| 00:00.01 | saint_ | the funny thing is that once I playback a file after the read , then while the file plays and after it i will see the DTMF in the traces |
| 00:00.09 | saint_ | that's @#$(*& up |
| 00:00.37 | saint_ | oh well.. |
| 00:00.52 | JasonL | fixed it by setting pedantic=no in sip.conf |
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| 01:02.38 | saint_ | kaldemar: hey, can you give me a quick hand with one stuff that i change in my odbc query from yesterday night ? |
| 01:09.08 | saint_ | I have this in my func_odbc.conf: readsql=SELECT '${ARG1}' FROM `FIREHOUSES` WHERE `number` = '${ARG2}' |
| 01:09.25 | saint_ | and I try to call it with INCOMINGCALLER_CHECK(id,xxxxx) |
| 01:09.39 | saint_ | id being a field of my table , and xxx being a number in the table |
| 01:10.09 | saint_ | when I verbose ${INCOMINGCALLER_CHECK(id,1234)} , all it shows is "id" |
| 01:15.23 | saint_ | never mind . working now. |
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| 02:09.26 | SeRi | p3nguin: no use for the wo100 yet? |
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| 02:48.03 | saint_ | how to I test a value with asterisk..? GotoIf(${value} = x) or GotoIf(${value} == x) ..? |
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| 02:53.29 | WIMPy | You have to use $[]. |
| 03:08.33 | saint_ | just got it to work, thanks |
| 03:08.44 | saint_ | now.. is there a way to change the value of strftime ? |
| 03:08.52 | saint_ | I would like to extract the time, but to add 5 minutes to it |
| 03:09.25 | saint_ | like if it's 13:56:00 , I'd like to put in a variable 14:01:00 |
| 03:10.29 | WIMPy | You can use it twice. First to give you the unixtime, then add the time in seconds and then use it again with the result and the wanted format. |
| 03:11.27 | saint_ | when you mean unix time, you mean epoch ? |
| 03:11.43 | WIMPy | yes |
| 03:12.34 | saint_ | mmhh.. |
| 03:12.35 | saint_ | let me try to do something with that. |
| 03:20.06 | saint_ | WIMPy: working like a charm, thanks for the trick ! |
| 03:30.24 | saint_ | WIMPy: can I do something like that ? Set(ETA=${STRFTIME($[${STRFTIME(,,%s)} + 300],,%H:%M)}) |
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| 04:02.43 | saint_ | Can I use pattern matching symbol _ in a gotoif for a value that I just got using read ? |
| 04:03.06 | saint_ | I want to make sure that the read has a value 0-9 and not * nor # |
| 04:18.43 | saint_ | can someone help me with this error : WARNING[4400][C-0000002c]: ast_expr2.fl:470 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '>=', expecting '-' or '!' or '(' or '<token>'; Input: |
| 04:18.43 | saint_ | ( >= 0) & ( <= 9) |
| 04:19.01 | saint_ | i believe this is in reference to this line: GotoIf($[(${ETA} >= 0) & (${ETA} <= 9)]?UPDATE) |
| 04:21.23 | saint_ | maybe i need to write $[${ETA}] ? |
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| 05:51.42 | Carp1 | Hello. |
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| 06:24.08 | ATS63 | dufaq... 11.1? I'm running 1.6 |
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| 08:44.24 | x1user | [Dec 14 03:39:51] WARNING[3435][C-00000008]: pbx.c:4398 pbx_extension_helper: No application 'Dial{${FOO})' for extension (billing, 0888111432, 4) |
| 08:44.53 | x1user | I have include => extensions in billing and this numbers falls into some context but cant find out which one? How can i debug the dialplan? |
| 08:45.15 | x1user | core verbose and debug are set to maximum |
| 08:47.08 | slash213 | x1user, dialplan show |
| 08:47.36 | x1user | That just lists the dialplan, i need to follow the call trough the dialplan. |
| 08:48.46 | slash213 | x1user, well, you can simulate the call with it |
| 08:49.10 | slash213 | dialplan show 12345@from-pstn or something like that |
| 08:49.14 | ChannelZ | you have a syntax error |
| 08:49.25 | x1user | slash213: as I said i got includes, and cant follow it so easy because the dialplan is big |
| 08:49.57 | ChannelZ | Dial{ <--- not supposed to be a brace |
| 08:50.02 | x1user | slash213 dialplan show 12312@CONTEXT is ok thanks |
| 08:50.47 | x1user | ChannelIZ: you are right thanks :) |
| 08:51.27 | ChannelZ | sure |
| 08:52.06 | ChannelZ | But anyway a console verbose 3 *should* show you the dialplan as it executes. |
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| 08:57.30 | x1user | Can I debug variables in some easy way? I got same => Dial(SIP/{EXTEN}) and i got [Dec 14 03:56:00] WARNING[3447][C-0000000b]: app_dial.c:2121 dial_exec_full: Dial requires an argument (technology/resource) |
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| 08:59.09 | ectospasm | x1user: your dereference of ${EXTEN} is wrong (missing dollar sign [$]) |
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| 09:09.26 | x1user | ectospasm: no it is ${EXTEN} really but it seems i dont get the variable set right |
| 09:10.36 | ChannelZ | What extension is executing? |
| 09:10.41 | ChannelZ | Look at the console |
| 09:10.43 | ectospasm | x1user: there needs to be an antecedent to "same" |
| 09:10.44 | ChannelZ | core set verbose 3 |
| 09:11.30 | kaldemar | that warning shouldn't be possible if EXTEN is not empty. if the call is in dialplan, EXTEN cannot be empty. there's a typo of some kind. |
| 09:12.08 | ChannelZ | (and if you're going to show examples, paste/pastebin (if long) them exactly rather than re-typing even more type-os into them |
| 09:13.50 | kaldemar | x1user: and setting a value to EXTEN is not a good practice. use another variable. |
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| 09:23.09 | x1user | The problem is EXTEN is empty for some reason, i will try to find out why so |
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| 09:26.57 | slash213 | x1user, are you sure? have you tried noop? |
| 09:27.40 | x1user | <PROTECTED> |
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| 09:28.54 | slash213 | x1user, try to add noop(${EXTEN}) at the beginning of the context |
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| 09:40.19 | x1user | http://codepad.org/nzMzSf6N |
| 09:40.20 | kaldemar | EXTEN is not empty. you're executing one thing and modifying another. that Dial does not even have "SIP/" in it. |
| 09:41.14 | kaldemar | your extension in the CLI output is 0899023432. the show dialplan lists a pattern of _2., which does not match. |
| 09:41.59 | kaldemar | you're dialing the wrong number of modifying the wrong extension. |
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| 09:43.50 | x1user | so why dialplan show 2006558@billing shows that is goes to ougoing?? |
| 09:46.00 | kaldemar | it doesn't matter what it shows. your call never hits the extension. |
| 09:46.03 | slash213 | x1user, because 2006558 matches _2. |
| 09:46.11 | slash213 | 0899023432 does not |
| 09:47.13 | ghost75 | münchen? |
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| 10:41.41 | hurdman | hi |
| 10:42.19 | hurdman | what are your max calls/seconds with asterisk and digium card ? |
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| 10:46.02 | ectospasm | hurdman: I don't understand your question |
| 10:48.38 | hurdman | i want to know how many calls per seconds can do an asterisk server withh xx 8 ports E1 digium cards |
| 10:49.04 | hurdman | it's to understand if my top is normal or not |
| 10:50.30 | hurdman | i do about 240 simultaneous calls, with 2 Digium cards, on a Bi Xeon and 6 Go de Ram , i'm near a top of 30 ( 16 core about 50% and 1200 Mo off RAM used ) |
| 10:55.40 | hurdman | ectospasm: any idea ? |
| 10:56.04 | ectospasm | per second? That metric is meaningless. |
| 10:57.36 | ectospasm | hurdman: you should check out the dimensioning page: http://www.voip-info.org/wiki/view/Asterisk+dimensioning |
| 10:58.08 | ectospasm | that's just a rough guide |
| 10:59.15 | ectospasm | hurdman: with dedicated PSTN hardware like your Digium TE420s (assumed that's the model you have/will get), it incurs relatively little CPU usage |
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| 10:59.41 | ectospasm | if you're bridging them with any technologies that would require transcoding, it will increase some. |
| 10:59.54 | ectospasm | Do you intend to have 100% usage most of the time? |
| 11:00.53 | hurdman | no, but sometimes, i'll have more, so i want to know if i have to buy one more serv ( but dadhi card are nice but a bit expensive for me :) ) |
| 11:01.57 | hurdman | it looks like i have a lot of TLB ( or now Function call interrupts ) |
| 11:02.00 | ectospasm | hurdman: don't forget that the PSTN adapters also need a physical PSTN connection. |
| 11:02.10 | hurdman | yes :) |
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| 11:16.07 | hurdman | ectospasm: any idea on my interrupts ? |
