| 00:03.14 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 00:03.49 | elliot98 | I am trying to get the port address from the sip->pvt |
| 00:04.38 | elliot98 | what is the method for doing this? |
| 00:04.49 | elliot98 | sorry, port number for a peer |
| 00:07.42 | elliot98 | what kind of variable is the port number in the sip data structure? |
| 00:09.05 | tuxx- | <PROTECTED> |
| 00:09.21 | elliot98 | gde? |
| 00:13.27 | *** join/#asterisk nobodyshome (~nobodysho@bas10-kitchener06-1279682439.dsl.bell.ca) |
| 00:14.10 | *** join/#asterisk slick_nic (~slick_nic@87-194-180-7.bethere.co.uk) |
| 00:17.12 | p3nguin | r0m|u: Maybe later. Have a meeting in a few minutes. |
| 00:18.31 | nobodyshome | he's at the school now |
| 00:18.37 | nobodyshome | waiting for class |
| 00:18.41 | p3nguin | oh |
| 00:18.46 | nobodyshome | just spoke with him |
| 00:18.54 | nobodyshome | and im eating so no worries |
| 00:22.33 | elliot98 | anyone know how to retrieve the port number in a sip data structure? |
| 00:26.06 | *** join/#asterisk obnauticus (~obnauticu@about/windows/regular/obnauticus) |
| 00:44.14 | ChannelZ | elliot98: not sure exactly what you're asking, but there's the SIPPEER function |
| 00:51.33 | *** join/#asterisk kaushal (~kaushal@115.118.229.164) |
| 00:51.36 | kaushal | Hi |
| 00:52.15 | kaushal | is there a way to know the call states of the Outbound call ? |
| 00:52.29 | kaushal | I mean is it failed or answered or ..... by the user |
| 00:53.37 | Greenlight | The dial command sets a chan variable |
| 00:54.13 | nobodyshome | ${CDR(amaflags)} |
| 00:54.16 | kaushal | Greenlight: are you referring to me ? |
| 00:54.23 | nobodyshome | is that it |
| 00:54.24 | nobodyshome | ? |
| 00:54.24 | Greenlight | mmmhmm |
| 00:54.29 | *** join/#asterisk netman (netman@176.171.76.188.dynamic.jazztel.es) |
| 00:55.07 | Greenlight | DIALSTATUS gets set after dial has completed |
| 00:55.44 | Greenlight | And you can use that to see what happened to the call |
| 00:55.55 | Greenlight | Busy, no answer, answered etc |
| 00:56.05 | nobodyshome | thats the right variable? |
| 00:56.19 | Greenlight | Umm I think so |
| 00:56.22 | kaushal | Greenlight: are you referring to me ? |
| 00:56.42 | Greenlight | kaushal: I am |
| 00:57.15 | kaushal | ok |
| 00:57.27 | kaushal | Greenlight: is it via CLI ? |
| 00:57.40 | kaushal | I mean DIALSTATUS |
| 00:57.41 | nobodyshome | or is it this? ${CDR(accountcode)} |
| 00:58.01 | Greenlight | It's ${DIALSTATUS} |
| 00:58.10 | Greenlight | And not via the CLI, in the dialplan |
| 00:58.14 | kaushal | ok |
| 00:58.16 | kaushal | got it |
| 00:58.53 | Greenlight | Of course, you can also get the channel status from the CLI |
| 00:58.57 | Greenlight | "Core show channels" etc |
| 00:59.49 | kaushal | ok |
| 01:02.29 | kaushal | Greenlight: can i pastebin the extensions.conf ? |
| 01:02.43 | Greenlight | Sure |
| 01:02.55 | Greenlight | What is it you're trying to acheive? |
| 01:03.28 | kaushal | Greenlight: http://pastebin.ubuntu.com/738826/ |
| 01:03.43 | kaushal | basically i am using call files to dial out 240 numbers |
| 01:04.15 | kaushal | and i need to know the call state once call files are pushed to asterisk spool |
| 01:04.41 | Greenlight | Hmm |
| 01:04.54 | *** join/#asterisk grandpapadot (~grandpapa@99.175.248.81) |
