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) |
08:41.57 | *** join/#asterisk frawd (~francois@133.Red-83-41-197.dynamicIP.rima-tde.net) |
08:43.57 | *** join/#asterisk Howlader (howlader@soft.bdcom.com) |
08:45.32 | Howlader | Hi zafrir |
08:46.15 | Howlader | tzafrir, looking for your kind help |
08:46.26 | tzafrir | What about? |
08:46.54 | Howlader | i am facing a little problem. when I enable 2nd e1 port machine takes reboot. i am using digium t400p. |
08:49.00 | Howlader | tzafrir, 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 |
08:49.45 | tzafrir | t400p? That card uses the tor2 driver? |
08:50.59 | Howlader | sorry we are using t410p 4 port e1 card with dahdi driver. |
08:56.25 | Howlader | tzafrir, 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 |
08:57.21 | Howlader | can you give some suggestion, what to do? |
08:57.34 | Howlader | tzafrir, can you give some suggestion, what to do? |
08:57.48 | tzafrir | Sorry, I don't understand the problem |
08:58.01 | tzafrir | What version of DAHDI (zaptel?) doyou use? |
08:58.57 | Howlader | currently used dahdi 2.4.0 version |
09:00.47 | Howlader | among 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) |
09:01.43 | jmls | morning all |
09:01.57 | jmls | would someone be so kind as to enlighten me on OpenSIPS ? |
09:02.24 | jmls | If I use asterisk , what is the purpose of OpenSips ? What does it give me that asterisk doesn't ? |
09:02.37 | jmls | *newbie* alert ;) |
09:02.48 | Howlader | tzafrir, 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. |
09:03.23 | Howlader | tzafrir, currently using only 1 port out of 4 port from t410p. |
09:03.56 | tzafrir | What do you mean by "enable 2nd port"? What do you technically do? |
09:04.01 | SunTsu | jmls: are you sure that that's a question you want to ask in the asterisk channel? |
09:04.30 | SunTsu | "Hey guys, what about xy, what does it give me your stuff doesn't?" |
09:06.58 | p3nguin | jmls: What does it give you that Asterisk doesn't? a SIP proxy |
09:07.57 | jmls | p3nguin: 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 |
09:08.15 | jmls | SunTsu: why ? Asterisk handles stuff that OpenSIPS doesn't. |
09:08.17 | p3nguin | jmls: Well, Asterisk is not a "proxy." |
09:08.38 | p3nguin | jmls: Asterisk is a back-to-back user agent (b2bua). |
09:09.09 | p3nguin | jmls: The primary usage of OpenSIPS/OpenSER that I know of is a load-balancing SIP proxy. Asterisk isn't a SIP proxy. |
09:09.13 | SunTsu | p3nguin: although you could configure asteisk to emulate one |
09:09.23 | SunTsu | asterisk even |
09:09.34 | jmls | hmm |
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09:38.02 | Howlader | tzafrir, apologize again. internet connection inturpts |
09:38.55 | Howlader | tzafrir, enable mean active the 2nd port |
09:39.54 | tzafrir | technically, what did you do? |
09:40.05 | tzafrir | Was it already configured to work? |
09:41.06 | Howlader | yes its already configured. we provide IP telephony service. and the e1 channels connected with ICX. |
09:41.42 | Howlader | tzafrir, its SS7 connectivity |
09:44.46 | Howlader | first e1 port is active an running smoothly. for the issue i deactive the other ports. |
09:45.58 | Howlader | facing problem rest of ports, which is unable to acitve or run which cause halt down the call manager |
10:07.02 | p3nguin | Is 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. |
10:10.55 | tzafrir | Howlader, what call manager? |
10:12.40 | Howlader | call manager mean we use those machine as a call manager gateway |
10:13.57 | tzafrir | Looks like an issue with that call manager |
10:17.36 | *** join/#asterisk Howlader (howlader@soft.bdcom.com) |
10:18.06 | *** join/#asterisk voila (IceChat77@117.212.66.136) |
10:18.30 | voila | hello guys , is there any gui for creating dialup plan in asterisk |