| 11:17.21 | ectospasm | I don't know what you mean by TLB, and I can't answer the question about interrupts. |
| 11:25.01 | v0lZy | hey guys, I have a question. I'm designing a punch clock system and the current issue we have is that people dont adheer to the regieme strictly when it comes to punching in/out. As a solution, we have a guy control other people's time punching, but documenting it at the end of the day is way too complex so it boggs him down a lot because he has to take notes all the time |
| 11:25.20 | v0lZy | Now I have an idea to use asterisk in combination with a bash written punch system |
| 11:25.56 | v0lZy | I want the bash script to use asterisk to call the guy's phone and connect him into an IVR when he answers |
| 11:26.29 | Faustov | is there a way to distinguish availability due to agent registered in sip.conf but not on the network and not registered in sip conf at all? |
| 11:26.30 | v0lZy | that way, people can punch in/out without his physical presence, and when they do, he gets a call to either 'approve or deny' |
| 11:26.54 | v0lZy | ideally id do this through ssh |
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| 11:34.33 | kaldemar | v0lZy: what is your question? |
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| 11:41.42 | v0lZy | kaldemar: if its fisible to do it |
| 11:41.51 | v0lZy | trigger a call to a number from a script |
| 11:42.06 | v0lZy | then have the ivr send information back to the script |
| 11:42.23 | v0lZy | all i need is an 'ok/not ok' check |
| 11:43.02 | v0lZy | people can then come in late, or leave early, and when they check the time clock, the controling person gets a call |
| 11:43.28 | v0lZy | he can then just select either 1 or 2 ... 1 meaning 'ok, excused' and number 2 meaning 'count as is' |
| 11:44.28 | v0lZy | i think starting a call from a bash script should be possible |
| 11:44.45 | v0lZy | i suppose originate call and on answer bridge with an ivr extension |
| 11:45.07 | v0lZy | and in ivr, on selection, run bash script |
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| 11:45.09 | x1user | I got write("exec SetCDRUserField PrePaid"); in old *.agi php script, how to convert this to work with the new Set(CDR(userfield)=Value) ? |
| 11:46.45 | kaldemar | v0lZy: sure. it's a very common task. originate is the keyword. |
| 11:47.36 | kaldemar | x1user: do you have that write in your code directly or is it coming from a library? |
| 11:49.59 | x1user | kaldemar: it is directly in the code |
| 11:50.15 | x1user | This should be the standart system call i guess |
| 11:54.53 | kaldemar | x1user: a system call is something completely different. that is the exec agi command. "agi show commands topic exec" shows you usage in CLI. |
| 12:00.06 | x1user | I though i was writting to stdin, write seem to be deprecated, what is the equivelent in asterisk 11 ? |
| 12:00.15 | jonno11 | Hmm is it possible to execute multiple commands in ExecIf()? |
| 12:02.06 | kaldemar | x1user: an agi is supposed to write to stdout, not stdin. also, write() has nothing to do with asterisk. |
| 12:03.04 | kaldemar | jonno11: not directly, but you could use a subroutine. |
| 12:03.16 | x1user | agi show commands show that exec agi command is dead? |
| 12:03.55 | kaldemar | x1user: and what made you think the Dead option has anything to do with deprecation? |
| 12:04.44 | x1user | No I want to exec SetCDRUserField LOCAL |
| 12:04.52 | x1user | but SetCDRUserField is depricated |
| 12:05.19 | kaldemar | you already said Set(CDR(userfield)=Value) yourself. |
| 12:05.37 | kaldemar | so you use Set instead of SetCDRUserField. |
| 12:06.26 | x1user | The problem is that i cant get the right syntax, tried with exec Set (CDR(userfiled)=Value) |
| 12:07.06 | con3x | Can anyone give me an example config of SLA in chan skinny, I'm finding it hard to get running with just the documentation in the patch |
| 12:09.06 | *** part/#asterisk BenC[UK] (~bcummins@88.215.47.180) |
| 12:09.38 | kaldemar | x1user: is it Set((CDR...)) or Set(CDR...) ? |
| 12:10.04 | jacekowski | i need to make my asterisk server accessible from outside |
| 12:10.16 | jacekowski | it's behind NAT at the moment but i can forward some ports |
| 12:10.59 | con3x | jacekowski: Do you need SIP access from the outside? |
| 12:11.04 | jacekowski | yes |
| 12:11.29 | jacekowski | so my setup would be phone - NAT - internet - NAT with port forwarding - asterisk |
| 12:12.09 | con3x | Its actually a little more complicated than just forwarding :) I think I've got a good guide sitting around. |
| 12:12.18 | kaldemar | ~sipnat |
| 12:12.18 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the wiki at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, directmedia, externhost or externaddr, and localnet. |
| 12:12.52 | v0lZy | thanks kaldemar, ill look into it |
| 12:12.53 | v0lZy | gotta run now |
| 12:12.54 | v0lZy | byez |
| 12:13.04 | jacekowski | i was hoping that i could somehow restrict RTP to like 10-20 ports |
| 12:13.07 | jacekowski | and then i could forward thost |
| 12:13.09 | jacekowski | those |
| 12:14.23 | kaldemar | jacekowski: you can. define the ports in rtp.conf. |
| 12:15.04 | jacekowski | hmm, can i limit it to just one port? |
| 12:15.16 | jacekowski | or it needs one port for each RTP stream? |
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| 12:16.49 | con3x | jacekowski: even if one port does work, its UDP traffic so it may degrade quite quickly with one port. |
| 12:17.34 | jacekowski | how? |
| 12:17.39 | jacekowski | it's port just like any other |
| 12:19.07 | kaldemar | jacekowski: afaik, you need one port per stream so that would be a no. |
| 12:20.05 | *** join/#asterisk blitzrage (~lmadsen@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:20.05 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 12:20.08 | con3x | There is no flow control on UDP, so if packets arrive out of order or late due to multiple RTP stream filling the buffers then it would drop packets and jitter |
| 12:20.50 | con3x | (and apparently it doesn't work anyway, which is a good thing IMHO :)) |
| 12:21.02 | kaldemar | actually, one port for rtp and another for rtcp. |
| 12:21.47 | jacekowski | routers don't care if traffic is UDP or anything |
| 12:23.58 | *** join/#asterisk elico (~Thunderbi@bzq-79-182-198-188.red.bezeqint.net) |
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| 12:49.18 | con3x | jacekowski: they don't, but ordering of packets really matters in RTP :) |
| 12:51.50 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 12:52.23 | leifmadsen | if you have a jitterbuffer then asterisk can reorder the packets for you |
| 12:53.57 | coppice | any self respecting endpoint reorders the packets |
| 12:54.47 | leifmadsen | :) |
| 12:56.46 | con3x | That makes a lot of sense, I'm pretty sure the standard also has it written in :). I had a few problems a little while ago with RTP and Wifi latency (Badly configured Wifi with 200ms+ latency =/) |
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| 12:59.33 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 12:59.47 | bombev | hi |
| 13:01.16 | *** join/#asterisk Dovid (~Dovid@ool-1826d7a8.dyn.optonline.net) |
| 13:01.32 | bombev | what is that line: chan_sip.c: Sending fake auth rejection for device 1001 |
| 13:01.46 | bombev | in my asterisk log, i dont have extension as 1001 |
| 13:08.17 | kaldemar | bombev: see alwaysauthreject in http://svn.digium.com/svn/asterisk/tags/11.1.0/configs/sip.conf.sample |
| 13:08.39 | *** join/#asterisk jonno11 (~jonno11@host86-141-231-97.range86-141.btcentralplus.com) |
| 13:08.51 | Faustov | is there a way to distinguish unavailability due to agent registered in sip.conf but not on the network and not registered in sip conf at all? |
| 13:08.58 | Faustov | in the dialplan, that is |
| 13:10.03 | bombev | kaldemar thanks |
| 13:10.39 | bombev | do you know what is tha normal core set debug 1,2,3,4,5,6,7,8,9.... |
| 13:10.47 | bombev | and core set verbose 1,2,3,4,5,6,7,8,9..... |
| 13:11.47 | kaldemar | bombev: http://downloads.asterisk.org/pub/security/AST-2011-011.html |
| 13:12.13 | ghost75 | if somebody needs nagios script: http://exchange.nagios.org/directory/Plugins/Telephony/Asterisk/check_asterisk_peers/details |
| 13:18.51 | bombev | kaldemar so where should I put this: alwaysauthreject=yes |
| 13:25.58 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 13:29.08 | kaldemar | bombev: did you read the sample config i handed to you? |
| 13:30.52 | ghost75 | how high is your sip peer delay showing when you do show sip peers ? |
| 13:30.54 | kaldemar | bombev: it is yes by default, you just saw a notice of asterisk sending such a reject. |
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| 13:34.06 | bombev | axa |
| 13:34.09 | bombev | i got it |
