample files for instance
10:28.42wdoekesoquidave: contrib/realtime
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10:32.05oquidavewdoekes: exactly what i need! thanks again...now i can just import the sql file
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10:55.06nappyhello everyone...
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10:56.52nappywould like to ask if there is a solution or somebody has implement a web conference in asterisk?
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10:59.08oquidaveasterisk realtime is really cool. now i can build a web app where users can register into the system so easily without tinkering with the sip.conf file.
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11:07.22nappyim using trixbox... is there any application/module for that??
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11:07.50nappyhi oquidave,,.. is that for web conferencing?
11:08.44nunneOn a "powerfull" system (2x6core xenon, 8-16gb ram) how may registrations (pref. with qualify) can asterisk 10 handle? or is it still recommended that openser handles the regs?
11:11.37kaldemarnunne: http://www.voip-info.org/wiki/view/Asterisk+dimensioning
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11:35.33gavimobilehow can I view the exact caller id that the destination caller sees?
11:36.21WIMPyPlace a web cam above hos phone.
11:36.25WIMPyhis
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11:37.54WIMPyWhat exactely do you want to know?
11:38.34nappyWIMPy: lol
11:38.35gavimobileWIMPy: my diagnosis is unclear as of now, however "some" landline destination parties see my telephone number as an international number
11:39.02gavimobileI used the word some cause only some complain, but I believe that other landlines using the same telco see the number as it should
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11:39.15WIMPyWhat do you send and what kind of connection?
11:39.49gavimobileI don't understand the question
11:39.53gavimobileI am using sip
11:40.19gavimobileand it only happens with this dialplan exten => _0NNXXXXXX,1,Set(CALLERID(num)=${OUTCID2})
11:40.20gavimobile<PROTECTED>
11:40.35gavimobile"sometimes"
11:40.36fredericveOn ubuntu lucid, while building asterisk 11 I get this error:
11:40.37fredericvei486-linux-gnu-ar: Command not found
11:40.44oquidavenappy: no, still just playing with asterisk....
11:40.57fredericveanyone knows how to fix this?
11:41.06WIMPyWell, it might be them using different routes then.
11:41.09nappyohhh okay.. i thought its used with conferencing
11:41.28gavimobileWIMPy: I have a log with sip debug when calling a destination party which sees an international number
11:41.39BorjaGVOHi everyone...I've got some weird thing happening here. Asterisk seems not finding a file (vm-youhave) although it's there, under /var/lib/asterisk/sounds/es/vm-youhave.gsm, Asterisk is not able to open it. I even did chmod 777 to the file and no sucess in case it could be permission issue. Anyone can help me with this?
11:41.47WIMPyfredericve: Does it have some pseudo package like build-essentials?
11:41.49gavimobileWIMPy: if it uses different routes, does this mean its not in my control? it never happenes when I call cellphones
11:42.30fredericveWIMPy: yes build-essential is installed
11:42.46gavimobileBorjaGVO: in your dialplan, try to leave the extention out
11:42.54oquidaveBorjaGVO: make sure the file extension is not included
11:43.01WIMPygavimobile: Possibly. Maybe you can use another format that won't show the issue. But you should check with your ITSP what they expect.
11:43.25gavimobileWIMPy: what do you mean by other format? im unclear
11:43.36gavimobilebut thanks, I will speak with my telco provider.
11:43.39BorjaGVOgavimobile, oquidave: I'm using FreePBX so I don't know if modifying the dialplan is a good idea
11:43.46BorjaGVOI mean....it worked before
11:43.48oquidaveBorjaGVO: plus the file path should be correct e.g Playback(en/vm-youhave)
11:43.54WIMPygavimobile: Like you sending your number in national or international format.
11:44.08gavimobileBorjaGVO: this is a channel for asterisk, not freepbx
11:44.32BorjaGVOWell, it's directly connected to asterisk...
11:44.39gavimobileWIMPy: well the weird thing is sometimes it works and sometimes it doesn't
11:44.54WIMPyBorjaGVO: Try #freepbx
11:44.58gavimobileBorjaGVO: you just said you don't know how to modify the dialplan
11:44.59BorjaGVOalright
11:45.04BorjaGVOyes
11:45.13gavimobileif you don't know how to modify the dialplan how are the folks in the channel suppose to help you?
11:45.21BorjaGVOI know how to..but I don't suppose to have to...
11:45.30WIMPygavimobile: Unless you do it differently, it must happen somewhere else.
