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| 01:25.00 | phix | hai hai! |
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| 05:37.57 | Micc_ | Is it normal for CDR(duration) to be 0 when used inside a hangup handler? |
| 05:47.53 | fling | [TK]D-Fender: hello :] |
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| 07:53.41 | bulkorok | hi |
| 08:05.03 | din3sh | my * box cannot read dtmf on DID (dahdi) from South Africa numbers but work fine with calls from other countries |
| 08:06.23 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 08:07.51 | WIMPy | So what do you think, where the problem would come from? |
| 08:10.15 | *** join/#asterisk slash213 (slash213@176.57.74.111) |
| 08:10.17 | wdoekes | Micc_: there is a flag for that in cdr.conf: endbeforehexten |
| 08:10.20 | slash213 | hi guys |
| 08:10.24 | slash213 | a little help please? |
| 08:10.35 | WIMPy | ~ask |
| 08:10.35 | 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. |
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| 08:12.08 | slash213 | i have set up a queue with strategy=linear and timeoutrestart=yes, but when a call gets into said queue, it goes through the agents for a single time and then drops out |
| 08:12.32 | slash213 | the intended way is leaving the call in the queue indefinitely until someone picks up |
| 08:12.57 | slash213 | what am i doing wrong? besides "asking if i can ask a question first" :) |
| 08:18.37 | TazzNZ | din3sh: what dahdi cards are you using ? |
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| 08:32.46 | wdoekes | slash213: do you have a timeout set in the Queue() call? what does ${QUEUESTATUS} say after Queue() (add a NoOp and look at the verbose output)? |
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| 08:33.42 | wdoekes | and what happens if you switch strategy to ringall? does it ring a single time or does it drop out then too? |
| 08:33.51 | wdoekes | s/or/and |
| 08:34.04 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 08:34.32 | slash213 | wdoekes, yes, i have timeout set to "10". |
| 08:34.34 | gavimobile | kaldemar: hey man you got a few? im not able to get this googletts.agi script to work with my pbx |
| 08:35.00 | wdoekes | slash213: so you tell it that users should be in the queue for at most 10 seconds? |
| 08:35.00 | slash213 | by "ring a single time" you mean that it rings for the "timeout" period, right? i'll look and get back with results |
| 08:35.09 | slash213 | wdoekes, eehhh, am i? |
| 08:35.19 | slash213 | isn't it "How long do we let the phone ring before we consider this a timeout"? |
| 08:35.35 | slash213 | so like "ring an agent for 10 seconds, then go to the next one"? |
| 08:35.50 | wdoekes | slash213: there's a difference between timeout= queues.conf option and timeout option in Queue() |
| 08:36.06 | slash213 | oh, i meant the queues.conf one |
| 08:36.44 | slash213 | i don't set anything in dialplan |
| 08:37.01 | wdoekes | well.. go look at the QUEUESTATUS |
| 08:37.03 | slash213 | it's just exten=> _X.,1,queue(qname) |
| 08:37.30 | gavimobile | here is my output http://pastebin.com/pdhPfe6k |
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| 08:51.38 | din3sh | my * box cannot read dtmf on DID (dahdi) from South Africa numbers but work fine with calls from other countries |
| 08:52.00 | din3sh | WIMPy: no idea where the problem might be |
| 08:52.25 | din3sh | TazzNZ:using TE121 digium cards |
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| 08:56.17 | gavimobile | how can I run an agi script from the cli? |
| 08:56.33 | gavimobile | I tried agi exec SIP googletts.agi && agi exec SIP/0000FFFF0000 googletts.agi |
| 08:56.46 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 09:01.59 | gavimobile | agi exec SIP/0000FFFF0000 "exec Dial(SIP/1234)" |
| 09:02.03 | WIMPy | din3sh: Obviousely somewhere on the way. Tell the users in South Africa not to use a telco that uses voip and can;t configure it propperly. |
| 09:02.13 | gavimobile | I have no idea how to diagnose this thing |
| 09:04.06 | *** join/#asterisk sawgood (~sawgood@173-13-158-29-sfba.hfc.comcastbusiness.net) |
| 09:07.37 | din3sh | users in S.A are the client's clients, cant impose anythg on em |
| 09:07.38 | din3sh | :o |
| 09:07.55 | din3sh | what does pulsedial=yes in chan_dahdi do? |
| 09:08.52 | *** join/#asterisk mmmamiens (3e402015@gateway/web/freenode/ip.62.64.32.21) |
| 09:08.58 | mmmamiens | Hello |
| 09:09.48 | mmmamiens | Can I send a call to external number, and if no answer, call back in the dialplan |
| 09:10.31 | sawgood | http://pastebin.com/62E4dm9v |
| 09:10.58 | sawgood | new Asterisk 1.8.20.1 on CentOS with registered SIP phone and a path to an ITSP (but calls fail) |
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| 09:11.22 | sawgood | Any tips on the CLI errors shown in the pastebin would be cool |
| 09:12.43 | mmmamiens | try : SIP/16508891224@Asterisk-Services ??? |
| 09:16.04 | sawgood | neat (got a 'loop detected error' and alot more stuff on the CLI |
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| 09:20.40 | sawgood | mmmamiens: tweak with your tip and things are working now (thank you) |
| 09:20.51 | mmmamiens | :) |
| 09:37.22 | slash213 | wdoekes, alright, it seems like user reported the original problem wrong. since i have no access to user hardware right i copied his setup and simulated queue exactly like his (hooray for pjsua) |
| 09:37.59 | slash213 | it rings up all four agents - for 10 seconds each, then goes to first one, then to second one and then unexpectedly drops the call |
| 09:38.26 | slash213 | no ${QUEUESTATUS}, just [Jan 28 13:35:05] VERBOSE[17178] pbx.c: == Spawn extension (from-pstn, 4995049150, 1) exited non-zero on 'DAHDI/i9/4957870887-12' |
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| 09:39.45 | slash213 | and for some reason it does not play hold music when call is from numbers in '495' area code. but plays music if the call is from '499'. i, i have no words. nothing. |
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| 09:57.39 | fling | 740/740 10.0.0.34 D 5060 OK (25729 ms) |
| 10:01.36 | realyneedhelp | hello everyone... im having a problem with our fax to mail service. the faxes that come in on the extension "72 22" are converted to pdf and then mailed to info@mydomain. the last email received this way, was on the 19.12.12, and we didnt change anything on that date or since then. |
| 10:03.27 | realyneedhelp | heres whats in our extension.conf |
| 10:03.28 | realyneedhelp | http://pastebin.com/Mn05PECw |
| 10:04.22 | realyneedhelp | if i manually execute the command: "mutt -d2 -s "mycompany AG - FAX EINGETROFFEN" xxx@xxxx.ch < /etc/asterisk/fax_email_notify.txt" it works |
| 10:04.39 | wdoekes | slash213: do you Answer() before Queue() ? |
| 10:05.35 | slash213 | wdoekes, i just tried it and the call no longer drops, thank you. can you explain me why is this important? |
| 10:05.36 | realyneedhelp | i watched the incoming fay on the asterisk CLI and this is what i got: http://pastebin.com/qyqYV4ZF |
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| 10:05.46 | realyneedhelp | *fax |
| 10:05.50 | wdoekes | slash213: if you don't answer, the call is still in Ringing state |
| 10:05.57 | wdoekes | you cannot ring indefinately |
| 10:06.17 | slash213 | wdoekes, is this hardcoded into asterisk? |
| 10:06.39 | wdoekes | pstn and sip works that way |
| 10:06.49 | realyneedhelp | SO... the Fax comes in, gets converted to pdf and is stored in the folder, but is not sent out per mail.... any ideas what causing this? |
| 10:07.24 | slash213 | oh, alright. thank you very much for your help |
| 10:08.11 | wdoekes | as for the musiconhold.. it could be that you can play premature (during ringing) audio (progress/183) to certain callers only |
| 10:09.25 | realyneedhelp | when i check the debug file that mutt created in the root directory i see this: mutt_write_fcc(): unable to open mailbox /sent in append-mode, aborting. |
| 10:09.34 | realyneedhelp | any idea what that means? |
| 10:09.42 | wdoekes | realyneedhelp: su to the asterisk user |
| 10:09.48 | wdoekes | and then run the mutt command |
| 10:11.01 | slash213 | wdoekes, is that defined explicitly? because every moh-related config is vanilla here, i haven't changed a single thing. but, well, it's no problem now, the queue-moh was bugging out, playback() after answer() works |