| 13:34.34 | kaldemar | ghost75: one says 11 ms. are you asking what is "normal"? |
| 13:34.46 | ghost75 | yes what is tolerable |
| 13:35.07 | [TK]D-Fender | ghost75, Anything lower than your qualify setting on the peer |
| 13:35.07 | kaldemar | how do you define tolerable? |
| 13:35.14 | ghost75 | i have avg 70ms and sometimes goes to 130ms |
| 13:35.20 | [TK]D-Fender | ghost75, Doesn't matter |
| 13:35.25 | bombev | am what about the normal number of core set debug or verbose? |
| 13:35.41 | kaldemar | qualify=yes means qualify=2000 i.e. 2000 ms. |
| 13:35.52 | ghost75 | nokia phone phone on wlan was having about 200ms |
| 13:35.53 | *** join/#asterisk jonno11 (~jonno11@host86-141-231-97.range86-141.btcentralplus.com) |
| 13:36.04 | kaldemar | bombev: there is no such thing as "normal". |
| 13:36.45 | [TK]D-Fender | ghost75, SIP options = layer 7. The number isn't really that important. |
| 13:36.48 | ghost75 | i only know if its like 1000ms the opposite party doesnt understand any single voice |
| 13:36.59 | ghost75 | word, not voice |
| 13:37.29 | bombev | kaldemar whats the diff between debug and verbose |
| 13:37.35 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.83.105) |
| 13:38.15 | kaldemar | bombev: they are different types of output, controlled by different commands and settings. |
| 13:38.17 | [TK]D-Fender | bombev, Verbose shows dialplan execution. Debug shows an INSANE amount more |
| 13:41.13 | bombev | [TK]D-Fender aha, so core set debug 0 is okay and core set verbose 5 is okay? |
| 13:41.49 | [TK]D-Fender | bombev, ok for what? |
| 13:42.06 | [TK]D-Fender | bombev, Don't ask if a tool is good. It depends what you need it for. |
| 13:42.11 | bombev | normal values |
| 13:42.19 | [TK]D-Fender | bombev, That says nothing |
| 13:42.27 | bombev | okay |
| 13:42.49 | bombev | if i need to investigate failed call |
| 13:42.58 | bombev | what are best values |
| 13:43.40 | [sr] | is it possible to have timestamp on the console? |
| 13:43.51 | kaldemar | bombev: depends on what you want to know about the call. |
| 13:43.59 | kaldemar | [sr]: yes, see asterisk.conf |
| 13:44.08 | bombev | i need to know why the call got failed |
| 13:44.10 | [sr] | kaldemar: merci, let me see |
| 13:44.28 | kaldemar | bombev: start with core debug as 0 and some verbosity. |
| 13:44.39 | bombev | for example 5 |
| 13:44.45 | bombev | is it good enough |
| 13:44.53 | kaldemar | try it... |
| 13:44.59 | bombev | ok :) |
| 13:45.00 | bombev | thanks |
| 13:45.16 | kaldemar | [sr]: not that throughly explained in there, but it is the timestamp option. |
| 13:46.17 | [TK]D-Fender | bombev, how it failed. |
| 13:46.20 | [TK]D-Fender | depends* |
| 13:49.02 | [sr] | kaldemar: just checked |
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| 14:20.57 | qakhan | all, is there any Licensing on VOIP service in USA? |
| 14:22.16 | [TK]D-Fender | qakhan, No. |
| 14:24.25 | qakhan | if i sale VOIP services to any company in US then i dont require any license expect company to be registered |
| 14:24.29 | qakhan | i m right |
| 14:25.26 | qakhan | ? |
| 14:28.20 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-qhwpvknjislujnra) |
| 14:30.18 | [TK]D-Fender | qakhan, Your wording is dangerously vague. You should be very careful what you ask.... |
| 14:31.29 | [TK]D-Fender | qakhan, There are differences depending on the kind of service you are trying to sell. Calling-card companies are not primary telcos ILEC/CLEC/etc. I sincerely hope this isn't a field you're looking to get into at this point... |
| 14:32.00 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 14:32.15 | *** join/#asterisk iEatChildren (~iEatChild@174-19-64-81.omah.qwest.net) |
| 14:33.42 | qakhan | [TK]D-Fender i am not planning to sell calling-cards. just trying to sell VOIP service, like Conferencing system, Voice mail, Exts, etc.... |
| 14:34.20 | [TK]D-Fender | qakhan, that is not "VoIP Service", that is a HOSTED PBX. |
| 14:34.45 | [TK]D-Fender | qakhan, There is a difference between building roads and selling cars. |
| 14:34.53 | jacekowski | is it possible to have two phones register to one extension |
| 14:34.58 | iEatChildren | the fist column listed when i run "sip show channels" returns SIP/xxxxxxxxxx-00004 but when i run channel request hangup its showing items like SIP/xxxxxxxxxx-00004abc |
| 14:34.59 | jmetro | and selling parts you find on the side of the road.. |
| 14:35.08 | iEatChildren | am i missing something on sip show channels? |
| 14:35.15 | [TK]D-Fender | jacekowski, Yes, and as each does it will knock out the other's registration |
| 14:35.16 | jacekowski | or i need two extensions and then just use ring groups or stuff |
| 14:35.27 | [TK]D-Fender | jacekowski, So from a standpoint of ringing both... no. |
| 14:35.44 | [TK]D-Fender | jacekowski, You need 2 PEERS. Do not call them "extensions". |
| 14:36.15 | [TK]D-Fender | iEatChildren, because the column clearly isn't wide enough to show the whole thing |
| 14:36.22 | iEatChildren | how do i widen it? |
| 14:36.46 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.220.221) |
| 14:36.47 | [TK]D-Fender | iEatChildren, You don't. You use another function instead |
| 14:36.53 | iEatChildren | what function? |
| 14:37.03 | [TK]D-Fender | "core show channels concise", etc. |
| 14:37.23 | qakhan | [TK]D-Fender yes you are right Hosted PBX, thats what i meant |
| 14:37.57 | iEatChildren | thank you [TK]D-Fender |
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| 14:38.25 | qakhan | [TK]D-Fender is there any way, we setup conference bridge for limited time like 30 mins, |
| 14:39.34 | qakhan | if conference keep going till 30 mins then call disconnect |
| 14:39.38 | [TK]D-Fender | qakhan, You've been here for over a year now. How many options are there in meetme.conf? What does the app's command-line instructions say? How about app_conference? What about your ability to script something outside of * that will tell it to end? |
| 14:40.31 | qakhan | i cannot make any scripts |
| 14:40.45 | qakhan | but i just started learning PHP |
| 14:41.17 | [TK]D-Fender | qakhan, And you're thinking of running a hosted PBX platform for others? |
| 14:42.23 | qakhan | i have this plan in future not now |
| 14:42.30 | [TK]D-Fender | qakhan, Go learn Asterisk first |
| 14:42.38 | qakhan | ok |
| 14:43.15 | qakhan | thats why i am here to learn * |
| 14:44.44 | [TK]D-Fender | qakhan, Your questions aren't about your usage and problems you are having and you evidently aren't looking at the documentation that is provided for it. What kind of effort are you actually putting into this? |
| 14:45.11 | [TK]D-Fender | qakhan, Results are typically proportionate. |
| 14:46.36 | qakhan | ok |
| 14:47.50 | kaldemar | ConfBridge has marked and end_marked options that enable setting the first user as marked and kick everyone else out of the conference when the marked user leaves. then it's just a matter of making the first user leave after the timeout. |
| 14:48.55 | kaldemar | rough way for doing that would be for example to use a local channel and app Dial option L() for the first user. |
| 14:52.54 | Faustov | is there a way to distinguish unavailability due to agent registered in sip.conf but not on the network and not registered in sip conf at all? Something for the dialplan? |
| 14:53.06 | *** part/#asterisk iEatChildren (~iEatChild@174-19-64-81.omah.qwest.net) |
| 14:53.27 | *** join/#asterisk jkroon (~jkroon@105.241.120.65) |
| 14:53.46 | [TK]D-Fender | Faustov, What does "not registered in sip conf at all" mean? |
| 14:54.11 | Faustov | [TK]D-Fender: lacking context definition in sip.conf |
| 14:54.27 | [TK]D-Fender | Faustov, Peers register. They either are, or aren't. Whether a peer EXISTS at all is another matter. |
| 14:54.36 | [TK]D-Fender | Faustov, Where are you looking to check for this? |
| 14:54.40 | Faustov | I have misused the word register |
| 14:55.01 | [TK]D-Fender | Faustov, "core show function SIP_PEER" <- |
| 14:55.20 | [TK]D-Fender | Faustov, It'll tell you if it even exists, and various state info depending what you ask it. |
| 14:55.26 | Faustov | [TK]D-Fender: in the dialplan, I would like different handling when user dials an exten defined in sip.conf and different if not defined |
| 14:55.42 | Faustov | checking |
| 14:55.51 | [TK]D-Fender | Faustov, Why would you be allowing users to dial things that aren't valid? |
| 14:56.36 | Faustov | I wouldn't, however I would have to play back a message informing them that the extension is not valid |
| 14:56.53 | Faustov | but sometimes an agent might be offline - in which case the number is valid but not available |
| 14:56.56 | Faustov | different message needed |