11:45.35BorjaGVOanyway, I'll try at #freepbx
11:45.37BorjaGVOthank you
11:45.50gavimobileBorjaGVO: for freepbx support you should speak to them in their channel
11:45.56gavimobilesorry :-p
11:46.21gavimobileWIMPy: this is why I want to monitor it.. in my cdr it doesn't show the caller id the party sees
11:46.22BorjaGVOgavimobile: np
11:46.28gavimobilethis is why I asked my initial question
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11:47.15gavimobileif I can monitor what they see (without a webcam in front of their phone :-p) than I can troubleshoot when and where it happenes
11:47.37WIMPygavimobile: If you explicetely set the callerID before dialing out, try to put a verbose in between. If you're not 100% sure it might do something.
11:47.38gavimobileWIMPy: unless you have another idea to approach the issue
11:48.13WIMPygavimobile: Try to find the pattern when it works as expected and when not.
11:48.22gavimobilehrm...
11:48.31gavimobileim just thinking what I would put inside verbose()
11:48.38gavimobilecallerid?
11:48.43gavimobile${callerid}
11:48.51WIMPyCALLERID(num)
11:48.53gavimobilesomething like that
11:48.57gavimobilegot it
11:49.14gavimobileWIMPy: ill go ahead and do that now
11:51.31gavimobileWIMPy: is this kosher? http://pastebin.com/ziUfFHkk
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11:56.19fredericveWIMPy: I fixed the error by symlinking /usr/bin/ar to /usr/bin/i486-linux-gnu-ar
11:57.47WIMPygavimobile: yes
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11:59.52gavimobileWIMPy: still no go
12:00.32WIMPyMeaning?
12:00.36gavimobilethe verbose works, and it shows which number is beind set by callerid
12:00.53gavimobileso I tried changing it to local so it wouldn't be international, but the call is still showing up as international
12:01.06gavimobileill send a pastebin to help you understand more what im refering to
12:02.06gavimobilehttp://pastebin.com/xxxz7ZHK
12:03.12gavimobilethe caller id which was set in example 2 is the caller id I want them to see. my trunk provider is from the states which is why I set the caller id the way it was in example 1.
12:03.18WIMPySo you have an idea, who to blame then.
12:03.28gavimobilemy trunk provider?
12:03.54WIMPyBut the two calls look the same.
12:04.20WIMPyOh, you changed tit after the verbose.
12:04.25WIMPy-t
12:04.56WIMPyYes, your provider or some other place that's outside of your control.
12:05.00gavimobileWIMPy: I lost you.. I thought the point to add verbose was to see in the output what number was being used in caller id
12:05.21gavimobileWIMPy: if its not in my providers control, than who else can I speak to
12:05.32WIMPyI see the difference in the voerbose output of the set.
12:06.01WIMPyNoone. Talk to your provider. If it's not them, they will have to find out who it is.
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12:13.23gavimobileWIMPy: thanks so much!
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14:03.50devdvdHi all, I'm looking at setting up an * box to do 200 users, voicemail, web interface, 50 concurrent calls and g729 (assume 50 simultanious conversions).  I realize this isn't a high amount for * but I was wondering if anyone here has experience with that many calls (using g729) and what kind of hardware you use.
14:04.07devdvdprobably do conferencing as well
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14:05.03oquidavehello,am trying query a mysql db from within asterisk dialplan using odbc...http://pastebin.com/5XE62SbK ...however am finding it hard to debug where the problem could be. Could someone give me some leads. ive pastebin has enough info of what i've already done. thanks
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14:07.13[TK]D-Fenderoquidave, Looks null... seems fine to me...
14:08.08cuscohello
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14:08.27cuscoI was reading up on http://forums.asterisk.org/viewtopic.php?p=170058 but im not sure that my issue is the timming source
14:08.38cuscobasically we have 2 PRI cards 4 span each
14:08.40cresl1ndevdvd: Just buy a modern, moderately spec'd server
14:08.55cuscoin one of them we have 2 PRI from telco1 and 1 PRI from telco2
14:09.20[TK]D-Fenderdevdvd, Your users are expected to be remote from your server?
14:09.22cuscoand what happens is that in telco1 some calls there is no audio from client calling in...
14:09.38devdvdTK, yes
14:10.13kaldemaroquidave: ${EXTEN} has 751 in it. your DB only has a row with "50" as exten.
14:10.17[TK]D-FenderdevAnd basic box from today should be more than enough.  I'd personally aim for an i5+
14:10.27[TK]D-Fenderoquidave, As kal said....
14:10.37[TK]D-Fenderkaldemar*
14:10.48cuscolooking at dmesg I have lots of lines like: http://paste.debian.net/202642/
14:11.02cuscocan I not use telco's timing source and use a internal timing source instead?
14:11.25cuscoI'm using: dahdi-linux-complete-2.5.0+2.5.0
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14:12.13oquidavekaldemar:  sorry, i think i get it now...i put 50 because am calling from extension 50...so the idea was that it takes the user extension as an argument!! thanks
14:12.44oquidavekaldemar: so i'll have to change it to caller id instead
14:12.54oquidavekaldemar: thanks again
14:12.59kaldemaroquidave: sounds like you need to give a thought or two on what an extension is in asterisk.
