IRC log for #asterisk on 20110327

08:25.42*** join/#asterisk infobot (~infobot@rikers.org)
08:25.42*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/
08:36.11*** join/#asterisk Howlader (howlader@soft.bdcom.com)
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08:43.57*** join/#asterisk Howlader (howlader@soft.bdcom.com)
08:45.32HowladerHi zafrir
08:46.15Howladertzafrir, looking for your kind help
08:46.26tzafrirWhat about?
08:46.54Howladeri am facing a little problem. when I enable 2nd e1 port machine takes reboot. i am using digium t400p.
08:49.00Howladertzafrir, by the way i am from bangladesh, i used work at onecall solution ltd. Now i am having job at www.bdcom.com as an IPTSP system administrator
08:49.45tzafrirt400p? That card uses the tor2 driver?
08:50.59Howladersorry we are using t410p 4 port e1 card with dahdi driver.
08:56.25Howladertzafrir, even port 1 is running fine without any error. we were using only 1 port as our require. now for the current traffic update we need to open 2nd e1 channel. from then i am facing the issue that when ever i active the 2nd channel the total machine takes reboot
08:57.21Howladercan you give some suggestion, what to do?
08:57.34Howladertzafrir, can you give some suggestion, what to do?
08:57.48tzafrirSorry, I don't understand the problem
08:58.01tzafrirWhat version of DAHDI (zaptel?) doyou use?
08:58.57Howladercurrently used dahdi 2.4.0 version
09:00.47Howladeramong 4 e1 port currently we are using only 1 e1 port according our last traffic require and rest of port we were disabled.
09:01.37*** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com)
09:01.43jmlsmorning all
09:01.57jmlswould someone be so kind as to enlighten me on OpenSIPS ?
09:02.24jmlsIf I use asterisk , what is the purpose of OpenSips ? What does it give me that asterisk doesn't ?
09:02.37jmls*newbie* alert ;)
09:02.48Howladertzafrir, last day when i try to active second port the total system takes reboot & its happen randomly when ever i try to active 2nd port.
09:03.23Howladertzafrir, currently using only 1 port out of 4 port from t410p.
09:03.56tzafrirWhat do you mean by "enable 2nd port"? What do you technically do?
09:04.01SunTsujmls: are you sure that that's a question you want to ask in the asterisk channel?
09:04.30SunTsu"Hey guys, what about xy, what does it give me your stuff doesn't?"
09:06.58p3nguinjmls: What does it give you that Asterisk doesn't?  a SIP proxy
09:07.57jmlsp3nguin:  which is ? This is why I'm a little confused. In a phone config, I have an "oubound Proxy" which I point to an asterisk server
09:08.15jmlsSunTsu: why ? Asterisk handles stuff that OpenSIPS doesn't.
09:08.17p3nguinjmls: Well, Asterisk is not a "proxy."
09:08.38p3nguinjmls: Asterisk is a back-to-back user agent (b2bua).
09:09.09p3nguinjmls: The primary usage of OpenSIPS/OpenSER that I know of is a load-balancing SIP proxy.  Asterisk isn't a SIP proxy.
09:09.13SunTsup3nguin: although you could configure asteisk to emulate one
09:09.23SunTsuasterisk even
09:09.34jmlshmm
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09:37.18*** join/#asterisk Howlader (howlader@soft.bdcom.com)
09:38.02Howladertzafrir, apologize again. internet connection inturpts
09:38.55Howladertzafrir, enable mean active the 2nd port
09:39.54tzafrirtechnically, what did you do?
09:40.05tzafrirWas it already configured to work?
09:41.06Howladeryes its already configured. we provide IP telephony service. and the e1 channels connected with ICX.
09:41.42Howladertzafrir, its SS7 connectivity
09:44.46Howladerfirst e1 port is active an running smoothly. for the issue i deactive the other ports.
09:45.58Howladerfacing problem rest of ports, which is unable to acitve or run which cause halt down the call manager
10:07.02p3nguinIs there a setting for app_voicemail for the file permissions for the voicemail message files?  I'd like to be able to set the umask for all msg files when they get created.
10:10.55tzafrirHowlader, what call manager?