| 14:57.46 | *** part/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 14:59.13 | [TK]D-Fender | <Faustov> I wouldn't, however I would have to play back a message informing them that the extension is not valid <- .... you ARE lettnig them dial something invalid. You jsut said so right here. |
| 14:59.43 | [TK]D-Fender | Faustov, DIALSTATUS will tell you what happened anyway. no need to check before.... |
| 15:01.11 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:01.19 | Faustov | [TK]D-Fender: problem is DIALSTATUS in both cases leads to "chanunavail", so I cannot play back different messages |
| 15:02.00 | [TK]D-Fender | Faustov, Again you are talking about allowing the to even THINK about dialing a peer that doesn't exist. Why are you letting them process a number that's not going to work? |
| 15:02.33 | Faustov | [TK]D-Fender: how would you recommend handling this? |
| 15:05.05 | jmetro | catch all your valid extens with dialplan... the "i" option we use for autoattendants means "invalid" which is almost what like youre suggesting but i'm not sure how that would work with outbound [still a noobie] |
| 15:07.03 | Faustov | the "i" option was heavily bugged |
| 15:07.05 | [TK]D-Fender | Faustov, If they dial direct from a SIP phone, "i" won't work. You would make a context that includes all your valid stuff in order and then a more generic catch-all to process the invalids |
| 15:07.26 | [TK]D-Fender | Faustov, I have not seen any "bugs", it's a question of understanding whewn it gets used |
| 15:07.38 | WIMPy | What's bugged about i? |
| 15:08.17 | Faustov | [TK]D-Fender: there were patches targetting "i", so I guess it is safe to assume it is bugged. They were not improvements |
| 15:08.32 | Faustov | WIMPy: sorry, I don't remember now, you'd have to check asterisk bug db |
| 15:08.33 | WIMPy | When? |
| 15:08.50 | [TK]D-Fender | Faustov, little is "safe to assume". It means you are making assumptions and not looking, reading, or validating. |
| 15:08.50 | Faustov | this year, sorry for not being precise |
| 15:08.57 | WIMPy | I do regularly and I don;t remember anything the like. |
| 15:09.15 | Faustov | [TK]D-Fender: nope, we were actually affected, I just don't remember the details |
| 15:09.38 | Faustov | well if it is really important/relevant, I could begin digging my mailbox |
| 15:14.17 | [TK]D-Fender | Faustov, Without details all it looks like is FUD-flinging... |
| 15:14.58 | [TK]D-Fender | Faustov, You are making assumptions on one side and unsupport vague hints at possible problems. All without any specifics on versions either. |
| 15:15.08 | [TK]D-Fender | Faustov, You do realize how horrible that looks, right? |
| 15:16.03 | Faustov | [TK]D-Fender: I'm already wasting my time finding that bug report only because you SEEM to have an idea how to resolve my problem ;) |
| 15:16.11 | Faustov | lets leave it at that |
| 15:16.40 | [TK]D-Fender | Faustov, Make you a deal.... I'll leave it at that if you don't repeat that approach again :) |
| 15:17.45 | Faustov | [TK]D-Fender: ok, lets assume (while I try to get the bug id), that "i" works exactly as designed - is it a part of your idea when it comes to my problem then? ;) |
| 15:18.37 | [TK]D-Fender | Faustov, Stop trying to weasel your way through this. They eat phone systems, human flesh is second nature to them.... |
| 15:19.33 | jmetro | playback(tt-weasels) |
| 15:20.36 | Faustov | [TK]D-Fender: I might get crucified for this, but I found this instead of the actual asterisk bug id: https://bugs.gentoo.org/show_bug.cgi?id=401015 |
| 15:21.26 | Faustov | nvm, here's the jira equivalent |
| 15:21.26 | Faustov | https://issues.asterisk.org/jira/browse/ASTERISK-17146?page=com.atlassian.jira.plugin.system.issuetabpanels%3Achangehistory-tabpanel |
| 15:22.04 | WIMPy | Chan_sip doesn't use the i extension. |
| 15:22.15 | [TK]D-Fender | Faustov, that has nothing to do with "i" |
| 15:22.26 | [TK]D-Fender | facepalms |
| 15:22.30 | Faustov | facepalms |
| 15:23.05 | [TK]D-Fender | Faustov, there is a PATTERN MATCH there. They HAVE a pattern. i is when they dial something that DOESN'T have a pattern match. |
| 15:23.39 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 15:23.55 | [TK]D-Fender | Faustov, you need to go read up on your dialplan basics.... |
| 15:24.42 | Faustov | [TK]D-Fender: perhaps I have misunderstood "i" then, however this brings us back to the basic question: pattern matches, but is invalid - how to distinguish between valid but offline? |
| 15:25.30 | [TK]D-Fender | Faustov, What part of "don't" was unclear? |
| 15:25.49 | [TK]D-Fender | Faustov, And I already gave you the answer ... to that thing I've told you you shouldn't be doing in the first place. |
| 15:27.09 | jmetro | hm. valid but offline..feels like if i have an extension 102, and his phone is off, it skips dialing him but goes to his VM? |
| 15:27.10 | Faustov | [TK]D-Fender: I'm not sure I understand how I could prevent dialing an invalid one in the first place |
| 15:27.40 | Faustov | SIPPEER is good, to a point - the sip.conf must be local |
| 15:28.04 | [TK]D-Fender | <[TK]D-Fender> Faustov, "core show function SIP_PEER" <- |
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| 15:28.24 | [TK]D-Fender | Faustov, Sorry... could you be a little more vague? |
| 15:28.57 | Faustov | [TK]D-Fender: to be precise, I could run the SIPPEER function locally to do exactly what I asked for |
| 15:29.10 | Faustov | but I have a number of agents on a remote system, which are advertised by dundi |
| 15:29.20 | [TK]D-Fender | ..... |
| 15:29.32 | Faustov | yes, I know, I didn't mention before, sorry ;) |
| 15:29.48 | WIMPy | Gret. They will only match if they exist. |
| 15:29.49 | [TK]D-Fender | grabs his rusty-nail enhanced ClueBat (tm) |
| 15:30.10 | [TK]D-Fender | Faustov, DUNDi is a DIALPLAN match. It is not a "sip peer discovery tool" |
| 15:30.28 | [TK]D-Fender | Faustov, It has no such functionality. This entire conversation has been a waste of time. |
| 15:30.43 | Faustov | well, I think I benefited a bit ;) |
| 15:30.51 | Faustov | so don't feel too bad about yourself |
| 15:31.29 | Faustov | WIMPy: correct, but to avoid advertising 50 extensions, I advertise paterns |
| 15:31.32 | [TK]D-Fender | I feel fine about myself .... you on the other hand are falling several notches for it.... |
| 15:31.55 | WIMPy | Then you're doing it wrong. |
| 15:32.01 | Faustov | damn |
| 15:32.05 | Faustov | that's what I was afraid of |
| 15:32.10 | [TK]D-Fender | <Faustov> WIMPy: correct, but to avoid advertising 50 extensions, I advertise paterns <- your advice sucks. Youa re going to pay a price for thinking sweeping patterns is a good idae and that you chould just validate afterwards. |
| 15:33.04 | Faustov | my logic was: why would the local admin have to remember to define a new context every time they add a new agent, AND THEN also add it to what is being advertised |
| 15:33.12 | Faustov | pattern would make it easier to manage |
| 15:33.22 | Faustov | I was not aware I shot my foot though |
| 15:33.29 | Faustov | or well, I hoped there was a smart way out of it |
| 15:33.47 | Faustov | do you guys consider the only option is to drop the patterns? |
| 15:34.02 | WIMPy | Yes. Define the estensions so that you don;t have to configure them elsewhere when using dundi. |
| 15:34.48 | WIMPy | If you advertise non existant extensions you're giving yourself a hard time. |
| 15:35.04 | Faustov | hmm fair enough |
| 15:35.28 | Faustov | well, thank you for the help, definitely not wasted time |
| 15:35.36 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
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| 15:38.35 | p3nguin | If you advertise a pattern on one system, but the extension doesn't actually exist on the other system, you'll end up with a congestion tone. |
| 15:38.55 | p3nguin | I think that would be enough for the dialing person to know he made an error. |
| 15:39.26 | [TK]D-Fender | p3nguin, it's so much worse than that... |
| 15:40.18 | p3nguin | Additionally, provide a catch-all for any extension that isn't explicitly defined. Then have the catch-all pattern play a message that you dialed an invalid number. |
| 15:40.52 | p3nguin | And this has almost nothing to do with sip.conf. |
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| 15:44.00 | [TK]D-Fender | Faustov, The context include method I mentioned does not apply to your situation. Only doing your systems right on each end will. |
| 15:44.13 | [TK]D-Fender | Faustov, No shortcuts |
| 15:46.08 | Faustov | point taken |
| 15:51.33 | Faustov | p3nguin: that's another way to look at it, but I think we got it now ;) |
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| 15:54.28 | *** mode/#asterisk [+o sruffell] by ChanServ |