10:19.11 | Howlader | tzafrir, yes the main issue is what reason makes the server halt down when active the other ports. |
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10:22.36 | tzafrir | Howlader, So this is the machine with the T410P card in it? |
10:22.57 | Howlader | yes this is the machine. |
10:23.00 | tzafrir | Your description is not clear, and is short on details |
10:23.21 | tzafrir | ok. So what exactly crashes: the machine completely shuts down? |
10:24.14 | Howlader | yes i apologized to describe it shortly. |
10:26.15 | Howlader | yes the issue arrived when try to acitve 2nd or 3rd or 4th port . |
10:28.20 | Howlader | currently 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 |
10:32.34 | p3nguin | What 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 |
10:33.03 | p3nguin | The 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 |
10:33.20 | p3nguin | Why is there a discrepancy? |
10:33.46 | p3nguin | It happens with every voicemail, as long as I have attach = yes set in voicemail.conf. |
10:38.25 | Howlader | tzafrir, can us suggest me how do i diagonesis to solve the issue? |
10:42.23 | p3nguin | If 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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11:15.40 | benngard | hmm, 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 :( |
11:19.30 | Howlader | tzafrir, u there? |
11:20.53 | tzafrir | Howlader, what exactly happens? Asterisk crashes? The system completely panics? |
11:21.57 | Howlader | the system completely panics & reboot the system. |
11:25.14 | Howlader | tzafrir, the system completely panics & reboot the system |
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11:40.21 | Howlader | tzafrir, please provide a suggestion? |
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11:46.51 | tzafrir | Howlader, any panic message on the screen? |
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11:48.45 | Howlader | no panic message showed. |
11:49.03 | Howlader | tzafrir, no panic message showed |
11:49.26 | tzafrir | Does it answer pings? |
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11:51.19 | Howlader | tzafrir, no pings. its take reboot shortly. |
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13:14.25 | tzafrir | Howlader, I suggest you try latest version of DAHDI (2.4.1) |
13:15.43 | Howlader | tzafrir, okay than i have prepare another call manager for implement with DAHDI (2.4.1) |
13:16.18 | Howlader | tzafrir, as i can not change on line server as it is running consumer services. |
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13:59.13 | LemensTS | I 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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18:06.20 | Killaklown | hi guys... anyone have spa3102? I'm having problems with hangup detection |
18:06.33 | Killaklown | phone keeps ringing even after the person calling hangs up |
18:10.48 | tonsofpcs | Killaklown: if you're searching forums, "hangup" is called "on-hook" |
18:11.03 | tonsofpcs | although you normally detect "off-hook" (or not) |
18:11.35 | Killaklown | tonsofpcs: I think its called "Detect Disconnect Tone" |
18:11.42 | Killaklown | in the spa3102 that is |
18:12.13 | tonsofpcs | no clue what an spa3102 is |
18:12.29 | Killaklown | oh |
18:12.33 | Killaklown | its an ATA |
18:12.36 | Killaklown | from cisco |
18:12.57 | tonsofpcs | then shouldn't it deal with standard POTS nomenclature?...... oh, cisco..... |
18:13.34 | Killaklown | yah |
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18:20.56 | dreamind | Hi folks |
18:23.12 | dreamind | I 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. |
18:23.54 | dreamind | I'm running everything on Squeeze now and use HFC ISDN-Cards if that matters anyhow... |
18:24.14 | dreamind | I guess its somehow related to iax2, but I couldn't figure out what exactly is wrong :( |