10:12.40Howladercall manager mean we use those machine as a call manager gateway
10:13.57tzafrirLooks like an issue with that call manager
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10:18.06*** join/#asterisk voila (IceChat77@117.212.66.136)
10:18.30voilahello guys , is there any gui for creating dialup plan in asterisk
10:19.11Howladertzafrir, yes the main issue is what reason makes the server halt down when active the other ports.
10:19.49*** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net)
10:22.36tzafrirHowlader, So this is the machine with the T410P card in it?
10:22.57Howladeryes this is the machine.
10:23.00tzafrirYour description is not clear, and is short on details
10:23.21tzafrirok. So what exactly crashes: the machine completely shuts down?
10:24.14Howladeryes i apologized to describe it shortly.
10:26.15Howladeryes the issue arrived when try to acitve 2nd or 3rd or 4th port .
10:28.20Howladercurrently we are running 1st port which is running the call manager without any inturption. but when ever try to acitve 2nd or 3rd or 4th port machine halt down
10:32.34p3nguinWhat does this mean?    [Mar 27 05:16:44] WARNING[2058]: app_voicemail.c:3100 base_encode: Failed to open file: /var/spool/asterisk/voicemail/default/201/tmp/h2VvrW.WAV: No such file or directory
10:33.03p3nguinThe temp file it was working on while the vm was being left was     -- x=0, open writing:  /var/spool/asterisk/voicemail/default/201/tmp/xg3ees format: wav49, 0x9ed2e18
10:33.20p3nguinWhy is there a discrepancy?
10:33.46p3nguinIt happens with every voicemail, as long as I have attach = yes set in voicemail.conf.
10:38.25Howladertzafrir, can us suggest me how do i diagonesis to solve the issue?
10:42.23p3nguinIf anyone is familiar with this voicemail issue, I'd be interested in hearing your thoughts about it.  I don't know if it is causing any problem or not, but I want to stop it from happening.
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11:15.40benngardhmm, i set a variable from ami, se the value of the variable in the dialplan, change it to something else, but from ami, i see the value i set from ami, not the value i modified from dialplan, guess i dont understand how variables work :(
11:19.30Howladertzafrir, u there?
11:20.53tzafrirHowlader, what exactly happens? Asterisk crashes? The system completely panics?
11:21.57Howladerthe system completely panics & reboot the system.
11:25.14Howladertzafrir,  the system completely panics & reboot the system
11:25.38*** join/#asterisk StaRetji (~BigAll@91.143.222.31)
11:40.21Howladertzafrir, please provide a suggestion?
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11:46.51tzafrirHowlader, any panic message on the screen?
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11:48.45Howladerno panic message showed.
11:49.03Howladertzafrir, no panic message showed
11:49.26tzafrirDoes it answer pings?
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11:51.19Howladertzafrir, no pings. its take reboot shortly.
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13:14.25tzafrirHowlader, I suggest you try latest version of DAHDI (2.4.1)
13:15.43Howladertzafrir, okay than i have prepare another call manager for implement with DAHDI (2.4.1)
13:16.18Howladertzafrir, as i can not change on line server as it is running consumer services.
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13:59.13LemensTSI have polycom 321's (no park button), its kind of confusing to hit transfer than press 700 to park...do you guys have a slicker imlplemntation?
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18:06.20Killaklownhi guys... anyone have spa3102?  I'm having problems with hangup detection
18:06.33Killaklownphone keeps ringing even after the person calling hangs up
18:10.48tonsofpcsKillaklown: if you're searching forums, "hangup" is called "on-hook"
18:11.03tonsofpcsalthough you normally detect "off-hook" (or not)
18:11.35Killaklowntonsofpcs: I think its called "Detect Disconnect Tone"
18:11.42Killaklownin the spa3102 that is
18:12.13tonsofpcsno clue what an spa3102 is
18:12.29Killaklownoh
18:12.33Killaklownits an ATA
18:12.36Killaklownfrom cisco
18:12.57tonsofpcsthen shouldn't it deal with standard POTS nomenclature?...... oh, cisco.....
18:13.34Killaklownyah
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18:20.56dreamindHi folks
18:23.12dreamindI happen to have a strange problem with asterisk (1.6.2.9), dahdi, iaxmodem and hylafax. Recently I upgraded from an older installation with zaptel and now incoming calls work with Record() and Playback(), but as soon as I forward the call to iaxmodem it gets after roughly 2 seconds terminated.