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| 15:59.12 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:59.12 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 16:03.38 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.220.221) |
| 16:10.49 | *** join/#asterisk k611 (~K610@94.139.41.21) |
| 16:12.21 | tonyclewis | does anyone know why in Asterisk 1.8 on every version I have tested when a queue member transfers a queue call to another extension i no longer see transfer events in the queue logs |
| 16:12.36 | tonyclewis | this is with both asterisk feature code transfer and the phone transfer button |
| 16:12.47 | tonyclewis | Asterisk 1.4 would show this event |
| 16:17.17 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 16:19.48 | *** join/#asterisk lorsungcu (~anonymous@24-196-56-142.static.stcd.mn.charter.com) |
| 16:22.47 | *** part/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
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| 16:38.30 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 16:39.01 | *** join/#asterisk rogers- (rogers@bling.bling.org) |
| 16:40.01 | rogers- | Trying to figure out a problem between 2 IAX2 peers.. status is showing UNREACHABLE. I enabled debug, but all I see is RX-Frame retries, POKE/PONG. Any other way to get more detailed logs? |
| 16:40.54 | rogers- | each host can ping/ssh/etc to eachother, no firewalls |
| 16:44.53 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.38) |
| 16:45.24 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:50.17 | [TK]D-Fender | rogers-, Go prove the packets are making it. |
| 16:50.32 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
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| 17:01.31 | *** join/#asterisk salz212 (~chatzilla@182.185.243.208) |
| 17:02.23 | salz212 | Hello all, I am looking for a scenarios in which caller calls and connect to the callee but caller get to hear specified playback and the callee get to hear the RTP stream of caller. Is it possible with ChanSpy? |
| 17:02.48 | p3nguin | core show application ChanSpy |
| 17:04.09 | p3nguin | It sounds like what you want to do is have a caller launch an extension which originates a call to a callee and against application PlayBack, then you want to ChanSpy on the callee. |
| 17:04.41 | salz212 | yes I have tried that with o and b after prefixes.. but not getting to heart the audio.. I can see CLI than chanspy is connecting channels. |
| 17:04.43 | p3nguin | It should be possible. |
| 17:05.11 | salz212 | hear'* . |
| 17:05.18 | *** join/#asterisk lorsungcu_ (~anonymous@65.103.31.38) |
| 17:05.25 | p3nguin | b option is wrong for that purpose. |
| 17:05.41 | p3nguin | The call will not be bridged since it is with application playback rather than another channel. |
| 17:05.46 | p3nguin | The o option is right, though. |
| 17:06.48 | salz212 | yes I tried o first but did not get to hear the audio.. and then got illogical and tried b as well :D |
| 17:07.13 | p3nguin | Pastebin your dial plan. |
| 17:07.18 | salz212 | anyways thanks I will figure it out.. at least know I know chanspy is my thing.. for this |
| 17:07.47 | p3nguin | I can see it in my mind how I would write the dial plan to make this happen. |
| 17:10.19 | *** join/#asterisk mobile_gordita (~Robert@66-87-94-136.pools.spcsdns.net) |
| 17:10.35 | salz212 | I have just hard coded the chanspy prefix to SIP/myCallerextension thats it and I am playing sound files on the caller side... |
| 17:11.20 | _Corey_ | Anyone using Verizon's IP trunking product? We've got a customer getting a little nonsense about product certification... |
| 17:12.05 | salz212 | yes I have used Verizon IP trunk TF platform, DID platform etc. |
| 17:12.48 | _Corey_ | salz212: You're using Asterisk I presume... ? |
| 17:12.54 | salz212 | whats the problem. Actually its Verizon's policy of certifying their users before giving them complete setuo. |
| 17:13.16 | salz212 | Asterisk and Opensips.. but these trunks are with Asterisk for some reason. |
| 17:13.34 | _Corey_ | They've never heard of Switchvox and want the customer to pay $4k for a certification process |
| 17:13.43 | salz212 | All they need is trances to their scenarios.. and you pass the certification... |
| 17:14.22 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 17:14.22 | salz212 | $4k .. aah we paid nothing for this.. are you sure Verizon is asking for this or some other broker is in the middle?> |
| 17:14.34 | _Corey_ | No, VZ it seems |
| 17:15.16 | _Corey_ | These guys are pretty large, so I think VZ is shaking them down a bit |
| 17:15.41 | rogers- | [TK]D-Fender I am seeing pokes on both sides now... rebooted both systems. One side is working 100%, the other cannot make calls, showing busy/congested |
| 17:16.03 | salz212 | yep may be. Actually we were moving from one platform to another and they charged us nothing for that.. |
| 17:16.16 | _Corey_ | salz212: How long did the testing process take for you guys? Was it a one session deal? |
| 17:16.42 | jmetro | tell them Digium switchvox, i'm sure theyve heard of digium |
| 17:16.47 | [TK]D-Fender | rogers-, That doesn't sound like you looking at actual packet flow, peer states, etc... |
| 17:17.17 | rogers- | I'm seeing the same call numbers - Timestamp: 00013ms SCall: 03490 DCall: 00000 |
| 17:17.20 | rogers- | on both sides |
| 17:17.30 | salz212 | they reserve like 10 -14 days. for that.. their process is way too long.. we were behind schedule with them .. it was like 1 and half month .. after we got thiings working. |
| 17:18.09 | _Corey_ | salz212: Yeah, doesn't surprise me |
| 17:18.57 | _Corey_ | jmetro: You'd be surprised... most of these carrier guys know very little about this stuff |
| 17:19.03 | salz212 | we tried all different ways but they took their time... they are not worried about any thing.. I don't know why but they seem to care very less.. |
| 17:19.22 | *** join/#asterisk shadar (~eugene@37.113.133.194) |
| 17:19.47 | p3nguin | salz212: Here's what I imagine doing for your scenario. I have not tested it. http://pastebin.com/aJK9y9YJ |
| 17:20.14 | _Corey_ | salz212: Thanks for the feedback on the process |
| 17:20.44 | salz212 | actually what happen is .. we (clients) get to talk to Account managers they are sales people they don't know much about SIP/RTP stuff. but their tech team is not bad. |
| 17:21.07 | salz212 | no need to mention.. :) |
| 17:22.37 | *** join/#asterisk kfife (~Miranda@kfife.com) |
| 17:23.26 | salz212 | p3nguin thanks let me check it ..actually my dns is not resolving for paste bin :( wierd.. I will get back to you in min.. |
| 17:26.24 | jmetro | time to switch to 8.8.8.8 |
| 17:26.25 | p3nguin | This is the very basic procedure. This does not account for answering machines and voice mail. |
| 17:26.51 | p3nguin | For that, I would do it differently. |
| 17:33.52 | *** join/#asterisk areski (~areski@95.169.242.177) |
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| 17:38.13 | salz212 | p3nguin: in system command you are executing 2 local channels on at outbound and the other for what? custom playback? |
| 17:38.15 | *** join/#asterisk minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
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| 17:45.30 | *** join/#asterisk salz212 (~chatzilla@182.185.173.222) |
| 17:45.34 | salz212 | p3nguin: in system command you are executing 2 local channels on at outbound and the other for what? custom playback? |
| 17:49.37 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 17:49.58 | saint_ | morning all. |
| 17:51.06 | [TK]D-Fender | p3nguin, no need for the & at the end.... |
| 17:51.28 | [TK]D-Fender | p3nguin, And it is a nifty minimal basis but doesn't really work when concurrency comes in. |
| 17:51.47 | [TK]D-Fender | p3nguin, Mind you .. it's not supposed to be your job to code his project for him :) |
| 17:53.30 | salz212 | agree to the last line.. but now he had share it.. it does not make sense to me... I think my question was not taken in the right context. anyways thanks. |
| 17:56.07 | saint_ | can someone help me with this error : syntax error: syntax error, unexpected '>=', expecting '-' or '!' or '(' or '<token>'; Input ( >= 0) & ( <= 9) |
| 17:56.31 | saint_ | that's the code: GotoIf($[(${ETA} >= 0) & (${ETA} <= 9)]?UPDATE) |
| 17:56.43 | saint_ | and ETA get 1 digit from Read() |
| 17:56.56 | saint_ | I just want to make sure that the caller does not press * or # |
| 17:57.44 | [TK]D-Fender | salz212, it's 5 lines of dialplan, 3 of which are filler.... |
| 17:57.55 | [TK]D-Fender | salz212, What out of that are you having trouble with? |
| 17:58.51 | [TK]D-Fender | saint_, The user entered nothing so the var is blank. You can't have one side of your expression completely empty |
| 17:59.04 | [TK]D-Fender | saint_, Or it will complain like it did there |