| 10:11.39 | wdoekes | slash213: whether premature audio works or not, depends on your itsp |
| 10:12.06 | wdoekes | so it could be that it depends on their upstream as well.. leading to different behaviour for different regions |
| 10:12.54 | wdoekes | after answer, it always works, because it's not premature anymore |
| 10:13.16 | slash213 | wdoekes, that was not the case, because i made the calls from my own box - but music behaved differently based on src_number |
| 10:13.35 | wdoekes | that I do not believe |
| 10:13.38 | *** join/#asterisk vlad_sta_ (~vlad_star@178.176.227.76) |
| 10:14.45 | realyneedhelp | well as far as i can tell, asterisk is running as root |
| 10:14.47 | realyneedhelp | an i am root |
| 10:14.58 | realyneedhelp | and the command works when i use it manually |
| 10:15.05 | realyneedhelp | but not from the extension file |
| 10:15.09 | realyneedhelp | those that make sense? |
| 10:15.29 | wdoekes | realyneedhelp: in that case I'm guessing env variables |
| 10:15.35 | wdoekes | e.g. an empty $HOME |
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| 10:16.05 | wdoekes | realyneedhelp: check mutt help to see if there is a mailbox/home folder you can pass to it |
| 10:16.29 | realyneedhelp | i looked at with another guy and he mentioned something like that too.... how would i fix this? can i set it in the extension file or something |
| 10:16.55 | wdoekes | you could do "HOME=/root mutt -d2 ..." |
| 10:17.09 | wdoekes | but it might be a different var.. I don't know |
| 10:18.05 | wdoekes | or you could strace(1) it and check the output for write/open failures |
| 10:18.42 | realyneedhelp | hen i check the debug file that mutt created in the root directory i see this: mutt_write_fcc(): unable to open mailbox /sent in append-mode, aborting. |
| 10:18.50 | realyneedhelp | is that relevant? |
| 10:23.06 | wdoekes | I saw that |
| 10:23.10 | wdoekes | relevant? probably |
| 10:24.25 | wdoekes | this is #asterisk, not #mutt. use strace(1) or some other way to stab mutt into telling you what's the problem exactly |
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| 10:37.56 | *** join/#asterisk PbxMan (c335d968@gateway/web/freenode/ip.195.53.217.104) |
| 10:38.49 | PbxMan | morning |
| 10:40.48 | fling | what do I need to build a simple click-to-call feature? |
| 10:41.04 | fling | for example to call from browser |
| 10:41.17 | fling | do I need some firefox plugin or what? |
| 10:41.36 | wdoekes | fling: you need a way to interface with asterisk: call files, ami, sip (refer) |
| 10:42.11 | fling | wdoekes: I have sip, do not have ima atm |
| 10:43.42 | wdoekes | if you use sip, you won't need to do anything on the asterisk side. but it might be more work to have a firefox plugin speak sip than to have it speak http |
| 10:45.14 | fling | wdoekes: so do I need to setup ami on a server? |
| 10:48.35 | din3sh | anyone got a cisco 6945 to work with asterisk |
| 10:49.39 | din3sh | ? |
| 10:50.10 | fling | wdoekes: what is the simpliest solution? |
| 10:52.46 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 10:56.34 | din3sh | fling: google it man, lots of example out ther |
| 10:56.37 | din3sh | there* |
| 10:56.38 | din3sh | http://www.netsapiens.com/2012/05/08/implement-click-to-call-web-page/ |
| 10:56.47 | Chainsaw | Anyone got a GPO 746 working with Asterisk? |
| 11:00.52 | realyneedhelp | well.... adding HOME=/root in front of "mutt -s ...." solved the problem..... but this is just a work around.... does anybody have an idea what caused this to happen. (we didnt change anyhting and on te 19.12.12 it stopped sending our emails) cause this seems like a pretty seriuos problem |
| 11:01.16 | realyneedhelp | if anybody has ANY idea, id love to hear it |
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| 11:05.56 | danfromuk | Hi, is there anything wrong with setting qualify to 3000 for a user with poor latency? will it cause any issues? |
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| 11:08.34 | fling | din3sh: thanks. so it may be included into crm php code |
| 11:08.42 | fling | din3sh: I will talk to my programmers :P |
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| 11:14.13 | srp_ | WIMPy: Hi, I have installed asterisk-11.2.1 on ubuntu-12.04 (with webRTC support). I followed the asterisk definitive guide book that you referred to me yesterday! calls between two SIP clients are working fine. But, when i try any application like Playback() in the dial plan, the sound is not heard. Is there anything I can check for? |
| 11:15.00 | WIMPy | Are the files found? |
| 11:15.27 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 11:16.13 | srp_ | yes! here is the output on asterisk console when i dialed the extension (http://paste.kde.org/658124) |
| 11:18.15 | srp_ | WIMPy: here is my extensions.conf (http://paste.kde.org/658130) and my asterisk console output here (http://paste.kde.org/658124) |
| 11:20.53 | srp_ | WIMPy: is there any way to test the GSM files by playing on any audio player? I tried on VLC an i can hear nothing there as well.. all other sounds in mp3 and other formats can be heard from the system.. |
| 11:21.34 | WIMPy | depending on your verison, sox may play them. |
| 11:23.52 | srp_ | WIMPy: i can hear them when i play using sox, but not in asterisk. |
| 11:24.46 | WIMPy | Must be a codec or strand networking issue. |
| 11:28.32 | srp_ | WIMPy: hmm.. okay. let me check that |
| 11:29.36 | WIMPy | s/rand/range/ |
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| 11:31.54 | srp_ | WIMPy: how do i resolve it? |
| 11:33.02 | srp_ | WIMPy: and moreover if you see my asterisk console log i shared. it says playing sample.slin not sample.sln. why is that? |
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| 11:44.48 | srp_ | ChannelZ: any ideas about my problems i posted a few minutes back? |
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| 12:19.43 | tzafrir_laptop | Anybody here has tor2 card and willing to test a patch? (separate out the firmware)? |
| 12:24.06 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
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| 12:35.48 | coppice | I wonder how many tor2 and tor3 cards were made, amongst all the various makers? |
| 12:40.28 | WIMPy | One of my clients has a tor3. |
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| 12:43.09 | *** join/#asterisk last1 (~last1@109.100.28.28) |
| 12:45.16 | last1 | I removed b |
| 12:45.31 | last1 | I removed my astdb file by mistake and now all calls terminate as 'unavailable' |
| 12:45.38 | last1 | how can I recreate it ? |
| 12:45.52 | WIMPy | Restart asterisk |
| 12:45.59 | last1 | I did |
| 12:46.09 | last1 | now the file is 8kb |
| 12:46.18 | WIMPy | But it seems like you're using something that expects some contents in there. |
| 12:46.20 | *** join/#asterisk rokjan (~jj2@static-190-181-29-206.acelerate.net) |
| 12:46.35 | last1 | yeah, this is trixbox |
| 12:46.52 | WIMPy | then you need to ask in #trixbox |
| 12:48.49 | last1 | thank you |
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| 13:35.16 | fling | which codec do I need for 1kbps? :p |
| 13:35.55 | WIMPy | What about morse? |
| 13:36.10 | dpilon | hahaha |
| 13:41.07 | fling | WIMPy: – ––– ·–· ··· · –·–· ––– –·· · |
| 13:42.00 | *** join/#asterisk TarCert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 13:42.22 | [TK]D-Fender | VoSS |
| 13:42.23 | Greenlight | considers if it's going to be interesting enough to Google and translate |
| 13:42.33 | [TK]D-Fender | or VoCP |
| 13:43.17 | fling | [TK]D-Fender: I'm trying to call a girl with very slow 3g connection |
| 13:43.54 | dpilon | just dial her cell phone :) |
| 13:44.13 | fling | dpilon: already dialed twice |
| 13:44.31 | fling | she is in another town, I want to call over sip |
| 13:44.54 | fling | she has ekiga connected over vpn to my server |
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| 14:03.19 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:04.56 | fling | http://ic.pics.livejournal.com/gosnorkokontrol/16477826/23710/23710_original.png |
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| 14:05.59 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:11.02 | fling | [TK]D-Fender: where are 4kHz codecs? :p |