14:13.59jeffspeffok, i'm missing something here. I'm trying to setup a secondary voicemail system. I'm using realtime configuration. when in asterisk console and I do a 'voicemail show users for PW' it says that the command failed. What could i be doing wrong?
14:14.24jeffspeffi'm using mysql for realtime driver. mysql is installed on the same localhost
14:14.30oquidavekaldemar: extensions are like code-lines in dialplan though they can also be used "phone numbers" for the user. I keep missing it up!!
14:14.47devdvdcuts jeffspeff
14:14.59jeffspeff?
14:15.06devdvd<twister
14:15.20jeffspeffoh
14:15.22devdvdlol
14:15.31jeffspeffany ideas on this?
14:16.24devdvdwhen you do voicemail show users for ?
14:16.28devdvddoes it show up in that list
14:16.31jeffspeffi've double checked the config files against my production system which also uses realtime for voicemail; the only difference is on the production system the voicemail was originally setup through flat file, then converted to realtime
14:16.59jeffspeffno
14:17.21jeffspeffi've got a sneaky suspicion that it's not actually parsing the database
14:17.36devdvdright, do you have anything else realtime on that box?
14:17.38devdvdor just voice mail
14:18.00jeffspeffvoicemail is the only thing setup
14:18.42devdvdwhat does realtime mysql status tell ya
14:19.07devdvdive never tried to setup voicemail in realtime, so im just stabbin at it here
14:19.22jeffspeffit shows that the db is setup
14:19.55devdvdwhats your extconfig entry look like for that?
14:20.02devdvdthe voicemail i mean
14:20.21jeffspeffvoicemail => mysql,cisco,voicemail
14:21.42jeffspeffthen in res_config_mysql.conf i specify:  dbhost = 127.0.0.1 dbname = cisco dbuser = root dbpass = password dbport = 3306 dbsock = /va/rlib/mysql/mysql.sock requirements=warn
14:21.43devdvdoh
14:21.49jeffspeffobviously all on seperate lines
14:21.55devdvdtry this
14:22.17devdvdmove voicemail.conf to some other directory (like /home/asterisk or whatever)
14:22.22devdvdthen just touch voicemail.conf
14:22.30devdvdthen change ownership to the asterisk user
14:22.37devdvdmake the voicemail.conf blank
14:22.48devdvdthen restart *
14:23.24jeffspeffok, done and same results
14:23.31devdvdok
14:23.44devdvdhold let me try to set realtime up on my side and we'll work through it
14:23.51jeffspeffok
14:24.02kaldemarjeffspeff: is "command failed" all it says?
14:24.24cuscocan I dial a sip peer, but on dialplan before dialing set the fromuser SIP option ??
14:24.38cuscooverriding whatever is set in sip.conf ?
14:24.43jeffspefflocalhost*CLI> voicemail show users for PW
14:24.44jeffspeffCommand 'voicemail show users for PW' failed.
14:25.38jeffspeffcusco, no users being set in sip.conf. only inbound sip user from other system
14:27.28cusco?=
14:27.58cuscoI mean.. in sip.conf I have a peer from a remote sip gateway. I dial say SIP/peer/xxxxxxxxx ...
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14:28.23devdvdjeff, he wasn't responding to you
14:28.30jeffspeffoh
14:28.35jeffspeff:p
14:28.38cuscobut I would like to be able to set the sip header "From: <sip:FROMUSER@...>" dinamically
14:28.50cuscoin dialplan
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14:29.31cuscojeffspeff: reading on your issue do you have any vm module loaded in asterisk? also vm in extconfig set to use a database?
14:29.52jeffspeffyes, vm module is loaded, and yes set to use correct db
14:30.27cuscomodule show like vm; module show like voicemail
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14:39.25devdvdyea not working for me either
14:40.28cuscohow do I increment a variable on ael? Set(a=$[${a} + 1]) ?
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14:45.30jeffspeffdevdvd, i fixed it
14:45.36devdvdah ok cool
14:45.39devdvdwhat'd ya do
14:46.09jeffspeffwell in res_config_mysql.conf dpbass != dbpass
14:46.12jeffspeffsimple type
14:46.14jeffspeff*typo
14:46.23devdvdinteresting
14:46.30jeffspeffi knew i was missing something rediculously simple
14:46.45devdvdbut it showed up as connecting fine with realtime mysql status
14:46.54jeffspeffyep
14:47.36devdvdinteresting
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15:03.07wtfitsmeHello, I have an issue where a carrier is sending me media (RTP) on a port that is not in the range I specified in rtp.conf - This is happening on a 1.4.23 box. Shouldn't Asterisk negotiate this in the SDP body?