| 01:04.58 | Greenlight | So you're wanting them output to somewhere? |
| 01:05.40 | grandpapadot | Hey guys, trying to use lame to convert asterisk recorded (1.8) wav files to mp3, it's working, but I'm loosing a lot of clarity and volume when playing back through asterisk ... any suggestions? |
| 01:05.47 | *** join/#asterisk SwK (~SwK@freeswitch/developer/swk) |
| 01:06.05 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 01:06.19 | grandpapadot | Using: lame --quiet -h -a -m m --resample 44.1 |
| 01:06.20 | Greenlight | kaushal: Take a look at http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS you'll see what I mean. I suspect if you're wanting to record the outcomes you'll maybe need to use AGI, or you could call the system command and write to a file perhaps? |
| 01:06.42 | Greenlight | grandpapadot: Two secs lemme check my syntax that I use |
| 01:07.10 | grandpapadot | Thanks, Greenlight ;) |
| 01:07.11 | Greenlight | nice /usr/local/bin/lame -b 16 -m m -q 8 --resample 8 "$i" "${i%.wav}.mp3"; |
| 01:07.30 | Greenlight | Works a treat for me. ymmv though |
| 01:08.35 | grandpapadot | Ya, that's worse, lol :) |
| 01:08.39 | Greenlight | lol |
| 01:08.39 | grandpapadot | hmm ... |
| 01:08.51 | kaushal | Greenlight: ok |
| 01:08.56 | Greenlight | Your ears must be more sensitive than mine |
| 01:09.00 | grandpapadot | lol |
| 01:09.18 | Greenlight | I use that for call recordings for quality and compliance in many call centres |
| 01:09.27 | kaushal | you could call the system command and write to a file perhaps |
| 01:09.35 | kaushal | Greenlight: Any example ? |
| 01:09.40 | grandpapadot | It's just that the wav file sounds loud, crisp, and clear and the mp3 sounds washed out with low volume |
| 01:09.57 | grandpapadot | Maybe it's the channel driver |
| 01:10.12 | Greenlight | If the wav is okay, then so should the mp3 be |
| 01:10.28 | Greenlight | mp3 is lossy but you shouldn't notice it really |
| 01:10.39 | kaushal | Greenlight: any example of ${DIALSTATUS} in extensions.conf ? |
| 01:10.47 | Greenlight | Try without any settings, just a straigh convert - still the same? |
| 01:11.01 | Greenlight | kaushal: There are some on that page I linked - down towards the bottom |
| 01:11.15 | kaushal | ok |
| 01:11.20 | Greenlight | kaushal: I'll see if I can dig out the system command syntax, can't remember of the top of my head |
| 01:11.25 | grandpapadot | I'm downloading them to my pc to see how they sound, one sec (thanks for the help) |
| 01:11.38 | Greenlight | grandpapadot: Where did you have them before? |
| 01:11.45 | Greenlight | Were you listing to the mp3's via Asterisk ? |
| 01:11.48 | Greenlight | *listening |
| 01:12.15 | grandpapadot | They sound identical .. has to be something with asterisk's playback |
| 01:12.22 | grandpapadot | Greenlight: yep |
| 01:12.29 | grandpapadot | Greenlight: was listening through asterisk |
| 01:12.53 | Greenlight | Ahh yea I'd avoid listening to mp3's direct |
| 01:13.13 | Greenlight | Always seems to end up sounding crap |
| 01:13.15 | grandpapadot | lol, well, I was kind of trying to get all my customers sound files in mp3 format for ease of listeningn through a browser ... |
| 01:13.27 | Greenlight | grandpapadot: Yea that's exactly what I do |
| 01:13.28 | grandpapadot | back to planning, lol ;) tnx for the help |