| 14:12.20 | [TK]D-Fender | LPC10 <- |
| 14:12.27 | [TK]D-Fender | Domo arigato .... |
| 14:14.16 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 14:15.08 | Penguin | fling: torse code? I think you missed a bit. |
| 14:15.24 | jmetro | keyboard upside down |
| 14:15.25 | fling | Penguin: umm? |
| 14:15.40 | fling | pɹɐoqʎǝʞ |
| 14:17.51 | *** join/#asterisk Kalidarn (~Kalidarn@unaffiliated/kalidarn) |
| 14:18.03 | Kalidarn | is it possible to get BETTER_BACKTRACES on FreeBSD 9.1? |
| 14:18.17 | Kalidarn | I did the usual ./configure and make menuselect |
| 14:18.19 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 14:18.29 | Kalidarn | i only saw DONT_OPTIMIZE and DEBUG_THREADS |
| 14:19.06 | fling | [TK]D-Fender: now I need to enable it in ekiga somehow… |
| 14:19.20 | fling | [TK]D-Fender: the codec looks great :p |
| 14:19.35 | fling | [TK]D-Fender: but sounds like 2.4kbps :D |
| 14:20.12 | Kalidarn | BETTER_BACKTRACES seems to have an XXX next to it |
| 14:21.41 | Kalidarn | hmm probably because I don't have libbfd |
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| 14:24.02 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:25.15 | Kalidarn | I guess I'll have to sort out why when installing devel/libbfd ports gives me this error: libbfd-2.19.1 conflicts with installed package(s): binutils-2.23.1 first. |
| 14:30.27 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:30.48 | gavimobile | who should be the owner of the .conf files in /etc/asterisk? the asterisk user or root? |
| 14:32.41 | Greenlight | asterisk |
| 14:37.10 | Kalidarn | interesting I think i might have found an asterisk crashl |
| 14:44.38 | *** join/#asterisk Minotaur01 (~minotaur0@24.102.53.84) |
| 14:45.01 | Kalidarn | im wondering if this is my fault |
| 14:45.39 | Kalidarn | see i tried making a port /usr/port/net/asterisk that had DONT_OPTIMIZE and DEBUG_THREADS enabled, and I found when starting the asterisk service it would coredump |
| 14:45.49 | Kalidarn | but i checked out the error log and found that it would repeat the same error over and over. |
| 14:46.00 | Kalidarn | so i decided to download the tarball and do it manually off the asterisk site |
| 14:46.10 | Kalidarn | and i'm getting the same thing. |
| 14:46.24 | Kalidarn | http://pastebin.com/5Uferdm2 |
| 14:47.24 | Kalidarn | using asterisk asterisk-1.8.20.1 |
| 14:47.45 | Kalidarn | the error that seems to be repeated is: |
| 14:47.47 | Kalidarn | logger.c line 1072 (logger_thread): mutex '&logmsgs.lock' freed more times than we've locked! |
| 14:47.49 | Kalidarn | logger.c line 1072 (logger_thread): Error waiting on condition mutex 'Invalid argument' |
| 14:48.06 | Kalidarn | i didn't want to file a bug report if i had done something wrong |
| 14:48.41 | Kalidarn | gavimobile: you should have an asterisk user |
| 14:48.46 | Kalidarn | one that can't login |
| 14:48.52 | Kalidarn | and run them as that user |
| 14:49.09 | *** join/#asterisk Tarcert (c1a9b80e@gateway/web/freenode/ip.193.169.184.14) |
| 14:51.26 | tzafrir_laptop | No real need for Asterisk to be the owner of files under /etc/asterisk . Asterisk needs to write voicemail.conf , but that's about it. |
| 14:52.11 | gavimobile | tzafrir_laptop: got it! |
| 14:52.12 | tzafrir_laptop | Asterisk does need to be able to read them, and ideally other users should not (Only reason for that is that a number of those files contain passwords) |
| 14:53.17 | gavimobile | I need some trying to get googletts.agi script to work. [TK]D-Fender gave me some ideas which were out of my ability to do |
| 14:53.27 | gavimobile | I need some help* |
| 14:53.49 | gavimobile | I've been fiddling with this for the last 24 hours |
| 14:55.50 | mjordan | Kalidarn: the crash is occurring when attempting to get the backtrace information. Most likely this is because, as you previously noted, you had conflicts with libbfd on your distro. I'm not sure how you get it installed, but something about the situation has gone horribly awry. If Asterisk calls into libexecinfo's backtrace function and the whole thing blows up in its face, there isn't much it can do. |
| 14:56.09 | Kalidarn | i've not got it installed :S |
| 14:56.24 | mjordan | And yet, it crashes |
| 14:56.26 | Kalidarn | that's the point when i tried to though it said it was in the binutils port |
| 14:56.33 | Kalidarn | also i didn't enable BETTER_BACKTRACES |
| 14:56.36 | Kalidarn | so it shouldn't matter |
| 14:57.11 | mjordan | you're on freebsd? |
| 14:57.15 | Kalidarn | yes 9.1 |
| 14:57.30 | mjordan | This does not fill me with confidence: "This is a quick-n-dirty BSD licensed clone of backtrace facility found in the GNU libc, mainly intended for porting linuxish code to BSD platforms, however it can be used at any platform which has a gcc compiler." |
| 14:57.35 | mjordan | http://www.freshports.org/devel/libexecinfo |
| 14:57.41 | mjordan | your mileage is probably going to vary. |
| 14:58.35 | Kalidarn | i don' tthink that's the right port |
| 14:58.48 | Kalidarn | http://www.freshports.org/devel/libbfd |
| 14:58.49 | *** join/#asterisk asr33 (~asr33@unaffiliated/asr33) |
| 14:59.23 | Kalidarn | i didn't actually have devel/libexecinfo installed |
| 14:59.36 | Kalidarn | oh wait yes i did :P |
| 14:59.48 | Kalidarn | hmm |
| 15:00.51 | *** join/#asterisk vlad_starkov (~vlad_star@176.74.9.254) |
| 15:01.01 | Kalidarn | mjordan: so, what should I do, post a bug on the asterisk tracker? |
| 15:01.19 | Kalidarn | because Digium does say they support freebsd do they not> |
| 15:01.22 | mjordan | you can, but it is highly unlikely that anyone will work it without a patch. |
| 15:01.23 | mjordan | No |
| 15:01.39 | Kalidarn | ah, and it's beyond my skills to patch it :P |
| 15:01.53 | Kalidarn | but someone at freebsd.org might know |
| 15:02.01 | mjordan | It is most likely a crash in the BSD library, which means patching Asterisk isn't the right approach anyway |
| 15:02.09 | Kalidarn | yeah. |
| 15:02.32 | Kalidarn | but as they don't offer a freebsd port that has debugging :P can't really post it there. |
| 15:02.38 | Kalidarn | well i guess i can |
| 15:03.23 | Kalidarn | mjordan: ill ask the asterisk port maintainer what I should do. |
| 15:03.40 | Kalidarn | considering it happens with both upstream tarball and the port. |
| 15:03.58 | Kalidarn | i might also try asterisk 1.10 while im at it and see if i can reproduce it. |
| 15:04.18 | mjordan | it should be consistent. The DEBUG_THREADS code is the same across the various versions of Asterisk |
| 15:04.30 | Kalidarn | oh okay then, no point then i guess. |
| 15:05.03 | Kalidarn | anyway mjordan thanks for answering my questions. |
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| 15:18.22 | mjordan | Kalidarn: np, good luck |
| 15:19.19 | Kalidarn | hopefully we come to a workable solution, though i would be interested to know how digium does any testing |
| 15:19.22 | Kalidarn | if they can't backtrace |
| 15:19.30 | Kalidarn | i guess they only test on linux :( |
| 15:19.56 | mjordan | Correct. Our build agents run various flavors of Linux (CentOS being the primary ones) |
| 15:20.04 | Kalidarn | yeah no doubt. |
| 15:20.24 | Kalidarn | unfortunately i very much like zfs :) and that's only very recently been re-implemented to linux. |
| 15:20.33 | mjordan | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Test+Suite+Documentation |
| 15:20.44 | Kalidarn | because the sun implementation was under CDDL which is incompatible with shitty GPL |
| 15:21.07 | Kalidarn | though i guess testing on other platforms other than linux is good anyway |
| 15:21.26 | Kalidarn | it's could cause certain odd bugs that don't occur on linux to be found. |
| 15:21.42 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:21.43 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:21.47 | Kalidarn | and just because they don't occur doesn't mean they are there. |
| 15:21.49 | mjordan | it's always a good idea, but we have to focus on bang for the buck. There are a number of people who find things on Solaris/Mac/BSD-variants and provide patches to fix them, which is always appreciated |
| 15:22.02 | Kalidarn | yeah |
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| 15:37.09 | *** join/#asterisk Kalidarn (~Kalidarn@unaffiliated/kalidarn) |