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15:14.03BorjaGVONew to Asterisk source code...I'm going over file.c, and I don't get this function: struct ast_filestream *ast_openstream(struct ast_channel *chan, char *filename, char *preflang)
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15:14.57BorjaGVOfilename2 is set t0 null: char filename2[256]=""; right? where is the filename assigned?
15:15.07danfromukHas anyone had problems integrating the new version of XLite into Asterisk? Specifically, the way that XLite forces a specific dialplan?
15:15.12BorjaGVOI'm not a programmer so I might be missing something...sorry.
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15:16.39jpsharpThat's not being set to null, that's being set to an empty string.  Big difference.
15:16.59BorjaGVOwell...sorry
15:17.10BorjaGVOit is a big difference...right. Empty string...
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15:18.14BorjaGVOsource code: http://doxygen.asterisk.org/asterisk1.0/file_8c-source.html
15:18.25Qwellso what are you asking?
15:18.46[TK]D-Fenderdanfromuk, Thre is no such "integration".  It is a dumb softphone.
15:20.14danfromuk[TK]D-Fender: yes however the new version doesnt seem to be compatible with my dial plans, and even forces incorrect dialling. They've tried to make the dialling plan simpler, but in fact broken it.
15:20.52[TK]D-Fendershow us where you even set it...
15:21.11danfromuk[TK]D-Fender: for example, the user dials 9 & local dial code & number, and xlite sends +countrycode & 9 & local dial code + number.
15:21.31danfromukThe 9 is messing everything up and I can't seem to remove it. I was wondering if anyones seen that with the latest version
15:22.42danfromukhttp://imagebin.org/232838
15:24.22[TK]D-Fenderdanfromuk, That looks like trash.  I'd just strip it out....
15:24.36danfromukCan't. It insists on having numbers.
15:24.50danfromukI think its a bug on counterpath's part
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15:32.46danfromukOk, got round it by adding x before each number dialled. That makes xlite think that i'm dialling a local extension and doesnt add any digits automatically
15:34.58jpsharpThat's awfully presumptious of xlite.
15:35.53danfromukBit annoying that xlite now comes with an advert/integration with softphone.com. Are there any alternatives that I can offer my clients? How easy is it to make one?
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15:39.15fullstopitsp?
15:39.47[TK]D-Fenderdanfromuk, Yes there is an alternative... their COMMERCIAL products
15:39.58[TK]D-FenderX-Lite is for chumps
15:40.27fullstopI don't mean to intrude, but I'm considering an itsp again and I was wondering if there was an updated list.
15:40.27danfromukAgreed however, I'm a pretty good programmer and therefore hate buying products that I know I could make.
15:40.31fullstopvoip-info is kind of strange now
15:42.19fullstopinfobot: itsp?
15:42.19infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.
15:43.25BorjaGVOno one can help me with the code? Come one! ;)
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15:49.47navaismowhat code BorjaGVO and what are you trying to do?
15:50.25BorjaGVOthis code: http://doxygen.asterisk.org/asterisk1.0/file_8c-source.html
15:50.31devdvdBorjaGOV, try asking in #asterisk-dev
15:50.45BorjaGVOI'm trying to see where filename2 is assigned..
15:51.04QwellBorjaGVO: in the snprintf() calls
15:51.15BorjaGVOI mean, I just see an empty string (in function struct ast_filestream *ast_openstream(struct ast_channel *chan, char *filename, char *preflang)
15:52.10jmls1ConfBridge([confno][,bridge_profile[,user_profile[,menu]]])
15:52.23jmls1doesn't this mean that all options are optional ?
15:53.00BorjaGVO@Qwell: alright,..didn't know that function...though it was obvious...
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16:58.52edveHi people, i've got a question for all of you that could help me with Fail2ban with asteriskNow. Actually i've install the software and it seems it does'nt work correctly ..
16:59.03d_preston215Is there a priority difference between a call generated via AGI and a call generated from a call file?
16:59.37d_preston215I have a weird issue where it seems calls generated from a call file have a lower priority over other calls.
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17:00.32JerJerd_preston215:  define priority
17:05.37[TK]D-Fenderd_preston215, AGi does NOT generate "calls"
17:06.01jpsharpd_preston215: Well, you have to remember that call files are generated when asterisk gets around to scanning the spool directory, whereas a call generated through something like AMI or the originate command is done as soon as the command is entered.
17:06.32jpsharps/call files/calls originated by call files/
17:06.54jpsharpinfobot: Thank you
17:06.54infobotjpsharp: de nada
17:09.43jeffspeffhas anybody tried to do dial-up modem connection through a sip provder using a sip to pstn adapater?