| 01:13.38 | Greenlight | If it's via browser it'll be okay |
| 01:13.47 | Greenlight | I have a link they click to access the mp3 |
| 01:13.58 | Greenlight | And a script that converts all wavs to mp3s nightly |
| 01:14.02 | grandpapadot | Right .. so I'll just have to maintain the wav for asterisk playback and the mp3 for downloads ... |
| 01:14.05 | Greenlight | kaushal: exten => s,n,System(umask 002 && /bin/mkdir -p -m 775 ${GLRecordingPath}) |
| 01:14.16 | grandpapadot | I think I can work with that .. just gotta think it through a bit |
| 01:14.30 | Greenlight | grandpapadot: Ahh, you want them available via Asterisk too... bummer |
| 01:14.41 | grandpapadot | Greenlight: Yea, that was my "pie in the sky" lol |
| 01:15.05 | Greenlight | I've never really played too much with the builtin mp3 stuff |
| 01:15.16 | grandpapadot | Greenlight: Apparently it sucks, lol |
| 01:15.19 | Greenlight | I gues you could always convert to a wav in the dialplan, then play it |
| 01:15.30 | kaushal | Greenlight: ${GLRecordingPath} ? |
| 01:15.36 | kaushal | what does it do ? |
| 01:15.38 | grandpapadot | Greenlight: Too high volume for that, it will be easier to just maintain the wav file and a seperate mp3 for downloading ... |
| 01:15.44 | kaushal | Greenlight> kaushal: exten => s,n,System(umask 002 && /bin/mkdir -p -m 775 ${GLRecordingPath}) |
| 01:15.55 | *** join/#asterisk hovel (~hovel@unaffiliated/hovel) |
| 01:15.59 | Greenlight | kaushal: I just copied and pasted an extract where I use the system command |
| 01:16.04 | kaushal | ok |
| 01:16.12 | Greenlight | ${GLREcordingPath} is a channel variable that my app sets |
| 01:16.34 | Greenlight | In this case I'm using the system command to ensure the directory for the recording exists |
| 01:16.48 | kaushal | Greenlight: still not completely understood |
| 01:17.09 | Greenlight | Umm |
| 01:17.25 | Greenlight | Well it depends on your requirements |
| 01:17.30 | kaushal | so using http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 01:17.42 | Greenlight | But, you might want to call a .sh file somewhere, with the telephone number, and the dialstatus |
| 01:17.48 | Greenlight | Or a php file |
| 01:17.49 | Greenlight | Etc |
| 01:18.02 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 01:18.02 | Greenlight | From there you can update your database, or whatever you need to do with the outcome |
| 01:18.11 | kaushal | Greenlight: so i push 200 call files to asterisk spool |
| 01:18.25 | kaushal | and i need to know the dialstatus of all 200 call files |
| 01:18.41 | Greenlight | So, you need a unique identifier, say the phone number |
| 01:19.17 | Greenlight | Then you could say do |
| 01:19.55 | Greenlight | exte => s,n,System(/sbin/MyScript.sh ${EXTEN} ${DIALSTATUS}) |
| 01:20.17 | Greenlight | And "MyScript.sh" would be called with the telephone number, and the dialstatus (call outcome) |
| 01:21.55 | kaushal | Greenlight: ok |
| 01:22.49 | kaushal | Greenlight: so where would the output be written to ? |
| 01:23.12 | Greenlight | Wherever you choose to write it to using MyScript.sh |
| 01:23.12 | kaushal | is it /var/log/asterisk/sangoma or /var/log/asterisk/messages file |
| 01:23.17 | kaushal | ok |
| 01:23.43 | kaushal | Greenlight: so let me pastebin the call file script |