| 15:37.31 | Kalidarn | oh mjordan one other question, what is debian support like? |
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| 15:39.14 | slax0r | Hello, recently I've noticed in my log files that someone has been trying to call through my pbx some weird long numbers, from even weirder IPs, like this: chan_sip.c: Call from '' (37.8.116.106:10932) to extension '00000972592910519' rejected because extension not found in context 'default'. How can I limit that only authenticated users can place calls? |
| 15:39.27 | [TK]D-Fender | kaldemar, They maintain their own packages. Source works fine. |
| 15:39.45 | Kalidarn | [TK]D-Fender: wrong person ;) |
| 15:39.49 | [TK]D-Fender | slash213, "allowguest=no" |
| 15:40.11 | [TK]D-Fender | Kalidarn, Good. You are paying attention. |
| 15:40.22 | [TK]D-Fender | I hate X-Chat auto-complete.... |
| 15:40.33 | jmetro | pidgin |
| 15:40.34 | Kalidarn | weechat ftw :D |
| 15:41.17 | slax0r | spose that was meant for me [TK]D-Fender ? |
| 15:42.26 | *** join/#asterisk echs (~echs@unaffiliated/echs) |
| 15:45.50 | Kalidarn | slax0r: looks like someone in Palestine wants to call you |
| 15:46.29 | slax0r | not likely |
| 15:46.41 | slax0r | they're trying to make calls out of my pbx |
| 15:47.17 | mjordan | Kalidarn: [TK]D-Fender is correct :-) |
| 15:50.39 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 15:54.28 | fling | what softphone do I need for lpc10? |
| 15:54.51 | fling | [TK]D-Fender: enabled lpc10 in opal, but ekiga does not support it |
| 16:01.34 | danfromuk | slax0r: if it interests you, we saw the same thing. Since the IP was from palastine, we just blocked the whole of palastine on our firewall since we don't have any clients there anyway. |
| 16:02.07 | jmetro | Country banning like a boss. |
| 16:02.23 | danfromuk | Ensure your default context has no way of dialling out. |
| 16:03.40 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 16:03.45 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
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| 16:07.19 | jmetro | Funny, my server is getting hit from palestine too |
| 16:07.39 | Greenlight | We had a server hacked from there once |
| 16:07.51 | *** join/#asterisk navaismo (~navaismo@189.241.23.9) |
| 16:08.03 | Greenlight | Figured the local police wouldn't be too interested so never even reported the IP ^^ |
| 16:08.28 | jmetro | Some guy in palestine is really desparate to call his mom in israel, all his calls go to 972 |
| 16:08.30 | danfromuk | Greenlight: same here. |
| 16:09.12 | danfromuk | 972 is also Israel. Maybe someones trying to find a server they can use to flood israel and palastine with calls. |
| 16:10.50 | jmetro | hm.. 00972 543 469 769 |
| 16:11.24 | Greenlight | Just fix the firewall and put it down to experience that's all you can do |
| 16:12.14 | danfromuk | We've got fraud/hacking systems in place to ensure accounts are shut down if they go above a limit. |
| 16:12.48 | danfromuk | Luckily, not had any issues of hacking etc... Only clients' servers being hacked. |
| 16:13.13 | Greenlight | Yea it's when clients are allowed to config the firewalls themselves when we have issues |
| 16:14.13 | danfromuk | I love country blocking. Its not perfect but its a great feeling! Since we implemented that, we've seen a total drop of ssh attempts and asterisk attacks. |
| 16:14.28 | Greenlight | Yea, we do the same at our gateway. UK IP's only |
| 16:14.33 | danfromuk | In fact, ssh is fully locked down now. |
| 16:14.38 | Greenlight | Almost 100% drop in hacking attempts |
| 16:15.33 | Greenlight | Guess all the UK hackers go over a VPN :) |
| 16:15.40 | Greenlight | Probably based in Israel :) |
| 16:15.49 | gavimobile | what? |
| 16:16.02 | gavimobile | the hackers are from israel? |
| 16:16.18 | WIMPy | As if anyone could know what country an IP is used in. |
| 16:16.32 | gavimobile | bogus |
| 16:16.42 | Greenlight | You can know roughly there's it's *allocated* |
| 16:16.45 | Greenlight | *where |
| 16:17.04 | danfromuk | Does israel have a different ip range to palastine? |
| 16:17.12 | danfromuk | ranges* |
| 16:17.15 | WIMPy | Yes, but with multi-national carriers that doesn't mean much. |
| 16:17.35 | danfromuk | Very true but its a start. |
| 16:17.50 | danfromuk | Obviously it means nothing when you have VPNs around |
| 16:17.53 | Greenlight | Ofcourse, it's not infallible, nothing at that level is |
| 16:18.14 | Greenlight | But it's enough to drop hacking attempts by 100% so it's worth doign! |
| 16:18.16 | Greenlight | *doing |
| 16:18.25 | danfromuk | Agreed |
| 16:18.30 | WIMPy | It just gives a false feeling of security. |
| 16:18.45 | gavimobile | danfromuk: yes. and I blocked all of palestine cause they were attacking me |
| 16:18.50 | Greenlight | No, we still lock down to IP when we can, and use fail2ban on top of that |
| 16:19.14 | Greenlight | You could even open up at an ISP level |
| 16:19.25 | danfromuk | I get mixed results with fail2ban. I dont really trust it but i use it |
| 16:20.00 | Greenlight | Only time it goes off now is if a customer puts in a wrong password |
| 16:21.07 | danfromuk | We had to implement our own firewall device because our datacentre's firewall used connection tracking so even when we added a new rule, it didn't affect existing connections. |
| 16:21.31 | danfromuk | Since then, we've been able to do things like both countries. Made a huge difference! |
| 16:23.10 | danfromuk | WIMPy: it means that the average browser can't come across access to our servers. For example if a client has stupidly posted a sip.conf file containing passwords, the hacker would have to realise that they need to spoof their address to a country we allow. |
| 16:25.00 | Greenlight | Indeed - I don't think it gives a false sense of security if you know what's actually being done, and are mindful that it's not complete protection. You can never be 100% secure with these things, and still provide the service that customers require. |
| 16:26.02 | danfromuk | I'm not looking forward to a DoS attack. However our datacentre seem quite good when it comes to those. |
| 16:26.18 | WIMPy | It doesn't give any security at all. It just gives you less messages to read. |
| 16:27.15 | Greenlight | So far it's prevented *ALL* hacking attempts. We went from 10-20 a day to NOTHING. To say that's not providing *any* security is nosense. |
| 16:27.19 | danfromuk | WIMPy: thats a start! |
| 16:27.56 | WIMPy | Greenlight: No. It's true. |
| 16:29.45 | danfromuk | WIMPy is correct since someone could use a VPN to reach an IP range that you allow. |
| 16:30.02 | Greenlight | I never said it provided "complete security" |
| 16:30.04 | danfromuk | Its interesting that they don't though. |
| 16:30.10 | Greenlight | Nothing does that while meeting customer needs |
| 16:30.52 | danfromuk | Very true. |
| 16:31.07 | danfromuk | Reminds me of google analytics advert i saw. http://www.youtube.com/watch?v=3Sk7cOqB9Dk |
| 16:31.14 | Greenlight | Opening yourself to hacking attempts when you don't need to is just silly. If WIMPy chooses to do that out of a matter of principle then that's fine, but I choose not to. |
| 16:31.15 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 16:31.20 | danfromuk | Skip to about 1 minute |
| 16:31.20 | saint_ | hi all |
| 16:31.39 | saint_ | is asteriskdocs.org down .. |
| 16:31.40 | saint_ | ? |
| 16:31.51 | leifmadsen | yes. |
| 16:31.54 | leifmadsen | I'll talk to the provider shortly |
| 16:32.09 | leifmadsen | file: ^^^ :) |
| 16:32.15 | *** part/#asterisk echs (~echs@unaffiliated/echs) |
| 16:32.19 | ChannelZ | for a few days.. |
| 16:32.21 | file | what? hi |
| 16:32.39 | saint_ | can someone tell me how can i test with ${IF how to check that a value entered with READ is between 1 and 9 ? |
| 16:32.50 | WIMPy | Greenlight: It's just like sitting in an location that's expose in all directions and build a wall in the direction the mosquitos seem to come from. |
| 16:33.11 | Greenlight | Or more like using mosquito cream |
| 16:33.41 | WIMPy | ... on only one spot. |
| 16:33.59 | Greenlight | On everything but one spot |
| 16:34.14 | Greenlight | You're going to get bitten less, if at all |
| 16:34.27 | WIMPy | You should take a look at the globe. |
| 16:34.28 | Greenlight | wonders how far we can stretch this analogy |
| 16:34.34 | WIMPy | indeed. |