17:09.47jeffspeff*adapter
17:09.50jpsharpAnd there is a 1 second pause at the beginning of every loop of the spool scan.
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17:10.34jpsharpjeffspeff: masochistic much?  I've heard of people doing it and failing miserably at it.
17:11.04jpsharpOr rather, trying to do it.
17:11.21jeffspeffsomebody was asking about changing their phones over to asterisk but, their business requires them to ocassionaly do a modem connection to medicare systems.
17:11.39WIMPyjeffspeff: Kind of. I tried a X.75 connection on my NGN line. It actually worked without issue apart from a horrible low speed.
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17:11.59jeffspeffit didn't sound like a good or stable idea to me, but didn't know how badly it would fail
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17:12.14jpsharpjeffspeff: Save yourself the heartburn and heartache and tell them to keep a POTS line for the modem.
17:12.22JerJer9600 baud, if you are lucky
17:12.34JerJerbut its not going to be reliable
17:12.47jeffspeffok, thanks for confirming
17:13.34WIMPyBetter tell them to exchange whatever they use for something that uses IP itself.
17:13.56jpsharpWIMPy: That would require the US government to come out of the technological dark ages.
17:14.32JerJerwe've gotten the stamp machines to work by disabling all data compression and error correction  (you will have to find the correct AT command string for your modem)
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17:17.14saxahi everybody, its a long time since i lastly was here :)
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18:04.45Micc_So are there any good pre astricon things to do today?
18:05.39WIMPyGet loaded?
18:06.04edveAnyone here knows about fail2ban ?
18:08.36WIMPyIt's said it can be used to automatically shoot yourself in the foot.
18:08.49jmetrohas anyone heard of transferring somehow breaking so that the transferred call goes to an outside number? somehow transferring to a 2 digit extension is hitting outside.
18:09.28WIMPyLooks like you have the wrong context somewhere.
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18:17.44jmetroHm. Anything else you can think of besides the context?
18:18.35WIMPyno
18:18.35dijibanybody got a fax machine they can send me a test fax from?
18:20.39WIMPyIf you want it via PSTN you should tell us, what country.
18:20.50drzedhi there, i'm connected to a PSTN via IO
18:21.03drzedhi there, i'm connected to a PSTN via ISDN, but foreign calls do not work
18:21.08[TK]D-Fenderjmetro, Show us it actually happening...
18:21.14dijibdoesnt matter. canada
18:21.30drzedthe error msg is: Channel 0/1, span 1 got hangup request, cause 3
18:21.34WIMPydrzed: define "doesn't work"
18:21.55dijibtoll-free canada
18:22.17drzedi can dial national numbers, but internationals have this problem
18:22.22WIMPydrzed: Are you allowed to call abroad?
18:22.30drzedyes, i am
18:22.42WIMPydijib: Probably not toll free from abroad.
18:22.49drzedat least it was working perfectly fine with my old telephone system
18:23.14drzedi thoght it could have to do something with the foreign double zero perfix
18:23.16dijibthen i have another number
18:23.25drzedbut removeing them does not help
18:23.35dijibsorry im mixing my issue ith drzed's
18:24.05WIMPydrzed: "No route to destination" is not a common one. Have you tried different destinations?
18:24.15saxahey WIMPy :)
18:25.22WIMPydrzed: Does the cause come from your telco or farther away?
18:26.03drzedWIMPy: how can i find outß
18:27.05WIMPydrzed: You will see the location if you enable pri debug.
18:29.34WIMPydrzed: Have you tried different destinations?
18:29.46drzedunfortunately my version does not support libpri
18:29.50Micc_Anyone have any experience playing with asterisk SCF? Is it getting good yet?
18:29.56drzedyes i tries three diffrent countries
18:30.21[TK]D-FenderMicc_,  It's dead.
18:30.58WIMPydrzed: You said you're connected via ISDN
18:31.20drzedyess, indeed i am
18:31.29[TK]D-FenderMicc_, http://blogs.digium.com/2012/09/14/asterisk-scf-pause/
18:31.39WIMPydrzed: So how are you connected?
18:32.00drzedusing an isdn card
18:32.12WIMPymisdn? lcr?
18:32.13drzedon the pci slot
18:32.16drzedzaptel
18:32.36WIMPyThe you are using libpri.
18:33.14drzedups, you are right
18:33.17WIMPyAnd zaptel has been replaced by dahdi several years ago, BTW.
18:33.49drzedWIMPy: http://nopaste.info/338996f93b.html <-- debug log
18:34.48cuscoIm reading on again about dahdi timing sources... telco provides us the service, so we should use its timing source, right?
18:34.51WIMPydrzed: Yes, you're setting the type of number to national. That doesn't work for international, obviousely.
18:34.55WIMPyAnd not for local, either.