| 17:59.39 | saint_ | [TK]D-Fender: no, I get this error only if I enter * or # |
| 17:59.57 | [TK]D-Fender | saint_, "#" is used to terminate Read() |
| 18:00.13 | [TK]D-Fender | saint_, terminating no other digits = blank |
| 18:00.37 | [TK]D-Fender | saint_, For further debugging plase feel free to pastebin the call attempt and raw dialplan |
| 18:00.39 | *** join/#asterisk ponyofdeath (~vladi@cpe-75-80-173-129.san.res.rr.com) |
| 18:00.54 | saint_ | [TK]D-Fender: here is my dialplan: http://pastebin.com/99CxMqWY |
| 18:01.16 | salz212 | system command.. I guess..My scenario is caller calls callee picks the call..... after that caller get to hear playback sound and calle get to hear the RTP steam of caller channel.. it means that caller does not know he is being spyed meanwhile listening to playback sound file. |
| 18:01.24 | [TK]D-Fender | saint_, Also, I don't recall () being valid in an expressions like that... |
| 18:01.25 | saint_ | all I am trying to do, is read 1 digit (under CallerOK) , and treat it only if it's 0-9 |
| 18:01.34 | ponyofdeath | hi, using asterisk 11.0.2 with google voice compiled on gentoo. After being able to receive calls for around an day asterisk stops accepting calls from google voice. I also see these in dmesg [170861.482788] asterisk[18921]: segfault at 0 ip 00007fc342aa4184 sp 00007fc311ce1fc0 error 4 in libiksemel.so.3.1.1[7fc342aa0000+d000] |
| 18:01.39 | ponyofdeath | any ideas? |
| 18:02.37 | [TK]D-Fender | saint_, Ass the call to the PB |
| 18:02.47 | saint_ | hu ? |
| 18:02.56 | [TK]D-Fender | Add* |
| 18:03.20 | saint_ | oh, ok . but does the code at n(CHECK_DIGIT) makes sens ? |
| 18:03.30 | [TK]D-Fender | <[TK]D-Fender> saint_, Also, I don't recall () being valid in an expressions like that... <- |
| 18:03.30 | saint_ | the GotoIf actually |
| 18:03.35 | [TK]D-Fender | gET RID OF THEM. |
| 18:04.03 | [TK]D-Fender | awaits correction for what was notified, testing, then new dialplan & CLI upon failure |
| 18:05.51 | saint_ | [TK]D-Fender: that's the log of the part of the call when i enter # http://pastebin.com/JaHCK5Dh |
| 18:06.03 | saint_ | if I enter a digit though, the error does not pop up |
| 18:06.26 | saint_ | i got ride of the () around the test of the values too in this example (and reloaded the dialplan) |
| 18:07.18 | [TK]D-Fender | saint_, "#" ends read by default. It is not part of the value returned by it. You are returning a BLANK answer |
| 18:07.45 | [TK]D-Fender | saint_, -- User entered nothing. <-------- this should have been a giant glowing neon sign to that fact |
| 18:08.33 | p3nguin | [tk]d-fender: If you don't use & to background it, System() will block and the dialplan will not continue execution. |
| 18:08.34 | saint_ | [TK]D-Fender: so why is the $READSTATUS = OK then ? |
| 18:08.38 | [TK]D-Fender | saint_, "core show application read" |
| 18:08.53 | saint_ | [TK]D-Fender: because if I enter litteraly nothing on the keyboard, then it shows TIMEOUT |
| 18:09.07 | [TK]D-Fender | p3nguin, yes but you are calling * with RX. It NEVER waits on that. Ther is no "effective" blackage.... |
| 18:09.48 | [TK]D-Fender | p3nguin, app_originate blocks. Calling "rx" like that jumps back pretty much instantly. |
| 18:10.13 | p3nguin | Every single time I have used it, it blocks and just sits there waiting for it to exit before continuing. |
| 18:11.58 | p3nguin | I made the same mistake thinking that it did not block and had to go back and change a bunch of lines of dial plan to background the command. |
| 18:12.18 | [TK]D-Fender | p3nguin, Well wudddyaknow.... |
| 18:12.34 | p3nguin | I guess you just tested it? |
| 18:12.44 | [TK]D-Fender | p3nguin, Yup, blocking. Guess I learned something today.... it shouldn't be.. |
| 18:13.13 | p3nguin | It was annoying when I found it that it was, but it was an easy workaround to make it proceed. |
| 18:13.32 | p3nguin | I don't think it always blocked like that. |
| 18:13.40 | [TK]D-Fender | p3nguin, True enough.... time to make an AMI-powered shell script ;) |
| 18:13.47 | p3nguin | It seems like back in 1.4 that it didn't block. |
| 18:14.03 | saint_ | [TK]D-Fender: ok, so I made a test with more verbose. with #, my value is empty. With *, my value has * , but it says syntax error: syntax error, unexpected '*', expecting $end; Input: * >= 0 & * <= 9 |
| 18:14.06 | [TK]D-Fender | saint_, Timeout != Opt out. |
| 18:14.41 | saint_ | [TK]D-Fender: so would you have by any chance a recommendation to validate or verify that the user entered only a digit from 0 to 9 without having the system yell at me ? |
| 18:14.56 | [TK]D-Fender | saint_, "core show application read" <- I recommend you read the options some more. |
| 18:15.08 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-237.west.biz.rr.com) |
| 18:15.13 | p3nguin | I haven't seen what you're doing, but it sounds like you are trying to compare a star character against a number. |
| 18:15.40 | [TK]D-Fender | p3nguin, He's thinking about validation without really having looked at what the app takes in. |
| 18:16.21 | [TK]D-Fender | saint_, * > 9 makes no sense. It should complain about it. |
| 18:16.31 | saint_ | i understand |
| 18:16.43 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 18:16.55 | [TK]D-Fender | saint_, So go read the app's instructions for limiting to VALID responses. |
| 18:18.52 | p3nguin | I might add a test right after the Read() to see if the caller entered valid characters. If not, I might play a message that it wasn't valid and then return them to the Read to try again. |
| 18:21.02 | *** join/#asterisk cyborg-one (~cyborg-on@212-178-2-212.broadband.tenet.odessa.ua) |
| 18:21.53 | p3nguin | Perhaps GotoIf($[${REGEX("[0-9]" ${ETA}]?:read) |
| 18:23.51 | saint_ | p3nguin: i don t see anything about regex in the book.. I will look online and try that. that might be perfect. make a test, and if it's something else than 0-9 then default ETA to 5. thanks ! |
| 18:24.14 | p3nguin | core show function REGEX |
| 18:27.09 | saint_ | can I do If($[${REGEX("[0-9]" ${ETA})}]?:Set(ETA=5)) ..? |
| 18:29.14 | p3nguin | Set(ETA=${IF($[${REGEX("[0-9]" ${ETA})}]?${ETA}:5)}) |
| 18:29.33 | [TK]D-Fender | saint_, REGEX is overkill and not required. |
| 18:29.35 | p3nguin | I think I got all my brackets and crap right on that. |
| 18:29.40 | saint_ | that s even better.. thanks |
| 18:29.49 | [TK]D-Fender | no need for brackets. Or regex. |
| 18:30.28 | saint_ | [TK]D-Fender: I search on google for "asterisk read limit input" but it did not come up with anything helpful to me.. |
| 18:30.39 | saint_ | [TK]D-Fender: i ll take your solution if you have something easier.. |
| 18:30.43 | p3nguin | I'm sure there are several ways to approach this. |
| 18:30.54 | [TK]D-Fender | <[TK]D-Fender> saint_, "core show application read" <- I recommend you read the options some more. |
| 18:31.04 | [TK]D-Fender | saint_, Read. The. Apps. Instructions. |
| 18:31.17 | p3nguin | In my version, there are no options indicative of this issue. |
| 18:31.22 | [TK]D-Fender | saint_, I did not say "Google stuff at random" |
| 18:32.25 | saint_ | [TK]D-Fender: Reads a #-terminated string of digits a certain number of times from the user . I understand that, but there is nowhere in the instructions where it says you can limit to a certain range of digits. |
| 18:32.40 | p3nguin | That's what I'm saying. |
| 18:32.43 | saint_ | since the end user can enter * and # , I need to make a test against the value |
| 18:33.53 | [TK]D-Fender | saint_, $["$ETA" = "" | "$ETA" = "*"] |
| 18:34.09 | [TK]D-Fender | saint_, $["$ETA" = "" | "$ETA" = "*" | "$ETA" = "#"] |
| 18:34.20 | [TK]D-Fender | Assuming you disabled # so it's read as a char at all |
| 18:34.27 | [TK]D-Fender | that';s your worst case. |
| 18:34.56 | saint_ | you are suggesting to replace the regex with this ? |
| 18:35.32 | p3nguin | I was testing for the chars to be digits, but this is checking to see if they are the bad things. |
| 18:36.20 | p3nguin | empty, star, or hash/pound |
| 18:37.20 | [TK]D-Fender | Hash by the pound! |
| 18:39.32 | SuperNull | mmm Hash. |
| 18:43.08 | saint_ | [TK]D-Fender: why arent the variables name in between { } ? |
| 18:43.25 | p3nguin | accidental omission |
| 18:43.55 | p3nguin | I think we've all done it at some point. |
| 18:43.55 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:43.55 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:44.25 | saint_ | ha ok. sorry to ask, the book does not talk about IF , and the IF examples in the voip-info are minimals |
| 18:44.43 | saint_ | [TK]D-Fender: but you still need to put ${ETA} in between " ? |
| 18:46.20 | p3nguin | core show function IF |
| 18:46.44 | p3nguin | I hope you're beginning to see a pattern about core show whatever. |
| 18:46.53 | saint_ | yes, I was going to say it |