| 16:34.48 | file | it went down all of 4 minutes ago lol |
| 16:36.29 | file | better? |
| 16:40.24 | [TK]D-Fender | saint_, https://wiki.asterisk.org/wiki/display/AST/Expressions |
| 16:54.01 | *** join/#asterisk din3sh (~din3sh@41.136.80.96) |
| 16:54.30 | din3sh | how to find out the route/path taken by voice traffic? |
| 17:01.35 | Greenlight | That's really down to your network setup. |
| 17:02.11 | Greenlight | Usually it would be the same path as other traffic, so a traceroute or mtr would show you |
| 17:08.27 | j4m3s | I'm using asterisk 1.4 and have a peer who's expire time is set to '-1' shouldn't it be set to the defaultexpiry setting? |
| 17:08.45 | Qwell | ~upgrade asterisk |
| 17:08.45 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 17:09.35 | j4m3s | Qwell, it's being used with vicidial and the latest version they support is 1.8. |
| 17:09.40 | Qwell | sucks |
| 17:09.41 | Qwell | ~upgrade asterisk |
| 17:09.41 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 17:09.48 | jmetro | lol |
| 17:10.35 | j4m3s | Qwell, so peer expiry is not impacted by the defaultexpiry setting in 1.4/1.8? |
| 17:10.38 | din3sh | IVR/dtmf from South africa is not working arriving on a DID in my asterisk box (E1 TE121 card) |
| 17:12.51 | saint_ | can someone help me with http://pastebin.com/rubBUmuS |
| 17:13.02 | saint_ | I'm trying to have the end user enter an ETA from 0 to 9 (0 being 10 minutes) |
| 17:13.11 | saint_ | I want to exclude * and # |
| 17:13.36 | saint_ | as of now, with the code in here http://pastebin.com/rubBUmuS , if I enter anything else than 0 , then I get ETA="" instead of the digit entered. Can someone explain me why ? |
| 17:15.04 | *** join/#asterisk igcewieling (~igcewieli@user-24-214-153-32.knology.net) |
| 17:26.15 | [TK]D-Fender | saint_, show us the failure |
| 17:26.40 | [TK]D-Fender | <j4m3s> Qwell, it's being used with vicidial and the latest version they support is 1.8. <- so upgrade to it |
| 17:31.02 | Qwell | NEW_ETA will never be both >= 1 and <= 9, AND 0 |
| 17:31.49 | Qwell | saint_: On line 11, what happens if NEW_ETA is 3? |
| 17:33.10 | [TK]D-Fender | saint_, same => n,Set(ETA="") <-- never ever set to quotes like that |
| 17:33.19 | [TK]D-Fender | saint_, they are literal |
| 17:33.37 | saint_ | [TK]D-Fender: ok. i m trying something else. hold on. |
| 17:33.52 | saint_ | [TK]D-Fender: but how would you setup a variable to null ? |
| 17:34.00 | j4m3s | [TK]D-Fender, i have suggested that to my client, but will that provide better dns srv support? |
| 17:34.30 | j4m3s | [TK]D-Fender, the issue they are experiencing is the registration never times out |
| 17:35.47 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.168.57) |
| 17:35.51 | igcewieling | saint_: Set(MyVar=) |
| 17:36.03 | igcewieling | You are setting ETA to be two quotes. |
| 17:36.15 | saint_ | [TK]D-Fender: I replaced my code with this, and it s working . http://pastebin.com/qi3itPuE |
| 17:36.18 | Qwell | igcewieling: valid, but not the reason for the failure. |
| 17:41.44 | igcewieling | j4m3s: Asterisk's SRV support is primitive at best for outgoing Dials |
| 17:44.06 | j4m3s | igcewieling, even in 1.8? |
| 17:44.47 | igcewieling | j4m3s: Yes, though in 1.8 you can compensate by writing dialplan to handle weights and failover |
| 17:46.06 | igcewieling | I did this recently (in AEL) though I think there are some bugs in my AEL code still. |
| 17:46.38 | gavimobile | nano #perl |
| 17:46.42 | gavimobile | lol |
| 17:46.54 | gavimobile | */join |
| 17:47.38 | igcewieling | http://pastebin.ca/2308172 |
| 17:48.21 | j4m3s | igcewieling, looking over that now thanks! |
| 17:49.04 | igcewieling | 1.8 apparently sorts the records by priority so the priority sorting stuff is likely redundant |
| 17:49.29 | TazzNZ | din3sh: you around ? |
| 17:49.53 | saint_ | if I write this test: Set(ETA=${IF($["${ETA}" = "" | "${ETA}" = "*" | "${ETA}" = "#"]?5:)}) will the value ETA be setup to null if I do not put anything after the : , or will it be left as whatever it is ? |
| 17:50.31 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 17:51.22 | kaldemar | saint_: set to an empty value |
| 17:51.43 | saint_ | kaldemar: so is there a way to do a simple if , that will not touch the value ? |
| 17:52.08 | kaldemar | saint_: ExecIf |
| 17:52.41 | j4m3s | igcewieling, i'm going to have to get this customer to update before i can start trying this out |
| 17:53.02 | kaldemar | or use your Set and put ${ETA} after : |
| 17:54.23 | saint_ | kaldemar: i m trying to change something here: http://pastebin.com/gBdcYtM2 |
| 17:54.28 | *** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net) |
| 17:54.33 | igcewieling | j4m3s: you could write an AGI to do the SRV lookups, weights priorities, etc. That should work for most any version of Asterisk |
| 17:54.40 | saint_ | I want to keep the test with the = "" |
| 17:54.52 | j4m3s | igcewieling, yuck |
| 17:54.55 | saint_ | but I want to goto to CallerNewETA if he pushes * |
| 17:54.55 | igcewieling | the SRV RFC has info on how to calculate the weights |
| 17:55.03 | saint_ | and WhatsMyETA if he pushes # |
| 17:56.46 | igcewieling | j4m3s: Asterisk's SRV supports (and CONNECTEDLINE and a few other things) are what I call Blue Balls Features. They get you all excited by the possibilities but for whatever reason you can't quite get what you want. |
| 17:57.06 | Qwell | igcewieling: patches welcome. |
| 17:57.21 | igcewieling | Qwell: I am not qualified for that. |
| 17:57.35 | saint_ | kaldemar: so I can do ExecIf($["${ETA}" = "*"])?Goto(xxx,y,z) ..? |
| 17:58.12 | Qwell | There are plenty of other ways to contribute. Complaining about the lack of a feature isn't one of those ways. |
| 17:58.45 | *** join/#asterisk AnGrYfUrBy (~MrX@pdpc/supporter/active/angryfurby) |
| 17:59.07 | igcewieling | Qwell: Not really a complaint, just a statement of fact. I either don't use those sorts of features or I find a work around. I already have permission to release my AEL SRV stuff and will do so when it has had more testing. |
| 17:59.39 | AnGrYfUrBy | can anyone help me install asterisk on my windows 7 box |
| 17:59.41 | *** join/#asterisk qakhan (~qakhan@50-200-52-14-static.hfc.comcastbusiness.net) |
| 17:59.49 | j4m3s | igcewieling, i will def give you some feedback when this customer pays up |
| 17:59.53 | kaldemar | saint_: there is also GotoIf |
| 18:00.13 | igcewieling | j4m3s: Any feedback is appreciated. |
| 18:00.17 | saint_ | kaldemar: yes but you told me that if I put an empty : then it set my variable to null |
| 18:00.39 | igcewieling | saint_: technically it sets it to empty |
| 18:00.40 | qakhan | hi all, i have a problem asterisk taking 8 sec to accept dtmf |
| 18:00.40 | kaldemar | saint_: in a set it will. |
| 18:01.13 | gavimobile | GOT ITTTT |
| 18:01.44 | gavimobile | [TK]D-Fender: my system didn't have a package called which |
| 18:01.59 | gavimobile | [TK]D-Fender: after I installed it, my googletts.agi is working! |
| 18:02.24 | kaldemar | saint_: only an application that is used to set a variable does that. gotoif and execif are not one of those. |
| 18:02.39 | qakhan | my dialplan is here |
| 18:02.40 | qakhan | http://pastebin.com/w0Bs0ZBy |
| 18:03.07 | kaldemar | saint_: and i said empty, not null. |
| 18:03.17 | [TK]D-Fender | qakhan, Show the call |
| 18:03.36 | Dovid | hi all |
| 18:03.40 | Dovid | been a while..... |
| 18:08.10 | saint_ | kaldemar: so would this be OK ? same => n,ExecIf($["${ETA}" = "#"]?Goto(WhatsMyETA,s,1)) |
| 18:09.17 | saint_ | does not seem to like it. |
| 18:09.31 | saint_ | there is no error, but if i press # it does not go to the WhatsMyETA part |
| 18:09.56 | gavimobile | hey Dovid |
| 18:10.08 | gavimobile | we miss you in #asterisk-il |
| 18:10.59 | WIMPy | What is -il? |
| 18:11.10 | [TK]D-Fender | Italian |
| 18:11.26 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:11.30 | gavimobile | [TK]D-Fender: israel |
| 18:11.41 | gavimobile | it is italian |
| 18:11.52 | gavimobile | [TK]D-Fender: it is not the norm that I have to correct ya ;-) |
| 18:12.04 | qakhan | [TK]D-Fender here http://pastebin.com/ixjtCWka |
| 18:12.57 | saint_ | ha.. i just realized that when you read # , it shows as blank ? |
| 18:13.01 | saint_ | but * works ? |
| 18:13.08 | kaldemar | saint_: use GotoIf |