18:23.54dreamindI'm running everything on Squeeze now and use HFC ISDN-Cards if that matters anyhow...
18:24.14dreamindI guess its somehow related to iax2, but I couldn't figure out what exactly is wrong :(
18:36.19dreamindah, sorry I figured out it was only because of the calling telephone I used for testing...
18:36.42dreamindsomehow my sip provider seems to have problems with fax on the other side. Calling from mobile worked just fine.
18:36.45dreamindbye :)
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18:45.53cusco__hi
18:46.53cusco__having either a PRI card TE420 or a TE410P, it seems that I cannot use two PRI connections from different telcos.
18:47.06cusco__seems that dahdi timers will only sync to one source
18:47.20cusco__it causes some not wanted behaviour on the other telco..
18:47.26cusco__what can I do?
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19:05.50Corydon76-homecusco__: Ask the two telcos to establish a timing loop
19:06.50BeltechsHello I was attempting to upgrade from *1.6 to 1.8 and now I get these errors. Any help is appreciated. http://pastebin.com/7aCMT3YZ
19:07.27Corydon76-homeBeltechs: see doc/backtrace.txt
19:07.57Beltechsis that a doc on * site?
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19:08.51Corydon76-homeOh, right, Russell got rid of it
19:09.14Corydon76-homeThey were detailed instructions on how to get a backtrace
19:09.53Beltechsim googling now and checking the wiki, look like what I need to do,
19:10.00Corydon76-homehttps://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
19:10.03BeltechsI will take a stab at it
19:10.38Beltechsthank you.
19:10.44Corydon76-homeyw
19:13.26Killaklownanyone know where I can get a good database of disconnect tones?  I need one specifically for the Bahamas
19:14.06Beltechsshould I reboot after #ps aux | grep asterisk ? to start * in "-g option" ?
19:14.47Corydon76-homeBeltechs: you should be able to use the init.d scripts to stop and start Asterisk again
19:15.16Corydon76-home<PROTECTED>
19:15.57Corydon76-homeIf Asterisk is started with -g, it should deposit the core files in /tmp
19:16.38Killaklowncisco spa3012 disconnect tone please anyone
19:16.39*** join/#asterisk wonderworld (~ww@port-92-201-102-102.dynamic.qsc.de)
19:16.55Beltechsits continuously trying to restart asterisk
19:17.15Beltechsdifferent segmentation faults
19:17.23cusco__Corydon76-home: thanks for replying. Would there be anything else I could do?
19:17.33cusco__can't I use a internal timer (pseudo timer) or so?
19:18.16p3nguinIf you install dahdi, you can use dahdi's dummy timer.
19:18.24cusco__with a real PRI?
19:19.10Beltechsshould I stop it at some point?
19:20.04Corydon76-homecusco__: Nope!
19:20.12p3nguinbeltechs: Does "ps -C asterisk u" show that asterisk is running?
19:20.23Corydon76-homecusco__: you cannot fix a telco problem without getting the telco involved
19:20.53p3nguinSometimes you can't get a telco problem fixed even with the telco involved.
19:21.10Corydon76-homep3nguin: then you have to fire the telco
19:21.11cusco__heh
19:21.12Beltechs[root@beltechsPBX ~]# ps -C asterisk u
19:21.12BeltechsUSER       PID %CPU %MEM    VSZ   RSS TTY      STAT START   TIME COMMAND
19:21.13Beltechs[root@beltechsPBX ~]#
19:21.20p3nguinAsterisk is not running.
19:21.23cusco__ok so I need a new pri card
19:21.31Beltechsno
19:21.32cusco__and a new machine, (since there are no more slots available)
19:21.41p3nguinbeltechs: That's the command to check if it is running.
19:22.00p3nguinbeltechs: What were you asking about stopping?  Asterisk isn't running, so you can't stop it.
19:22.01Beltechsya, my asterisk is not
19:22.29Corydon76-homeBeltechs: As root, type:  /usr/sbin/asterisk -vvvvvcg
19:23.01Corydon76-homeCore should be left in the current directory
19:23.27Corydon76-homeJust in case, probably: you should first do:  cd /tmp
19:23.30p3nguinI would have used "/usr/sbin/asterisk -G asterisk -U asterisk -cddddvvvvg"
19:23.36p3nguinNote the -G and -U.