18:35.05cuscocould I use a internal timing source? such as timerfd ?
18:35.16coppice[TK]D-Fender: was there ever any significant work on SCF?
18:35.34WIMPydrzed: See the last one on http://voice.yeti.dk/Asterisk_vs_ISDN/5
18:36.16[TK]D-Fendercoppice, Never really looked at it myself.  I figure it's all vaporware until a real beta test or RC anyway, and in this case proven to be a worthwhile metric for how much I should care...
18:37.08coppice[TK]D-Fender: I looked a couple of times over the last couple of years and it mostly seemed to be hot air
18:38.24[TK]D-Fendercoppice, You already know that I hold your opinions in high regard.  I suspect that their methodologies proved to be more stop-gap than anything revolutionary and the code just swam upstream to Asterisk itself
18:38.54[TK]D-Fendercoppice, Diminishing returns plus probable challenges in marketability
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18:40.27drzedWIMPy: great, thx very much!
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18:44.19WIMPyAdded a litte explanation.
18:45.44fr0ggieHello all! I've got asterisk setup with a Android SIPUA (remote, behind NAT) over UDP connecting. Asterisk make calls via google talk (gv) and i dont hear echo on my end.. However, the other end hears themself echoing pretty bad. Anything I can do?
18:45.55drzedWIMPy: perfect, thx again
18:46.05WIMPyIs that correct terminology? Do you call the prefix to place national calls (if required) a "trunk prefix" or is there a better term?
18:46.18drmessanofr0ggie:  What is the other end using?
18:46.25fr0ggiedrmessano: PSTN
18:46.39cuscohow would I tell dahdi to use res_timing_timerfd.so as a timing source?
18:47.19fr0ggieAndroid phone (builtin UA) -[SIP]-> asterisk -[gtalk/jabber]-> google voice -[PSTN]-> Other end
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18:49.20drmessanofr0ggie:  I suppose its possible your endpoint is looping the speaker audio back down the line.. Are you on speakerphone?
18:50.01fr0ggienope
18:50.27fr0ggiespeakerphone annoyingly enough doesnt work in android's builtin sipua
18:51.22drmessanoTried another client?
18:51.54fr0ggieNot yet, wasnt sure if could be my client since the other end is only one that hears any echo at all
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18:52.57drmessanoIt's not Asterisk.. so it's either your client or something on the PSTN end of the google voice connection
18:53.32fr0ggiedrmessano: Not what i was hoping to hear :) I was hoping for a config edit and it work, ill try different client, back soon
18:53.39drmessanook
18:53.51fr0ggiedrmessano: any recommendations for on android?
18:54.01drmessanoNo, I hate Android :)
18:54.55fr0ggiedrmessano: me too, but there isnt so much choice until  i cobble a working linux onto my phone and free it from the clutches of google >:)
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18:57.01drmessanoI like iOS.. it just works.  No need to tinker or guess or wonder if your particular android implementation is borked because your provider decided to tweak it
18:57.50drmessanoI spend all day dealing with crappy android devices that stop sending email, stop receiving email, start authenticating using old credentials, and other mysterious crap.  Cant stand it
18:57.53fr0ggiedrmessano: nah, i run cyanogen, my provider sucks
18:58.23drmessanoBesides that, it's Java based.  Nothing that I hate more than Java
18:58.27fr0ggieProblem i have with iOS devices is the fact that im stuck running whatever OS they decide i should run, its like i dont really own the hardware merely paid a bunch to lease it
18:58.36fr0ggiedrmessano: Amen. java is like sewage
18:58.44fr0ggieWhen i see it or smell it, i vomit all over
18:58.53cuscodrmessano: whenever I try to explain that to people I tend to get bashed, so I just stay quiet
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18:59.44fr0ggiei only use android because $50 for android phone beats $500 for ipod phone (besides, i already have ipod)
18:59.49cuscoand... jailbreak your ios, and you can controll your hardware
18:59.56drmessanoI like the fact that my IOS devices work.  Like, when I want to make a call, or send an email, or need to VNC/RDP/SSH into a machine while I am heading down the highway 4 hours from home, I know it works.  I wouldn't last a week praying over an Android device to comply
19:00.37fr0ggiecusco: can i port my kernel to it? sadly not, apple has cryptographic controls that force you into iOS
19:00.53drmessano"It's fine.. if it starts acting up, I just reboot it"  "Oh, and what happens if it fucks up 4 times in one day?"  "I just reboot it 4 times, duh"
19:00.55drmessanoYeah....
19:00.56fr0ggiemy ipod, it is jailbroken 4.3.?