| 18:13.21 | saint_ | kaldemar: nah, the issue is the read i think |
| 18:13.35 | saint_ | if I verbose the value of the read, and # is entered, I have nothing |
| 18:13.36 | gavimobile | do I need to restart anything if I change an agi script from the agi-bin dir? |
| 18:13.40 | saint_ | but if I press * , then I see it |
| 18:14.15 | kaldemar | saint_: read the app documentation for Read |
| 18:14.33 | WIMPy | saint_: Yes, Read doesn't read #. I did a patch for that once. You can find it on jira somewhere. |
| 18:14.35 | kaldemar | saint_: but still, use Gotoif |
| 18:14.44 | WIMPy | Possibly on my sire as well. |
| 18:14.48 | [TK]D-Fender | qakhan, So the call gets answered, then plays back that recording. How long do you wait before hitting "1", and how many times did you hit it? |
| 18:15.04 | saint_ | WIMPy: are they going to implement your patch at some point ? |
| 18:15.13 | saint_ | I'll work around .. |
| 18:15.17 | saint_ | but it sucks for the future :) |
| 18:15.28 | WIMPy | NFI. Maybe it should go to reviewboard? |
| 18:15.38 | qakhan | i put 10 Sec to be wait for dtmf, 3 times |
| 18:16.49 | WIMPy | You can always patch yourself. |
| 18:16.59 | qakhan | [TK]D-Fender here is with dtmf log http://pastebin.com/1BGWnr79 |
| 18:17.02 | [TK]D-Fender | qakhan, What do you mean you "10 Sec to be wait"? |
| 18:17.24 | mjordan | https://issues.asterisk.org/jira/browse/ASTERISK-18454 |
| 18:18.29 | qakhan | it wait for max 10 sec for dtmf input |
| 18:18.35 | qakhan | see my dialplan |
| 18:18.48 | qakhan | <PROTECTED> |
| 18:19.26 | [TK]D-Fender | qakhan, 10 plus the announcement. |
| 18:19.52 | *** join/#asterisk artyx (U2FsdGVkX1@junction.googleplex.net) |
| 18:20.22 | qakhan | yes |
| 18:20.24 | artyx | what's a good cheap FXO/FXS card suitable for everyday residential/light commercial use.. the ability to run two calls at once is a plus but not required |
| 18:20.37 | [TK]D-Fender | qakhan, Clearly it hadn't finished the annoucement |
| 18:20.51 | qakhan | but it works with other context |
| 18:21.04 | [TK]D-Fender | qakhan, You announcement clearly hadn't finished <- |
| 18:21.04 | artyx | I thought the sangoma would be a good solution, but someone brought up some usb skype deal you can get for like 20$ |
| 18:21.12 | *** join/#asterisk dpilon (~dpilon@c-50-138-178-238.hsd1.ct.comcast.net) |
| 18:21.25 | [TK]D-Fender | artyx, Which * won't support |
| 18:21.33 | TazzNZ | artyx: I have some sangoma cards - which rock ! :) |
| 18:21.36 | jmetro | Is there any way to add a delay to the beginning of a queue before the first announce for queue wait time? |
| 18:21.37 | artyx | What do you mean * wont' |
| 18:21.45 | TazzNZ | artyx: * = asterisk |
| 18:21.46 | qakhan | even if i press 1 during IVR it accept it and forwward the call |
| 18:21.58 | TazzNZ | artyx: as in the software |
| 18:22.03 | [TK]D-Fender | qakhan, Because Background is listening. |
| 18:22.03 | qakhan | is it not working in this context |
| 18:22.24 | [TK]D-Fender | qakhan, It apparently IS workign. Ias picked up the DTMF |
| 18:22.45 | TazzNZ | artyx: the sangoma cards I have setup is deployed in a moderate use company |
| 18:22.47 | artyx | I see tutorials all over the place... its out of mfr ... But its supported (since you didnt even see the hardware I have to assume you are psychic and looking at my monitors "glow" and know its an SPA 3000 |
| 18:23.02 | qakhan | is there any way to reduce the time? |
| 18:23.07 | artyx | It certainly looks supported to me |
| 18:23.41 | TazzNZ | artyx: the USB skype thing is what [TK]D-Fender was refering to |
| 18:24.11 | TazzNZ | the spa300 is "just another sip device" for * |
| 18:24.27 | artyx | Well the spa-3000 is cheaper than a sangoma b600 ... Which i use for work already and am very happy with it, but i'm not sporting 4 analog lines at home |
| 18:24.55 | artyx | The usb sangoma fxo/fxs is still in the $100-$140 range |
| 18:25.19 | artyx | And so i ask, what bargain basement solutions do you guys use, any strong recommednations for a CHEAP one with versatility |
| 18:25.34 | Qwell | ~cheap |
| 18:25.34 | infobot | cheap is probably a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 18:25.53 | *** join/#asterisk rgsteele (~rgsteele@12.150.6.65) |
| 18:25.55 | artyx | It makes no sense to dish out $$$$ for a pet project on the homefront |
| 18:26.11 | TazzNZ | artyx: what are you after - requirements wise |
| 18:26.32 | TazzNZ | FXO/FXS etc |
| 18:27.01 | artyx | Under 70$ .. one line out, the ability to do multiple incoming lines over sip and pstn, 1 fxs connection port 1 fxo sip call from fxs to sip or fxs to fxo |
| 18:27.15 | [TK]D-Fender | qakhan, autofallthrough=no and end your "s" on that background. Implement "t" for your timeout |
| 18:27.18 | artyx | And it should work with asterisk ;) |
| 18:28.33 | TazzNZ | artyx: imho, I would go with the spa3000 |
| 18:28.44 | artyx | Groovy, because its like $20-$30 on ebay |
| 18:28.58 | TazzNZ | and you can have multiples of them |
| 18:29.52 | TazzNZ | but - for anything more than a home project - I would switch to the sangoma |
| 18:29.57 | artyx | There was this sangoma b600 i was trying to pick up for $100 ... but someone sniped it at 2 mins to go for 2 bucks more |
| 18:30.10 | artyx | I will not pay $150 for a home telephony card ... outo f my price range for playign around |
| 18:30.37 | TazzNZ | yeah - can't argue that |
| 18:30.46 | artyx | plus the buyer turned around and relisted it for $150 with a $250 buy it now |
| 18:30.47 | WIMPy | That's the catch. If yu have a cheap line, you need expensive hardware. |
| 18:30.49 | artyx | and its like KMA |
| 18:31.08 | *** join/#asterisk Mon|A|rch (~SBean@72.29.180.35) |
| 18:31.40 | Mon|A|rch | so, I'm trying to pass a name through an originate action so that festival can read it out |
| 18:31.45 | artyx | WIMPy: If you have an expensive line, you need expensive hardware, so wheres the counterpoint to that bit of wisdom |
| 18:31.49 | Mon|A|rch | but I'm getting garbled results |
| 18:31.57 | [TK]D-Fender | Mon|A|rch, Which Originate action? |
| 18:32.03 | Mon|A|rch | does adding a space cause issues when passing channel variables? |
| 18:32.19 | Mon|A|rch | the AMI action, through gui-type urls |
| 18:32.33 | WIMPy | artyx: No, you don't. |
| 18:32.40 | [TK]D-Fender | Mon|A|rch, Look at how the function is actually called in the dialplan on execution |
| 18:32.47 | Mon|A|rch | variable=var1=robert robert&variable=... |
| 18:32.50 | Mon|A|rch | that sort of thing |
| 18:33.06 | Mon|A|rch | Festival(${NAME})? |
| 18:33.21 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1003) |
| 18:33.29 | Mon|A|rch | i have some brief waits before and after |
| 18:33.35 | Mon|A|rch | since I've noticed festival needs some wait times |
| 18:34.30 | Mon|A|rch | what i think is happening, is the first festival call reads the first name, then the second festival call reads the last name |
| 18:34.37 | TazzNZ | can anyone point me to some docs on how asterisk 1.8 logs an agent out of a queue for not answering ? I can't seem to find out how to do this in ast1.8 |
| 18:34.40 | Mon|A|rch | since the garbled sounds I'm getting are different |
| 18:35.09 | TazzNZ | all of the docs I find are pre 1.6 (iirc) where is was still "supported" via agents.conf |
| 18:35.54 | Qwell | TazzNZ: autopause no goos? |
| 18:35.56 | Qwell | good* |
| 18:36.36 | TazzNZ | Qwell: I will go and research that - I don't mind pausing them |
| 18:37.06 | TazzNZ | I am using snom300's as the agent phones, so it would be awesome to have a BLF to show them paused or not |
| 18:37.14 | TazzNZ | I already have it for the queue login |
| 18:37.35 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 18:44.10 | saint_ | why would asterisk say "Function If not registered" ..? http://pastebin.com/V4VhbJXz |
| 18:44.46 | Qwell | Because If isn't the name of a function. |
| 18:45.04 | kaldemar | saint_: IF |
| 18:46.14 | saint_ | arf.. thanks kaldemar |
| 18:46.32 | jmetro | hmm.. trying to set up a queue, how do i introduce a delay in the start of the queue to stop it from immediately announcing the wait time? |
| 18:46.46 | *** join/#asterisk navaismo (~navaismo@189.241.115.147) |
| 18:48.55 | *** join/#asterisk Elleni (b253c357@gateway/web/freenode/ip.178.83.195.87) |
| 18:52.09 | *** part/#asterisk AnGrYfUrBy (~MrX@pdpc/supporter/active/angryfurby) |