19:24.02Beltechswhats the difference in the commands?
19:24.05Corydon76-homep3nguin: that just ensures that it's a different user, and if it's in a root-only directory, ensures the core won't be produced
19:24.41Corydon76-homep3nguin: I prefer as simple as possible, but no simpler
19:24.48p3nguin-G and -U run Asterisk under a specified user and group.  Asterisk shouldn't be run as root, so you should test it as you run it normally.
19:25.19Corydon76-homep3nguin: yeah, but that user and group should be specified in asterisk.conf, anyway
19:25.23Beltechsok i did cd /tmp then /usr/sbin/asterisk -vvvvvcg
19:25.33Corydon76-homeBeltechs: did it crash?
19:26.52Beltechs<PROTECTED>
19:26.52BeltechsSegmentation fault (core dumped)
19:26.52Beltechs[root@beltechsPBX tmp]#
19:27.02Beltechsthore are the last 3 lines
19:27.14Corydon76-homeBeltechs: now:  ls
19:27.15p3nguincore dumped; check your core file.
19:27.27Corydon76-homeSomething like core.12345
19:27.31Beltechsls in the tmp dir?
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19:27.58Corydon76-homeBeltechs: in whatever your current directory is
19:28.17Beltechsok did it let me look
19:28.35Beltechscore 5771?
19:28.44Beltechscore.5771
19:28.56Corydon76-homeOkay, so:  gdb /usr/sbin/asterisk core.5771
19:29.37Beltechscommand not found
19:29.43p3nguin:/
19:29.58Corydon76-homeapt-get install gdb
19:30.04Corydon76-homeor: yum install gdb
19:30.32Beltechsinstalling...
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19:33.45Beltechsresults http://pastebin.com/RGmtbNP4
19:40.17Beltechsok so im learning about what we did here. GDB is a debugging tool/tracker.
19:40.54Beltechsso do I submit all or a portion of what was on the dump?
19:47.56Beltechsok so I see that the gdb tool created "New Thread 5772-5776" are these new bugs particular to my machine? or are these existing bug numbers that are already in the database?
19:48.33Corydon76-homeNone of the pastebin is helpful
19:48.47Corydon76-homeYou need a 'bt' after that, for starters
19:48.52Corydon76-homeSee the wiki again
19:49.48Beltechsthis >> gdb -se "asterisk" -ex "bt full" -ex "thread apply all bt" --batch -c core > /tmp/backtrace.txt
19:49.59Beltechs?
19:51.17Beltechsno that didnt do anything
19:53.34Beltechs(gdb) bt
19:53.35Beltechs#0  0x08066fda in ast_append_ha ()
19:53.36Beltechs#1  0x08120d08 in __init_manager ()
19:53.36Beltechs#2  0x0808183a in main ()
19:54.24Beltechs(gdb) bt full
19:54.25Beltechs#0  0x08066fda in ast_append_ha ()
19:54.25BeltechsNo symbol table info available.
19:54.25Beltechs#1  0x08120d08 in __init_manager ()
19:54.25BeltechsNo symbol table info available.
19:54.25Beltechs#2  0x0808183a in main ()
19:54.27BeltechsNo symbol table info available.
19:54.29Beltechs(gdb)
19:55.25Beltechsok I went all the way to thread apply all bt
19:57.46Beltechsheres the results http://pastebin.com/dj7R6S5G
19:59.53Beltechsdo i need to post this to the issue tracker?
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21:02.41X-RobI am amused.
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21:30.06X-Robtzafrir, ping?
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22:27.01ccccp123any one alive here
22:29.38tonsofpcsnope
22:29.42tonsofpcsno one here but us chickens
22:30.26ccccp123got a question how to make extantion act as trunk?
22:30.38ccccp123sorry extention
22:32.52leifmadsen~ask
22:32.53infobotQuestions 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.
22:33.11ccccp123well I am trying to connect fritzBox PSTN to asterisk using FB callthrough any idea this how far I went http://pastebin.com/uy763Q4B
22:33.11*** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil)
22:34.18p3nguinccccp123: I'm not exactly sure what you're trying to say, but it sounds like you are asking how to configure a SIP peer.  Is that what you're having problems with?  Or you don't know how to create the extension to send calls through the peer after you configured it?