| 18:47.07 | saint_ | too bad core show function IF does not show anything interesting.. |
| 18:48.37 | p3nguin | Let me tell you about the quotes. In asterisk, quotes are literal quotes. In the case of comparing one quoted string to another quoted string, you will never have a null value to compare to something else. In the case where ${ETA} can be null, you cannot compare it to anything else without a horrific error. Where ${ETA} is null, "${ETA |
| 18:48.44 | p3nguin | crap |
| 18:48.57 | p3nguin | Where ${ETA} is null, "${ETA}" = "" |
| 18:50.31 | [TK]D-Fender | saint_, $["${ETA}" = "" | "${ETA}" = "*" | "${ETA}" = "#"] |
| 18:51.14 | saint_ | okay.. thank you guys |
| 18:52.28 | saint_ | [TK]D-Fender: but we agree that I can get ride of the = "#" part, since when I push # on a phone, I get " " , correct ? |
| 18:53.58 | [TK]D-Fender | saint_, if it's filtered |
| 18:54.02 | saint_ | okay, so that's working like a charm. thanks p3nguin / [TK]D-Fender : Set(ETA=${IF($["${ETA}" = "" | "${ETA}" = "*"]?5:${ETA})}) |
| 19:07.51 | Qwell | Penguin: how the heck is that not registered already? |
| 19:09.03 | Penguin | It was when I switched to it. |
| 19:09.05 | Penguin | But now it's mine. |
| 19:09.24 | Qwell | O.o |
| 19:09.40 | Penguin | It was expired. |
| 19:09.44 | Qwell | ahh |
| 19:10.17 | Penguin | I've been trying to have it for eight years. |
| 19:10.31 | Qwell | >.> |
| 19:11.17 | Penguin | This makes me a happy penguin. |
| 19:13.14 | [TK]D-Fender | Penguin, Sucks.. p3n<tab> had so much better odd for auto-complete |
| 19:13.20 | [TK]D-Fender | s |
| 19:13.46 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 19:14.38 | Penguin | Even just p3 was almost never going to match another nick, but unless we get a penpal or pencil, pen<tab> will still have reasonable odds. |
| 19:15.02 | Penguin | (or any of several other nicks starting with pen) |
| 19:15.11 | [TK]D-Fender | Penguin, REASON HAS NOTHING TO DO WITH IT! |
| 19:15.17 | Penguin | ;) |
| 19:15.28 | [TK]D-Fender | </unexpectedhonesty> |
| 19:15.39 | *** join/#asterisk fibres (~no@77.107.115.242) |
| 19:15.45 | fibres | Evening All. |
| 19:15.48 | [TK]D-Fender | Penguin, You are killing nostalgia, one letter at a time.... |
| 19:15.49 | Penguin | Since when do we use reason, anyway, right? |
| 19:16.20 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:17.08 | *** join/#asterisk rgsteele (~rgsteele@12.150.6.65) |
| 19:17.41 | fibres | Can anyone give me any idea why on all versions of asterisk above 1.4 I am getting very poor CPS throughput? |
| 19:18.27 | fibres | On 1.4 I can get 80 cps with 1200 concurrent calls. Move to any newer version and at 10cps it starts to fail to drop calls and system starts having issues. |
| 19:19.16 | rgsteele | So, I recently moved my asterisk server to a new subnet. It had been logging CDR to a PG server on the same subnet. When I moved it, I updated all relevant network configs, and I could still talk to the PG server, but for some reason, cdr_pgsql.so seems to send empty queries and columns: INSERT INTO cdr () VALUES () |
| 19:19.35 | rgsteele | I tested connecting to PG manually from the asterisk server, and it works fine. |
| 19:19.49 | rgsteele | Also, the local logger (master.csv) contains all the appropriate fields and data |
| 19:19.58 | Penguin | I used to be able to have 200+ tabs open in firefox 2. Now that they've got like firefox 96, I bet I can't run 50 tabs. Same crappy reasons behind it, I'd guess. |
| 19:20.34 | rgsteele | I didn't change any of the configs, and I have all the proper libs installed. Reloading cdr_pgsql.so doesn't generate any errors. |
| 19:20.43 | fibres | Hi Penguin. I know where your comming from, but thats one hel of a decrease. |
| 19:20.59 | Penguin | I'm speculating. |
| 19:21.31 | rgsteele | Here are the errors in the Asterisk logs: http://pastie.org/5531820 |
| 19:21.38 | g_r_eek | i have modified app_voicemail.c and recompiled when i try to load the new app_voicemail.so module i have the error: "Module 'app_voicemail.so' was not complied with the same compile-time options as this version of Asterisk." Is there something wrong or is it just a stability measure than i can force somehow> |
| 19:21.44 | rgsteele | The PG server shows the same - empty column and value lists. |
| 19:22.04 | fibres | Im just wondering if there is a setting that needs changing, tweaking somewhere to increae throughput? |
| 19:22.25 | [TK]D-Fender | g_r_eek, You'll need to recompile all of * along with your custom ver. your binary fell out of spec with the rest of the * you kept |
| 19:22.38 | Penguin | I don't have that kind of call volume, so I'm not going to be able to give you a valuable answer. |
| 19:22.39 | [TK]D-Fender | fibres, We have no machine specs, OS, etc. |
| 19:23.04 | rgsteele | the cdr_pgsql.so file hasn't changed since February, so I doubt that's the cause |
| 19:23.10 | g_r_eek | [TK]D-Fender: i modified the .c file and than i did ./configure and make |
| 19:23.36 | g_r_eek | [TK]D-Fender: do i need to do a make clean b4? |
| 19:23.40 | [TK]D-Fender | g_r_eek, Yes, and the compile time options do not match the binaries of the rest of your installed system |
| 19:23.57 | [TK]D-Fender | g_r_eek, "make" alone for no install your binaries... |
| 19:24.03 | [TK]D-Fender | does not* |
| 19:24.33 | fibres | Hi [TK]D-Fender, This is running on a Dual Quad core Xeon 2.0 Ghz with 8gb ram and 146gb SAS 10k drivers running ubuntu 12.04 64 bit. |
| 19:24.41 | fibres | It is same machine I have tried all version on. |
| 19:24.43 | g_r_eek | [TK]D-Fender: so i need to run make install? Can i just do it for this specific module only? |
| 19:25.35 | g_r_eek | [TK]D-Fender: finish first with fibres and talk to me after :) |
| 19:26.40 | Penguin | You shouldn't have ran configure. |
| 19:26.40 | [TK]D-Fender | g_r_eek, you should do this for all modules. because YOUR's isn't matching everything else you left as-is/. |
| 19:26.50 | jacekowski | Penguin: firefox always had problems with tabs |
| 19:27.21 | jacekowski | Penguin: only browser so far that always survived when i had 200+ tabs open was opera |
| 19:27.45 | Penguin | I used to run way over 200 when firefox 2 was current. |
| 19:27.56 | Penguin | I used 2 for a long time after they went to 3. |
| 19:28.06 | g_r_eek | [TK]D-Fender: i am just to afraid of doing it , its a freepbx distro install and i just wanted to change voicemail app to send to multiple emails |
| 19:28.18 | Penguin | But then, as updates came to the system, 2 eventually stopped working for me and I never bothered to try to make it work. |
| 19:28.59 | Penguin | I thought app_voicemail would send to more than one email address already. |
| 19:29.11 | jacekowski | g_r_eek: it may be easier to do it with postfix |
| 19:29.14 | g_r_eek | Penguin: really? how? |
| 19:29.39 | g_r_eek | jacekowski: they need to put the emails in on fpbx gui |
| 19:30.08 | g_r_eek | jacekowski: they are 200 and more extensions i can set aliases for each one and change it when they want |
| 19:30.48 | Penguin | I'm not saying it does, I'm saying I thought it does. |
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| 19:31.08 | g_r_eek | Penguin: np :( it does not |
| 19:31.36 | Penguin | If I needed it to do that and found that it could not, I would probably handle it at the mail server level. |
| 19:31.59 | Penguin | I understand why you don't want to go about it in that way, though. |
| 19:32.36 | jacekowski | i would modify freepbx to update mail server config |
| 19:32.46 | g_r_eek | [TK]D-Fender: so i understand right, after i modify the .c files i run make clean, ./configure, make, and i have to do make install right? |
| 19:33.07 | g_r_eek | jacekowski: i could not do that |
| 19:33.24 | Penguin | You can take the app_voicemail.so file from the source tree and manually put it into the modules directory. |
| 19:33.43 | [TK]D-Fender | g_r_eek, YES |
| 19:34.02 | g_r_eek | Penguin: i put it and when i try to load it i get: Module 'app_voicemail.so' was not compiled with the same compile-time options as this version of Asterisk. |
| 19:34.14 | [TK]D-Fender | <jacekowski> g_r_eek: it may be easier to do it with postfix <- proper way |
| 19:34.18 | Penguin | Oh, right. |
| 19:35.03 | g_r_eek | [TK]D-Fender: thanks, will the make install destroy any conf's, freepbx settings or so? |
| 19:35.12 | jacekowski | g_r_eek: build it with same options |
| 19:35.44 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 19:35.55 | g_r_eek | jacekowski: is just a stock freepbx iso distro…i dunno all the options it uses |
| 19:36.43 | jacekowski | get srpm for it |
| 19:37.35 | g_r_eek | the rpm package for freepbx? what is an srpm? |
| 19:37.56 | navaismo | source rpm i guess |
| 19:38.04 | g_r_eek | the source from asterisk was not there i dl manually |