18:36.19 | dreamind | ah, sorry I figured out it was only because of the calling telephone I used for testing... |
18:36.42 | dreamind | somehow my sip provider seems to have problems with fax on the other side. Calling from mobile worked just fine. |
18:36.45 | dreamind | bye :) |
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18:45.53 | cusco__ | hi |
18:46.53 | cusco__ | having either a PRI card TE420 or a TE410P, it seems that I cannot use two PRI connections from different telcos. |
18:47.06 | cusco__ | seems that dahdi timers will only sync to one source |
18:47.20 | cusco__ | it causes some not wanted behaviour on the other telco.. |
18:47.26 | cusco__ | what can I do? |
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19:05.50 | Corydon76-home | cusco__: Ask the two telcos to establish a timing loop |
19:06.50 | Beltechs | Hello 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 |
19:07.27 | Corydon76-home | Beltechs: see doc/backtrace.txt |
19:07.57 | Beltechs | is that a doc on * site? |
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19:08.51 | Corydon76-home | Oh, right, Russell got rid of it |
19:09.14 | Corydon76-home | They were detailed instructions on how to get a backtrace |
19:09.53 | Beltechs | im googling now and checking the wiki, look like what I need to do, |
19:10.00 | Corydon76-home | https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
19:10.03 | Beltechs | I will take a stab at it |
19:10.38 | Beltechs | thank you. |
19:10.44 | Corydon76-home | yw |
19:13.26 | Killaklown | anyone know where I can get a good database of disconnect tones? I need one specifically for the Bahamas |
19:14.06 | Beltechs | should I reboot after #ps aux | grep asterisk ? to start * in "-g option" ? |
19:14.47 | Corydon76-home | Beltechs: you should be able to use the init.d scripts to stop and start Asterisk again |
19:15.16 | Corydon76-home | <PROTECTED> |
19:15.57 | Corydon76-home | If Asterisk is started with -g, it should deposit the core files in /tmp |
19:16.38 | Killaklown | cisco spa3012 disconnect tone please anyone |
19:16.39 | *** join/#asterisk wonderworld (~ww@port-92-201-102-102.dynamic.qsc.de) |
19:16.55 | Beltechs | its continuously trying to restart asterisk |
19:17.15 | Beltechs | different segmentation faults |
19:17.23 | cusco__ | Corydon76-home: thanks for replying. Would there be anything else I could do? |
19:17.33 | cusco__ | can't I use a internal timer (pseudo timer) or so? |
19:18.16 | p3nguin | If you install dahdi, you can use dahdi's dummy timer. |
19:18.24 | cusco__ | with a real PRI? |
19:19.10 | Beltechs | should I stop it at some point? |
19:20.04 | Corydon76-home | cusco__: Nope! |
19:20.12 | p3nguin | beltechs: Does "ps -C asterisk u" show that asterisk is running? |
19:20.23 | Corydon76-home | cusco__: you cannot fix a telco problem without getting the telco involved |
19:20.53 | p3nguin | Sometimes you can't get a telco problem fixed even with the telco involved. |
19:21.10 | Corydon76-home | p3nguin: then you have to fire the telco |
19:21.11 | cusco__ | heh |
19:21.12 | Beltechs | [root@beltechsPBX ~]# ps -C asterisk u |
19:21.12 | Beltechs | USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND |
19:21.13 | Beltechs | [root@beltechsPBX ~]# |
19:21.20 | p3nguin | Asterisk is not running. |
19:21.23 | cusco__ | ok so I need a new pri card |
19:21.31 | Beltechs | no |
19:21.32 | cusco__ | and a new machine, (since there are no more slots available) |
19:21.41 | p3nguin | beltechs: That's the command to check if it is running. |
19:22.00 | p3nguin | beltechs: What were you asking about stopping? Asterisk isn't running, so you can't stop it. |
19:22.01 | Beltechs | ya, my asterisk is not |
19:22.29 | Corydon76-home | Beltechs: As root, type: /usr/sbin/asterisk -vvvvvcg |