| 18:52.19 | Mon|A|rch | i guess i have a different question then |
| 18:53.03 | *** part/#asterisk tmkt (~david@206.223.170.189) |
| 18:53.03 | Mon|A|rch | what sort of debug do i need to enable to see ami actions as/before/after they execute? |
| 18:53.10 | Mon|A|rch | any of those three will work |
| 18:53.40 | WIMPy | manager set debug on |
| 18:53.55 | Mon|A|rch | thanks WIMPy |
| 18:54.52 | Mon|A|rch | also, does anyone know what app_mp3.c:113 timed_read: Poll timed out/errored out with 0 |
| 18:54.54 | Mon|A|rch | means? |
| 18:54.57 | Mon|A|rch | the audio file plays |
| 18:57.33 | WIMPy | Never tried it before, but that manager debug doesn't show anything for me. |
| 18:57.53 | *** join/#asterisk ra21vi (~ravi@122.177.49.158) |
| 18:57.54 | WIMPy | always did the debugging at the other end. |
| 18:58.13 | Mon|A|rch | well, the other end is sending the information (i assumed) correctly |
| 18:58.55 | ra21vi | I wrote an AGI script in python. But its breaking, because if I dial 7777 (simulate incoming) in elastix, it says "call declined" |
| 18:59.12 | ra21vi | I want to see the error.. Where are AGI errors logged? |
| 18:59.47 | navaismo | agi set debug on |
| 19:00.14 | [TK]D-Fender | That logs AGI command calls and responses... it won't debug if one simply ends |
| 19:02.26 | *** part/#asterisk Elleni (b253c357@gateway/web/freenode/ip.178.83.195.87) |
| 19:02.58 | igcewieling | generally I stop asterisk, and then start it as "asterisk -cvvv". This will make STDERR display on the Asterisk console. Keep in mind if you CTRL-S or CTRL-Z or something like that Asterisk still stop processing calls and if you lose your connection Asterisk will exit. Otherwise you won't see any errors from your script which are sent to STDERRO |
| 19:03.10 | *** join/#asterisk Elleni (b253c357@gateway/web/freenode/ip.178.83.195.87) |
| 19:04.35 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.30.159) |
| 19:13.35 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:16.17 | Mon|A|rch | so, what characters will festival read as a pause besides spaces? |
| 19:16.25 | Mon|A|rch | since apparently i can't pass spaces into variables |
| 19:16.55 | ra21vi | navaismo: i did agi set debug on. But the output is not showing any error. |
| 19:17.01 | artyx | why not use bg or nohup igcewieling |
| 19:20.06 | Mon|A|rch | nvm, urlencoding spaces and ... works |
| 19:20.38 | Mon|A|rch | does festival.conf have more than one robot voice? |
| 19:21.22 | Mon|A|rch | because the default robot voice is awful |
| 19:23.47 | Mon|A|rch | even changing the volume would be nice |
| 19:23.54 | ra21vi | here is the code and output in console: http://pastebin.com/3pWg6mwm |
| 19:23.55 | Mon|A|rch | it doesn't look like there are any default settings for that though |
| 19:24.26 | rrittgarn | anyone play much with keepalive in 11? Just need to add keepalive=yes to sip.conf right? |
| 19:24.53 | [TK]D-Fender | ra21vi, exten => s,n,AGI(/root/workspace/ivr/app.py) |
| 19:25.13 | [TK]D-Fender | r2Elastix does NOT run Asterisk as ROOT. You should fix your file's location and owner |
| 19:25.44 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 19:25.45 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 19:26.12 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 19:27.09 | gavimobile | im trying to get the speech recognition to work from http://zaf.github.com/asterisk-speech-recog/ it works when I disable ssl. I installed the ssl perl module listed in the readme but I cannot get this to work with ssl. |
| 19:29.27 | [TK]D-Fender | gavimobile, And you aren't showing us what you've done... |
| 19:30.26 | gavimobile | I will in just a few mins |
| 19:30.30 | gavimobile | brb |
| 19:31.53 | navaismo | ra21vi, yep there in no agi debug output have you tried with phpagi? |
| 19:32.10 | ra21vi | [TK]D-Fender: its in /root/ dir, but has been assigned 777 +x . So should be able to be executed by any user |
| 19:32.22 | ra21vi | navaismo: I am using Python |
| 19:32.45 | [TK]D-Fender | ra21vi, No. That folder is owned by root. Move the file. Change the owner. |
| 19:32.47 | ra21vi | navaismo: if I write to STDERR, will it be shown in console |
| 19:32.58 | ra21vi | [TK]D-Fender: oh, got it :) |
| 19:35.03 | *** join/#asterisk BriGuy (~BriGuy@74.115.41.6) |
| 19:46.43 | *** join/#asterisk sewellj (~chatzilla@2607:0:3:1::4) |
| 19:47.52 | ra21vi | [TK]D-Fender: I moved to other location.. and it shows now <SIP/100-00000011>AGI Script /opt/workspace/ivr/app.py completed, returning 0. |
| 19:48.30 | ra21vi | [TK]D-Fender: but still, when I call, i don't hear the sound. In AGI scipt I did play intercom sound |
| 19:48.57 | [TK]D-Fender | ra21vi, I do not see the new file and it's contents, nor the new call attempt with AGI debug enabled. |
| 19:50.41 | ra21vi | here it is: http://pastebin.com/fEYYDH1B |
| 19:51.26 | ra21vi | [TK]D-Fender: at line 71 it shows script completed |
| 19:52.42 | [TK]D-Fender | ra21vi, I would check on the location for the library it uses and the rest of your syntax. This is now a python issue |
| 19:56.04 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 20:06.41 | ra21vi | [TK]D-Fender: probably there is some problem with Library I am using. If I write same actions in dialplan in extensions_vdp.conf if works. |
| 20:08.11 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 20:08.18 | danfromuk | #asterisk-social |
| 20:08.21 | danfromuk | oops |
| 20:09.05 | ChannelZ | Let's exchange recipes! |
| 20:09.16 | danfromuk | I know this might be off topic but I'm not sure where to ask.... does anyone have experience for developing a sip softphone? |
| 20:10.01 | danfromuk | of* |
| 20:10.35 | igcewieling | Does anyone have any idea what might cause the following message (with various channel numebrs) to be spewing endlessly on the CLI? |
| 20:10.35 | igcewieling | <PROTECTED> |
| 20:11.56 | mjordan | igcewieling: that's usually an indication of a SIP channel reference count bug. What version are you running? |
| 20:12.06 | igcewieling | 1.8.19.1 |
| 20:12.19 | mjordan | hm |
| 20:12.50 | mjordan | the trick with those is figuring out the SIP call flow leading up to the problem. Is it happening with a particular group of channels that use the same dialplan extension? |
| 20:13.07 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 20:13.49 | igcewieling | Likely, but I do not know for sure. this is production we we must restart. |
| 20:16.11 | igcewieling | I'll try to find out. |
| 20:17.47 | *** join/#asterisk vlad_starkov (~vlad_star@178.176.30.159) |
| 20:18.05 | igcewieling | One unusual thing we are doing is using this format Dial(SIP/peer/dest/hostname) which is not a common way of dialing as far as I can tell |
| 20:19.23 | igcewieling | restarting asterisk appears to not fix the issue. |
| 20:21.34 | gavimobile | ok im back. spech_recognition.agi from google won't work for me if I enable ssl. here is my output with age set debug on http://pastebin.com/U2R8SKqg |
| 20:22.25 | gavimobile | here is my dialplan http://pastebin.com/pSche5ce |
| 20:25.07 | mjordan | igcewieling: if you're able to reproduce the issue, please file a bug report with a pcap showing the SIP call flow and the relevant portion of the dialplan |
| 20:25.37 | igcewieling | mjordan: This is FreePBX so there will be 50 bazillion dialplan lines. It appears that parking is also failing. |
| 20:25.41 | gavimobile | and here is my agi script http://pastebin.com/mbseP641 |
| 20:32.51 | rrittgarn | gavimobile: I got that script working with SSL By installing libcrypt-ssleay-perl and liburi-perl |
| 20:33.01 | rrittgarn | i have it working 100% in about 15 minutes... nice find... |
| 20:33.39 | gavimobile | rrittgarn: was it working WITHOUT ssl before you installed those as well? |
| 20:33.45 | rrittgarn | correct |
| 20:34.03 | rrittgarn | Wouldn't work with SSL until i installed those. Gave me -1, -1, -1, -1 as the results from their example code |
| 20:34.05 | gavimobile | rrittgarn: ill give it a try |
| 20:34.07 | rrittgarn | running Debian Squeeze |
| 20:34.19 | gavimobile | the documentation is soo poor online |
| 20:35.56 | rrittgarn | still no dice? |
| 20:36.27 | gavimobile | rrittgarn: im on centos, so im finding simmilar packages |
| 20:36.40 | rrittgarn | gavimobile: ah... best of luck |