19:01.11cuscofr0ggie: I'm sure you can, there is even a project on running android on ios.. once the jailbreak hacks the boot firmware
19:01.12fr0ggiei definitely dont like android. Not one bit
19:01.45fr0ggieuntil obama gives me my change back, i cant afford to buy more shiney things :(
19:01.58cuscoI agree on that part
19:02.16saxahttp://pastebin.com/j35p5NUr
19:02.21drmessanoI agree completely there.. if you want hardware freedom, and IOS device won't give you that.  I prefer IOS.. The hardware is pretty and all, and works.. but the OS is what sells me on Apple devices.
19:02.29saxaany idea why i'm getting that error ?
19:03.33drzedfr0ggie: ++
19:03.35fr0ggiedrmessano: i like iOS, i even sorta like the shiney hardware it runs on.. but i feel like i am indefinitely renting instead of buying.. like.. if you piss apple off, no more SHSH signing == no more restore/updates (unless of course you save them, and you risk getting stuck without apps then, since no app store)
19:04.50navaismosaxa, routing issues
19:05.04drmessanoI save all my hardware freedom for my PCs and Laptops.  My mobile devices are too important to not think about feature set and stability
19:05.48fr0ggiesaxa: I may be wrong here, but i had that problem when i was using tcp mode sip (trying to avoid a braindamaged NAT firewall), i switched to udp, enabled nat=comedia and updated firmware on router == fixed. Probably simpler fix tho that these guys will know :)
19:05.54drmessanoIf Ubuntu pitches a fit and I need to hammer out a fix, I can tolerate that.  I don't need 100% uptime for my desktop or laptop
19:05.59fr0ggiedrmessano: I like to play with my hobby OS when i get bored
19:06.36fr0ggieeven a pure linux on phone, i would find that fantastic
19:06.50drmessanoI want an Ubuntu phone one of these days
19:07.07fr0ggie<PROTECTED>
19:07.17drmessanoYES I LOVE UNITY DESKTOP NOW STOP YELLING AT ME AND RINGING MY DOORBELL AND RUNNING OFF YOU DAMN KIDS
19:07.53fr0ggiei dont think i'd be very happy with ubuntu on a phone, mostly because phones tend to be ram starved and more CPU usage == more heat and less battery life. Battery life is why i want to piss from a helicopter onto google hq >:)
19:08.54fr0ggieif i boot my phone into my nice happy little linux install, i can use finch and irssi all day long over 3g, minus the whole having to carry a keyboard with me
19:09.07fr0ggieIf i run android, after 6-8 hours, i have to swap batteries
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19:10.01drmessanoThey've done a lot of work with power management on Ubuntu in preparation for mobile devices, specifically phones.
19:10.08drmessanoI would love to see some benchmarks
19:10.27puzzledevening all
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19:11.26drmessanoShuttleworth can see the writing on the wall.  I am confident they will be able to do with Ubuntu what M$ wont be able to do with Windows 8
19:12.12chuckfif they don't piss off all the unpaid contributiors before that happens
19:12.21drmessanoWant to talk about scary?  BSOD OS on my mobile device
19:12.31drmessanochuckf +1
19:12.53fr0ggiei have no problem with trying to be cluebie friendly, hell its great getting more people to come to the shiney, tasty penguin meat... but i dont so much care for ubuntu's lets make it shiney n blingy til it needs 1gb video ram, 16gb of ram, and a 16384 core CPU to even move the mouse :(
19:13.31fr0ggietaking over 15 seconds to boot to a login manager? Unacceptable
19:14.19drmessanoHardly.. I have Ubuntu running on some pretty crappy machines and they hum right along
19:14.38saxanavaismo: ok, thx, but i do not remember that i changed anything
19:14.44fr0ggiemy netbook, i hit power button and within 15 seconds, im sitting at the login screen, 3 seconds after i login, i've got a desktop waiting for me.
19:15.07fr0ggieIts more acceptable for 30-45 seconds on a desktop/laptop, but a phone? I should turn it on and within 5 seconds be able to use it
19:15.17saxafr0ggie: thx
19:15.32navaismosaxa,  is that a remote extension?
19:15.39WIMPySame is true for desktop VOIP phones
19:15.43saxayes the one behind 2 nats
19:15.48drmessanoUm..