19:23.01 | Corydon76-home | Core should be left in the current directory |
19:23.27 | Corydon76-home | Just in case, probably: you should first do: cd /tmp |
19:23.30 | p3nguin | I would have used "/usr/sbin/asterisk -G asterisk -U asterisk -cddddvvvvg" |
19:23.36 | p3nguin | Note the -G and -U. |
19:24.02 | Beltechs | whats the difference in the commands? |
19:24.05 | Corydon76-home | p3nguin: 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 |
19:24.41 | Corydon76-home | p3nguin: I prefer as simple as possible, but no simpler |
19:24.48 | p3nguin | -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. |
19:25.19 | Corydon76-home | p3nguin: yeah, but that user and group should be specified in asterisk.conf, anyway |
19:25.23 | Beltechs | ok i did cd /tmp then /usr/sbin/asterisk -vvvvvcg |
19:25.33 | Corydon76-home | Beltechs: did it crash? |
19:26.52 | Beltechs | <PROTECTED> |
19:26.52 | Beltechs | Segmentation fault (core dumped) |
19:26.52 | Beltechs | [root@beltechsPBX tmp]# |
19:27.02 | Beltechs | thore are the last 3 lines |
19:27.14 | Corydon76-home | Beltechs: now: ls |
19:27.15 | p3nguin | core dumped; check your core file. |
19:27.27 | Corydon76-home | Something like core.12345 |
19:27.31 | Beltechs | ls in the tmp dir? |
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19:27.58 | Corydon76-home | Beltechs: in whatever your current directory is |
19:28.17 | Beltechs | ok did it let me look |
19:28.35 | Beltechs | core 5771? |
19:28.44 | Beltechs | core.5771 |
19:28.56 | Corydon76-home | Okay, so: gdb /usr/sbin/asterisk core.5771 |
19:29.37 | Beltechs | command not found |
19:29.43 | p3nguin | :/ |
19:29.58 | Corydon76-home | apt-get install gdb |
19:30.04 | Corydon76-home | or: yum install gdb |
19:30.32 | Beltechs | installing... |
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19:33.45 | Beltechs | results http://pastebin.com/RGmtbNP4 |
19:40.17 | Beltechs | ok so im learning about what we did here. GDB is a debugging tool/tracker. |
19:40.54 | Beltechs | so do I submit all or a portion of what was on the dump? |
19:47.56 | Beltechs | ok 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? |
19:48.33 | Corydon76-home | None of the pastebin is helpful |
19:48.47 | Corydon76-home | You need a 'bt' after that, for starters |
19:48.52 | Corydon76-home | See the wiki again |
19:49.48 | Beltechs | this >> gdb -se "asterisk" -ex "bt full" -ex "thread apply all bt" --batch -c core > /tmp/backtrace.txt |
19:49.59 | Beltechs | ? |
19:51.17 | Beltechs | no that didnt do anything |
19:53.34 | Beltechs | (gdb) bt |
19:53.35 | Beltechs | #0 0x08066fda in ast_append_ha () |
19:53.36 | Beltechs | #1 0x08120d08 in __init_manager () |
19:53.36 | Beltechs | #2 0x0808183a in main () |
19:54.24 | Beltechs | (gdb) bt full |
19:54.25 | Beltechs | #0 0x08066fda in ast_append_ha () |
19:54.25 | Beltechs | No symbol table info available. |
19:54.25 | Beltechs | #1 0x08120d08 in __init_manager () |
19:54.25 | Beltechs | No symbol table info available. |
19:54.25 | Beltechs | #2 0x0808183a in main () |
19:54.27 | Beltechs | No symbol table info available. |
19:54.29 | Beltechs | (gdb) |
19:55.25 | Beltechs | ok I went all the way to thread apply all bt |
19:57.46 | Beltechs | heres the results http://pastebin.com/dj7R6S5G |
19:59.53 | Beltechs | do i need to post this to the issue tracker? |
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21:02.41 | X-Rob | I am amused. |
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21:30.06 | X-Rob | tzafrir, ping? |
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22:27.01 | ccccp123 | any one alive here |
22:29.38 | tonsofpcs | nope |
22:29.42 | tonsofpcs | no one here but us chickens |
22:30.26 | ccccp123 | got a question how to make extantion act as trunk? |
22:30.38 | ccccp123 | sorry extention |
22:32.52 | leifmadsen | ~ask |