| 19:38.25 | [TK]D-Fender | g_r_eek, no |
| 19:38.46 | g_r_eek | [TK]D-Fender: thanks i try |
| 19:39.20 | g_r_eek | [TK]D-Fender: do i need to do any menuselect? |
| 19:40.10 | Hive | I know you can have a macro executed on the recieving party of a multiple peer dial command with Dial(SIP/101&SIP/102,ringtime,M(macroname[^arg1[^arg2]])), but is there a way to have a macro execute on the hangup of the channel? |
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| 19:46.05 | Hive | I know that I could put a macro in the hangup context, however that uses channel variables for the CALLER, but i need it for the CALLEE (which is what the above dial macro executes on) |
| 19:51.25 | *** part/#asterisk navaismo (~Administr@189.144.211.20) |
| 19:54.11 | Hive | Or perhaps there is a way to cause the calling party to inherit some channel variables from the called party |
| 19:54.27 | g_r_eek | [TK]D-Fender: make install stops: res_timing_pthread.so |
| 19:54.28 | g_r_eek | <PROTECTED> |
| 19:55.33 | Qwell | g_r_eek: read the whole message. |
| 19:56.01 | g_r_eek | Qwell: from where? |
| 19:56.15 | Qwell | from the screen. |
| 19:58.10 | g_r_eek | after the waring just the prompt |
| 19:58.15 | g_r_eek | *Warning |
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| 19:59.26 | g_r_eek | hmm asterisk seems to run but no extensions are registering and sip show peers gives me: |
| 19:59.27 | g_r_eek | No such command 'sip show peers' (type 'core show help sip show' for other possible commands) |
| 19:59.28 | g_r_eek | sip2*CLI> |
| 20:01.09 | Penguin | Your chan_sip.so doesn't seem to be loaded. |
| 20:01.35 | jpsharp | module load chan_sip.so |
| 20:01.52 | g_r_eek | Penguin: oh…nothing works now what can i do? i knew is should not mess with make install.... |
| 20:02.30 | g_r_eek | Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_xml_get_root |
| 20:02.45 | Qwell | g_r_eek: read the whole message. |
| 20:03.33 | g_r_eek | on the configure i had to do: ./configure --disable-xmldoc because even the xmldic seems to be installed it did not find it |
| 20:03.51 | Qwell | Did you read the message? |
| 20:04.04 | g_r_eek | Qwell: Command 'module load chan_sip.so' failed. |
| 20:04.04 | g_r_eek | [2012-12-14 15:02:09] WARNING[7735]: loader.c:460 load_dynamic_module: Error loading module 'chan_sip.so': /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_xml_get_root |
| 20:04.05 | g_r_eek | [2012-12-14 15:02:09] WARNING[7735]: loader.c:850 load_resource: Module 'chan_sip.so' could not be loaded. |
| 20:04.06 | g_r_eek | sip2*CLI> |
| 20:04.21 | Qwell | stops bothering |
| 20:04.24 | Qwell | good luck |
| 20:04.54 | g_r_eek | Qwell: sorry did you mean the message on the cli or any other message? |
| 20:06.27 | g_r_eek | please help i need to have my machine running again... |
| 20:07.15 | jpsharp | You've got a jacked up install. Remove asterisk, run "make clean", re run make, then reinstall. |
| 20:08.13 | g_r_eek | jpsharp: how do i remove asterisk? its a freepbx distro |
| 20:09.05 | Qwell | jpsharp: The best part? He doesn't even need to change anything in order to do what he wants. |
| 20:09.39 | jpsharp | g_r_eek: You're boned, then. You mess with anything under the hood of freepbx and you're going to break everything. |
| 20:09.44 | jpsharp | fdisk, format, reinstall...doohdah. |
| 20:10.09 | g_r_eek | jpsharp: thanks :( |
| 20:12.20 | Penguin | fdisk, format... really? All over the software being broken? |
| 20:12.54 | [TK]D-Fender | Penguin, His system am screw from running that command! |
| 20:13.06 | Penguin | My system is screw! |
| 20:13.20 | Penguin | I bet I still have that shirt. |
| 20:14.04 | g_r_eek | [TK]D-Fender: i fought you told me it will not break anything on fpbx... |
| 20:15.29 | [TK]D-Fender | <g_r_eek> [TK]D-Fender: thanks, will the make install destroy any conf's, freepbx settings or so? <[TK]D-Fender> g_r_eek, no |
| 20:16.20 | Penguin | http://imagebin.org/index.php?mode=image&id=239384 |
| 20:16.34 | Penguin | Still got it. |
| 20:17.14 | SuperNull | Damn my international provider.. they want a list of all source numbers.. but me being retarded.. i kept stealing all the 8675309 #s.. and routing it to my android sip client (amoung other numbers) |
| 20:17.51 | jmetro | Penguin I dont get it |
| 20:19.36 | [TK]D-Fender | jmetro, YOU WOULDN'T UNDERSTAND, YOU WEREN'T THERE! |
| 20:19.39 | [TK]D-Fender | </nam> |
| 20:20.25 | Penguin | <PROTECTED> |
| 20:20.27 | Penguin | <PROTECTED> |
| 20:21.21 | Penguin | Shall I paste what led up to that? |
| 20:21.53 | jmetro | im guessing it was somethign like rm -rf * |
| 20:22.08 | Penguin | Precisely, but with more comedy. |
| 20:22.33 | jmetro | i bet he tried to press control-z to undo it |
| 20:22.58 | Penguin | <PROTECTED> |
| 20:23.20 | jmetro | hah |
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| 20:38.56 | igcewieling | I'm using spool files to send faxes. Asterisk considers a spool call "OK" and deletes it if the outgoing call was answered, as expected. But sometimes faxes fail so I need the fax re-queued. Does anyone know of a way to force Asterisk to think a spool call failed when it didn't? I'm using Channel: Local/blah and using context, extension, and priority which runs an AGI |
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| 20:53.32 | ghenry | with dahdi show, how can you tell if there is an error on an FXO analgoue line? |
| 20:54.15 | ghenry | I have two in a g0 and the first line has an error so shouldn't the second one be used by dahdi for the outbound call? |
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| 21:17.58 | n8ideas | Anyone aware if there is an issue with periodic queue announcements in realtime? |
| 21:19.37 | pabelanger | n8ideas: like what? |
| 21:19.49 | n8ideas | well... they don't work |
| 21:20.01 | n8ideas | although not 100% sure I have the field names correct |
| 21:21.58 | pabelanger | well, check the field names |
| 21:22.00 | pabelanger | :) |
| 21:22.01 | n8ideas | periodic_announce and periodic_announce_frequency |
| 21:22.03 | n8ideas | ? |
| 21:22.08 | n8ideas | which match up to what I expect |
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| 21:28.02 | n8ideas | not sure where to find the actual field names other than in the *.sql files which seem to omit it |
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| 22:03.24 | g_r_eek | i am getting on asterisk install error: mysql/mysql.h: No such file or directory -- and there is no such directory |
| 22:03.54 | g_r_eek | any clues what it might be? |
| 22:05.43 | jpsharp | Missing some mysql development headers? |
| 22:07.59 | g_r_eek | maybe but which package to install on centos? |
| 22:08.34 | Qwell | what is failing to build? |
| 22:10.06 | g_r_eek | asterisk |
| 22:10.16 | Qwell | Be more specific. |
| 22:10.23 | jacekowski | g_r_eek: mysqlclient dev packages |
| 22:10.26 | jacekowski | and mysql dev packages |
| 22:10.32 | jacekowski | install them |
| 22:10.35 | g_r_eek | i messed up freepbx by make install asterisk again |
| 22:11.19 | g_r_eek | so i dl again and trying to reinstall asterisk, did make clean, .configure selected the add ons and gsm sound extra files |
| 22:11.22 | Qwell | Can you answer a single question properly? |
| 22:12.05 | g_r_eek | Qwell: does this help you: [CC] app_mysql.c -> app_mysql.o |
| 22:12.05 | g_r_eek | app_mysql.c:35:25: error: mysql/mysql.h: No such file or directory |
| 22:13.11 | g_r_eek | Qwell: after the make install i get http://pastebin.com/vdWjd5qy |
| 22:13.45 | Qwell | Do you need app_mysql? |
| 22:14.20 | n8ideas | you need mysql-devel |
| 22:14.24 | n8ideas | yum install mysql-devel |
| 22:14.28 | n8ideas | rerun configure |
| 22:14.45 | g_r_eek | frepbx needs it |
| 22:14.51 | Qwell | No it doesn't. |
| 22:15.10 | g_r_eek | n8ideas: i think thats it let me try |
| 22:15.15 | Qwell | How did you enable it? |
| 22:15.42 | g_r_eek | i did just enable the mysql cdr that fpbx says it needs |
| 22:15.52 | saint_ | I'l be back |
| 22:15.53 | g_r_eek | let me try the dev |
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| 22:20.43 | g_r_eek | …waiting for make |
| 22:20.53 | g_r_eek | Qwell: where r u from? |
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| 22:28.34 | g_r_eek | n8ideas: seems that it passed the point with app_mysql |
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| 22:37.17 | g_r_eek | Yes! installed and running Freepbx also all ok! Thanks a million: jacekowski Qwell n8ideas !!! |
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| 23:36.22 | KNERD | WHat's this all about in v 1.8.19? /usr/sbin/safe_asterisk: line 145: 26558 Illegal instruction nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} |
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