19:16.07saxai mean one nat at home and the other side has an * server behind a nat
19:16.33drmessanoWhile unity may be the basis for the UI, the boot and login processes will be very different on a production Ubuntu device.  I dont expect you'll have to boot and login to make a call :)
19:17.23fr0ggiedrmessano: correct-- it should boot immediately on receiving power to a fully booted state, with your lock screen sitting active (background init basically)
19:17.50navaismosaxa, did you have configured the nat settings on your asterisk box? externhost or externip, localnet, and nat=yes in the peer
19:17.52drmessanoIt takes as long for Ubuntu to boot on my laptop as it does to cold boot my ipad 2.  15 seconds or so
19:18.35saxanavaismo: yes, all is like it was
19:18.55fr0ggiehow fast laptop? my netbook is 1.6ghz single core, 2gb ram, 16gb ssd
19:19.02saxanavaismo: I just need to recheck my * side modem firewall
19:19.24saxasince on the home side i know I have not touched anything at all
19:19.42saxathx for now
19:19.54saxatomorrow in my office i check that thing
19:20.33drmessanoDual core 2.53GHZ, 3GB RAM, 200GB non-SSD
19:20.37drmessanoI think
19:23.13fr0ggiedual core helps IMMENSELY with startup times
19:23.49drmessanoDual 2.16 actually
19:24.28drmessanoWith an encrypted home and swap it takes a little closer to 45 seconds
19:24.40drmessanoBut I am paranoid
19:25.17fr0ggieunderstandable
19:25.22fr0ggiepeople steal shit too often
19:25.32fr0ggiealso, feds like to steal shit
19:25.44jmetro2.5gh 6 core, 8gb ram, 2x500gb sata
19:25.52fr0ggielaptop?
19:25.57jmetropc =D
19:26.03fr0ggiewhat is pc/
19:26.21jmetroTower. Big tower.
19:26.30fr0ggieOhh, space heater
19:26.42fr0ggiei like my lap warmer
19:26.58fr0ggiei hear it temporarily reduces fertility, i hope so.
19:27.01jmetroonly a space heater cause its AMD 6 core XD
19:27.05drmessanoI run an Atom box at home.  All that heat is a waste
19:27.52drmessanoI stopped worrying about being the fastest kid on the block when I became an adult lol
19:33.57fr0ggieif it can make world overnight, im happy
19:36.29*** join/#asterisk qdel (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net)
19:36.35jmls1damn, 11 beta2 is pretty stable
19:36.52Qwelljmls1: You're still on the beta?
19:36.58jmls1ugh.
19:37.00jmls1rc
19:37.03Qwellahh
19:37.08jmls1latest svn, actually :)
19:37.23jmls1latest 11 svn branch to be pedantic
19:37.52drmessanoYou're supposed to call it TRUNK
19:37.55drmessanoAs in
19:37.58drmessanoI R ON 11 TRUNK
19:38.07drmessanoSVN == TRUNK
19:38.20jmls1yikes. Shouting !
19:38.24drmessanolol
19:38.40drmessanoNow, repeat after me...
19:38.50jmls1Asterisk SVN-branch-11-r374932
19:39.08jmls1doesn't say TRUNK in there :-D
19:39.10drmessanoI R USE SIP TRUNK ON ASTERISK 11 TRUNK
19:39.24drmessanoYou may need to practice it a bit
19:39.56jmls1System uptime: 5 days, 12 hours, 9 minutes, 2 seconds (up since we installed and fired it up)
19:40.06Qwelldays behind
19:40.17jmls157663 calls processed
19:40.20drmessanoand when someone says "Thats not trunk, that Asterisk 11 branch from SVN".. You tell them "I R WGET IT FROM SVN TRUNK!  U NO TELL ME!"
19:40.55drmessanoI should give a talk at Astricon next year about this.  Geesh
19:41.12jmls1jeez. All I wanted to say was "thanks devs for making this so stable and helping me run my business!" :P
19:41.24drmessanolol
19:42.05drmessanoI'm not a dev.  I am here for the clown show and the chicken nuggets.
19:42.25drmessanoOOOPS.. I mean, NOT CHICKEN NUGGETS.  BACK OFF
19:42.26jmls1yeah. I got slapped in -dev for mentioning the figures last time :'(
19:42.53jmls1nm. I'm pleased to be using *
19:42.58Nuggeteyes drmessano
19:42.58drmessanoSome of those guys in -dev are pretty psycho.  I appreciate their work, but glad they are kept in cages and fed with a long spork
19:43.40drmessanotelnet?
19:43.41Nuggettelnet is eeeeeeevil!
19:44.25drmessanoY R TALNET EVAL?  I R TALNET TO PREEPBX FOR SSH WITHOUT WINDOWS
19:44.36drmessano^_^
19:55.40*** join/#asterisk mjordan (mjordan@conference/digium/x-fbotxcbskjnaueww)
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20:01.46*** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net)
20:03.08chuckfSVN? Why are they not using GIT like everyone else?
20:03.52saxabecause there is not much difference between one and the other :)
20:04.32dijibfr0ggie: its a lie! i believe im now super fetile.
20:04.42dijibthanks R200
20:08.14*** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com)
20:36.49*** join/#asterisk infobot (~infobot@rikers.org)
20:36.49*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.9.0 (2012/10/08), 1.8.17.0 (2012/10/08), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.13 (2012/10/09) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Coming to AstriCon? /join #astricon

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