22:32.53 | infobot | Questions 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. |
22:33.11 | ccccp123 | well I am trying to connect fritzBox PSTN to asterisk using FB callthrough any idea this how far I went http://pastebin.com/uy763Q4B |
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22:34.18 | p3nguin | ccccp123: 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? |
22:36.53 | tzafrir | X-Rob, pong |
22:37.00 | yoda1410 | ping |
22:37.08 | ccccp123 | all calls starting 0044 must go through PSTN |
22:37.08 | ccccp123 | for 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 |
22:37.08 | ccccp123 | I 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 |
22:37.08 | ccccp123 | Please guide me here I am a quick learner. |
22:38.05 | ccccp123 | all calls starting 0044 must go through PSTN |
22:38.05 | ccccp123 | for 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 |
22:38.05 | ccccp123 | I 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 |
22:38.05 | ccccp123 | Please guide me here I am a quick learner. |
22:38.20 | ccccp123 | sorry guys new to this |
22:39.17 | ccccp123 | p3nguin yes sip peer pass call through it |
22:39.27 | yoda1410 | you must use Read and then GotoIf |
22:39.43 | yoda1410 | for authentication with pin 0000 |
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22:40.58 | yoda1410 | ccccp123: Where is the problem ? |
22:41.12 | ccccp123 | yoda can review this idea please http://pastebin.com/uy763Q4B |
22:41.18 | p3nguin | I don't know if you received that before I got disconnected, so I'll repeat it. |
22:41.23 | p3nguin | ccccp123: 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. |
22:41.26 | p3nguin | _0044XXXX.,1,Dial(DAHDI/g1/${EXTEN}) for example |
22:42.51 | ccccp123 | <p3nguin> I think you got me wrong FritzBox connected to pstn, aterisk running on VPS |
22:44.00 | yoda1410 | ccccp123: your dialplan does'nt contain authentication with pin code ? |
22:45.35 | ccccp123 | fritzBox doing authentification ,asterisk doesn't |
22:46.32 | ccccp123 | exten => 111222,2(calltrough),Dial(SIP/fbox1,,D(4321:)) |
22:46.42 | ccccp123 | 4321 my pin |
22:48.51 | p3nguin | You didn't tell me about all that bullshit when you started asking how to send calls to the PSTN. |
22:49.28 | p3nguin | And I don't have time to go over all that right now, so maybe someone else will be willing to get you where you need to go. |
22:50.37 | ccccp123 | you must be upset,sorry for not been clear BULL |
22:51.56 | yoda1410 | exten = 200,1,NoOp(Authentication with dial 200) |
22:51.56 | yoda1410 | exten = 200,n,Read(pin-code,please-enter-your-pin-code) |
22:51.56 | yoda1410 | exten = 200,n,GotoIf($[{pin-code} == "0000]?dial:hangup) |
22:51.56 | yoda1410 | exten = 200,n(dial),Dial(SIP/fbox1/004402033344111,,rR) |
22:51.56 | yoda1410 | exten = 200,n,Hangup() |
22:51.58 | yoda1410 | exten = 200,n(hangup),Hangup() |
22:52.07 | p3nguin | pastebin |
22:52.09 | p3nguin | ~pb |
22:52.10 | infobot | [~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. |
22:52.22 | yoda1410 | exten = 200,1,NoOp(Authentication with dial 200) |
22:52.23 | yoda1410 | exten = 200,n,Read(pin-code,please-enter-your-pin-code) |
22:52.23 | yoda1410 | exten = 200,n,GotoIf($[{pin-code} == "0000"]?dial:hangup) |
22:52.23 | yoda1410 | exten = 200,n(dial),Dial(SIP/fbox1/004402033344111,,rR) |
22:52.23 | yoda1410 | exten = 200,n,Hangup() |
22:52.29 | p3nguin | PASTEBIN |
22:52.39 | yoda1410 | certe ... |
22:53.50 | ccccp123 | <yoda1410> THANK YOU SO MUCH,did I mention 200 is my extention? |
22:55.58 | ccccp123 | <yoda1410> sorry I think you did not get me FritzBox is mini PBX and doing all authentification |
22:58.17 | ccccp123 | what 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 |
22:59.33 | ccccp123 | <yoda1410> thank for help but I think I will have to strugle on my own |
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