| 20:36.57 | gavimobile | rrittgarn: thanks, will let you know results |
| 20:39.54 | gavimobile | this is what I had instead of libcrypt-ssleay perl-Crypt-SSLeay |
| 20:40.56 | gavimobile | unfortunetly it still doesn't work. regarding the other package, im not sure which replacement package is the right one for centos |
| 20:41.47 | gavimobile | here's a list of available packages |
| 20:41.49 | gavimobile | http://pastebin.com/RY9JZ70z |
| 20:53.56 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 21:02.44 | *** join/#asterisk vlad_starkov (~vlad_star@178.177.199.235) |
| 21:23.14 | ra21vi | [TK]D-Fender: I got the script working.. Actually the interpretor was not getting resolved. So python agi file was not executed. It is really strange that asterisk console did not show any error about it even the agi debuggging was on |
| 21:25.03 | ra21vi | With AMI, is it possible to get the current call info and channel vars for a inbound call connected to a SIP ext. |
| 21:34.07 | *** join/#asterisk darkdrgn2k (~darkdrgn2@69.165.131.20) |
| 21:34.09 | darkdrgn2k | hi all |
| 21:34.26 | darkdrgn2k | what is the rtpmap for g711 and g722 ? |
| 21:34.35 | darkdrgn2k | ie: how can you tell from a wireshark which protocol is being used |
| 21:38.46 | *** part/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 21:53.10 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 21:55.52 | *** join/#asterisk Defraz (~Defraz@24-116-129-18.cpe.cableone.net) |
| 21:56.22 | *** join/#asterisk justsomedood (~warren@mail.serverplus.com) |
| 22:01.41 | *** join/#asterisk sewellj (~chatzilla@2607:0:3:1::4) |
| 22:02.22 | *** join/#asterisk melanieplageman (~melaniepl@105.230.11.90) |
| 22:08.34 | *** join/#asterisk jsjc (~Adium@91.Red-83-60-132.dynamicIP.rima-tde.net) |
| 22:17.12 | ra21vi | will it be a valid dialplan with single line: exten => s,1,AGI(/opt/workspace/ivr/app1.py) |
| 22:17.31 | ra21vi | I mean, just calling AGI which handles all answer/hangup in script |
| 22:17.42 | WIMPy | sure |
| 22:17.58 | ra21vi | oh. Hi WIMPy |
| 22:18.11 | ra21vi | thanks |
| 22:19.44 | *** join/#asterisk ankurgupta0001 (~ankurgupt@216-75-234-234.static.wiline.com) |
| 22:20.39 | ankurgupta0001 | MWI not working when moved voicemail storage to mysql database,please help |
| 22:22.12 | navaismo | that remember me katty's pbx |
| 22:22.45 | *** join/#asterisk mathi (Matthew@ip-62-235-214-191.dsl.scarlet.be) |
| 22:22.54 | navaismo | ankurgupta0001, show us configs |
| 22:23.00 | mathi | hi |
| 22:23.28 | mathi | I have bad sound quality with SIP, can anyone have a look ? http://forums.asterisk.org/viewtopic.php?f=1&t=85531 |
| 22:23.32 | ankurgupta0001 | which all files,do I need to show |
| 22:23.47 | *** join/#asterisk gtTuna (Tuna@d149-67-253-75.try.wideopenwest.com) |
| 22:24.05 | gtTuna | Is there anyway to tell what file exactly this is occuring at when I do a dialplan reload? |
| 22:24.07 | gtTuna | <PROTECTED> |
| 22:24.54 | navaismo | gtTuna, i guess is extensions.conf line 24 |
| 22:25.06 | gtTuna | nope |
| 22:25.14 | navaismo | what version of asterisk are you using? |
| 22:25.33 | gtTuna | [macro-voicemail] is line 24 |
| 22:25.41 | gtTuna | 1.6 |
| 22:26.32 | gtTuna | even did something like |
| 22:26.35 | gtTuna | find /usr/local/etc/asterisk/ -type f -exec sed -n '24p' {} \; | grep same |
| 22:26.50 | gtTuna | for the different lines its having issues with |
| 22:27.12 | navaismo | that version support same? |
| 22:27.32 | gtTuna | yep! |
| 22:27.54 | gtTuna | i believe so...pretty sure it's used on other 1.6 servers i manage |
| 22:28.11 | jmetro | gtTuna your username makes me want a can of tuna fish. |
| 22:28.40 | igcewieling | You can tune a file system, but you can't tune a fish. |
| 22:29.17 | jmetro | you can tune a piano, to get to the other side. |
| 22:29.48 | Qwell | ~upgrade asterisk |
| 22:29.48 | infobot | Before requesting assistance, you should be running the latest version of a supported release branch. See the channel topic for the latest versions available in currently supported branches. |
| 22:29.49 | WIMPy | You can upgrade Asterisk, but you can't live in the past. |
| 22:29.54 | ankurgupta0001 | I am using Asterisk 11.2.0,voicemails are getting saved to db,but no mwi |
| 22:30.45 | gtTuna | not really an option at this time |
| 22:31.00 | igcewieling | ankurgupta0001: make sure you have pollmailboxes set (or whatever the option is called) |
| 22:31.20 | ankurgupta0001 | yes already did it in voicemail.conf |
| 22:31.32 | mathi | WIMPy, can you have a look at my link pleaaase? |
| 22:32.19 | navaismo | <PROTECTED> |
| 22:32.53 | ankurgupta0001 | yes the peer has mailbox defined |
| 22:33.10 | WIMPy | has a tendency to ignore sip topics |
| 22:33.29 | mathi | :-p |
| 22:34.23 | navaismo | gtTuna, the version of asterisk is only 1.6 or 1.6.something |
| 22:34.57 | gtTuna | symetrically enough, 1.6.1.6 |
| 22:35.05 | gtTuna | was same added in 1.6.2? |
| 22:39.23 | navaismo | can't find the changelog |
| 22:48.54 | navaismo | gtTuna, according with this links seems like same apper on 1.6.2 |
| 22:48.58 | navaismo | http://www.sinologic.net/blog/2010-04/cambios-importantes-en-asterisk-1-6-2/ |
| 22:49.39 | gtTuna | makes sense then...this server has the oldest version on it of all our servers |
| 22:50.08 | gtTuna | looks like someone transplanted some dialplan code from a server that runs 1.6.2+, 1.8+ |
| 22:51.18 | *** join/#asterisk janmate (~janmate@chello089173160127.chello.sk) |
| 22:51.39 | gtTuna | i guess that leads to another question |
| 22:52.08 | gtTuna | are there any forseeable issues with upgrading from 1.6.1 to 1.8.20.1 ? |
| 22:52.54 | [TK]D-Fender | gtTuna, upgrade.txt <- |
| 22:55.05 | navaismo | ankurgupta0001, what phones are you using? the phones are pinting to the mailboxatserver if have that option? |
| 22:59.48 | *** join/#asterisk melanieplageman (~melaniepl@105.230.11.90) |
| 23:01.01 | navaismo | mathi, are you sure your bandwidth its ok? If you are using sip sotfphone have you tried with gsm codec? |
| 23:01.23 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:01.24 | mathi | navaismo, bandwidth is ok, I posted the results, it's ADSL by the way |
| 23:01.36 | mathi | navaismo, I am not using a softphone |
| 23:02.43 | mathi | navaismo, what can it be else than bandwith ? |
| 23:03.03 | mathi | navaismo, a firewall problem ? |
| 23:03.46 | mathi | navaismo, what is the difference in sending wav files through fxo or sending wav files through SIP ? |
| 23:04.06 | mathi | if I can answer this question, I can start to find the problem ... |
| 23:04.21 | [TK]D-Fender | none |
| 23:04.55 | mathi | then why does SIP give me bad quality and not PSTN/FXO on the exact same server ? |
| 23:06.24 | [TK]D-Fender | Because of network conditions and the quality of what is terminating it |
| 23:06.45 | mathi | [TK]D-Fender, can it be because of firewall ? |
| 23:06.47 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 23:06.54 | [TK]D-Fender | yes |
| 23:07.25 | mathi | [TK]D-Fender, well you see, that doesn't matter with FXO, so there is a difference. what ports should I try to open ? |
| 23:08.50 | *** part/#asterisk justsomedood (~warren@mail.serverplus.com) |
| 23:12.32 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 23:15.27 | [TK]D-Fender | mathi, It isn't a question of ports |
| 23:15.42 | [TK]D-Fender | Packets are flowing. If they weren't you would have NO audio and your call would die |
| 23:15.51 | mathi | what can I do then ? ... |
| 23:15.58 | [TK]D-Fender | mathi, Your QUALITY is screwed up somewhere along the way |
| 23:16.13 | [TK]D-Fender | Either fighting for bandwidth, packet-loss, jitter, etc |
| 23:16.32 | [TK]D-Fender | And that's just up to your router... then there is the entire INTERNET between you and your provider |
| 23:18.13 | mathi | [TK]D-Fender, but about the internet, I tried on a server at home and it works. that server that gives bad audio quality is also located in Brussels |
| 23:26.19 | *** part/#asterisk jsjc (~Adium@91.Red-83-60-132.dynamicIP.rima-tde.net) |
| 23:30.18 | *** join/#asterisk saint_ (~saint@c-68-38-56-184.hsd1.nj.comcast.net) |
| 23:32.56 | janmate | all: anyone who uses asterisk with LDAP and knows howto solve the AstAccountPermit and AstAccountDeny attributes ordering problem (LDAP attributes are unsorted by definition) |