22:36.53tzafrirX-Rob, pong
22:37.00yoda1410ping
22:37.08ccccp123all calls starting 0044 must go through PSTN
22:37.08ccccp123for example if I like to do it manually I would dial 200 than wait 2sec put pin 0000 then dial 0 and dial uk number (0044)02033344111
22:37.08ccccp123I am not sure what the best way to do that .I was thinking set-up custom trunk and outbound route with prepends 0000 not sure
22:37.08ccccp123Please guide me here I am a quick learner.
22:38.05ccccp123all calls starting 0044 must go through PSTN
22:38.05ccccp123for example if I like to do it manually I would dial 200 than wait 2sec put pin 0000 then dial 0 and dial uk number (0044)02033344111
22:38.05ccccp123I am not sure what the best way to do that .I was thinking set-up custom trunk and outbound route with prepends 0000 not sure
22:38.05ccccp123Please guide me here I am a quick learner.
22:38.20ccccp123sorry guys new to this
22:39.17ccccp123p3nguin yes sip peer pass call through it
22:39.27yoda1410you must use Read and then GotoIf
22:39.43yoda1410for authentication with pin 0000
22:40.52*** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com)
22:40.58yoda1410ccccp123: Where is the problem ?
22:41.12ccccp123yoda can review this idea please http://pastebin.com/uy763Q4B
22:41.18p3nguinI don't know if you received that before I got disconnected, so I'll repeat it.
22:41.23p3nguinccccp123: That has to be done using extension pattern matching and a Dial() command that sends the call through a device that connects to the PSTN.
22:41.26p3nguin_0044XXXX.,1,Dial(DAHDI/g1/${EXTEN})  for example
22:42.51ccccp123<p3nguin> I think you got me wrong FritzBox connected to pstn, aterisk running on VPS
22:44.00yoda1410ccccp123: your dialplan does'nt contain authentication with pin code ?
22:45.35ccccp123fritzBox doing authentification ,asterisk doesn't
22:46.32ccccp123exten => 111222,2(calltrough),Dial(SIP/fbox1,,D(4321:))
22:46.42ccccp1234321 my pin
22:48.51p3nguinYou didn't tell me about all that bullshit when you started asking how to send calls to the PSTN.
22:49.28p3nguinAnd I don't have time to go over all that right now, so maybe someone else will be willing to get you where you need to go.
22:50.37ccccp123you must be upset,sorry for not been clear BULL
22:51.56yoda1410exten = 200,1,NoOp(Authentication with dial 200)
22:51.56yoda1410exten = 200,n,Read(pin-code,please-enter-your-pin-code)
22:51.56yoda1410exten = 200,n,GotoIf($[{pin-code} == "0000]?dial:hangup)
22:51.56yoda1410exten = 200,n(dial),Dial(SIP/fbox1/004402033344111,,rR)
22:51.56yoda1410exten = 200,n,Hangup()
22:51.58yoda1410exten = 200,n(hangup),Hangup()
22:52.07p3nguinpastebin
22:52.09p3nguin~pb
22:52.10infobot[~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.us , 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.
22:52.22yoda1410exten = 200,1,NoOp(Authentication with dial 200)
22:52.23yoda1410exten = 200,n,Read(pin-code,please-enter-your-pin-code)
22:52.23yoda1410exten = 200,n,GotoIf($[{pin-code} == "0000"]?dial:hangup)
22:52.23yoda1410exten = 200,n(dial),Dial(SIP/fbox1/004402033344111,,rR)
22:52.23yoda1410exten = 200,n,Hangup()
22:52.29p3nguinPASTEBIN
22:52.39yoda1410certe ...
22:53.50ccccp123<yoda1410> THANK YOU SO MUCH,did I mention 200 is my extention?
22:55.58ccccp123<yoda1410> sorry I think you did not get me FritzBox is mini PBX and doing all authentification
22:58.17ccccp123what i need is dialplan for any number starting 0044 should be pass to extention 200 where pin 0000 and 0 for pstn then pass all number 0044XXXXX
22:59.33ccccp123<yoda1410> thank for help but I think I will have to strugle on my own
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