| 01:23.44 | Greenlight | Or |
| 01:23.50 | Greenlight | Say, if its easier: |
| 01:24.20 | Greenlight | exten => s,n,System(echo "${EXTEN} : ${DIALSTATUS}" >> outputfile) |
| 01:24.32 | Greenlight | And you can parse if from outputfile |
| 01:24.39 | kaushal | ok |
| 01:24.42 | kaushal | got it |
| 01:25.00 | kaushal | so this i need to set it in dial plan ? |
| 01:25.08 | Greenlight | Exactly |
| 01:25.20 | kaushal | and outputfile should be something like /var/tmp/Channel: DAHDI/g0/9930963298 |
| 01:25.21 | kaushal | Application: Playback |
| 01:25.21 | kaushal | Data: hello-world |
| 01:25.24 | kaushal | sorry |
| 01:25.45 | kaushal | <PROTECTED> |
| 01:25.54 | Greenlight | Bingo |
| 01:26.03 | kaushal | Greenlight: Thanks |
| 01:26.17 | kaushal | Greenlight: much appreciated |
| 01:26.28 | Greenlight | No problemo - hope you get it all working |
| 01:26.38 | kaushal | also if i need to use call files ? |
| 01:26.44 | kaushal | does it hold the same |
| 01:26.48 | kaushal | since i am using AMI |
| 01:27.08 | Greenlight | Tbh I've never used call files - I do all my stuff via the AMI |
| 01:27.16 | kaushal | not sure if i am understanding that bit |
| 01:27.38 | kaushal | so using call files meaning it also parse dialplan ? |
| 01:28.05 | Greenlight | Yea, any call made, no matter what the source, goes through the dialplan |
| 01:28.16 | Greenlight | Be it from a phone, a call file or the AMI |
| 01:28.22 | kaushal | ok |
| 01:28.52 | Greenlight | But if you were say originating calls from the AMI, you can track things directly from there |
| 01:29.15 | kaushal | Greenlight: so /tmp/outputfile file would have numbers and callstate ? |
| 01:29.21 | Greenlight | Indeed |
| 01:29.42 | kaushal | so for example 9930963298 Answered ? |
| 01:29.58 | Greenlight | Almost, it would be ANSWER |
| 01:30.01 | Greenlight | But yea, pretty much |
| 01:30.04 | kaushal | and 9930961298 NotAnswered ? |
| 01:30.12 | Greenlight | Or, NOANSWER |
| 01:30.13 | kaushal | something like that ? |
| 01:30.16 | Greenlight | The actual outcomes are on http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 01:30.37 | kaushal | great |
| 01:31.07 | Greenlight | You could also perhaps track the variable ${HANGUPCAUSE} |
| 01:31.24 | Greenlight | Make a few tests and see if you get the output you need |
| 01:31.45 | kaushal | also is there a way to know every bit of the line exten => s,n,System(echo "${EXTEN} : ${DIALSTATUS}" >> /tmp/outputfile) ? |
| 01:32.04 | Greenlight | What do you mean, know every bit? |
| 01:33.03 | kaushal | I mean is it exten => s,n,System(echo "${EXTEN} : ${DIALSTATUS} : ${HANGUPCAUSE}" >> /tmp/outputfile) ? |
| 01:33.31 | Greenlight | Yes, so that would give you for example: |
| 01:33.52 | Greenlight | 01382111111 : ANSWER : NORMAL |
| 01:34.12 | kaushal | ok |
| 01:34.15 | Greenlight | You'd need to replace the "s" extension to suit your own dialplan as well |
| 01:34.23 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 01:35.00 | kaushal | also is there a way to know dry run dial plan ? |
| 01:35.13 | kaushal | just to ensure if my dialplan is correct |
| 01:35.47 | Greenlight | Just "dialplan reload" and see if it groans |
| 01:35.50 | Greenlight | Then test :) |
| 01:35.52 | kaushal | ok |
| 01:36.34 | kaushal | Greenlight: please give me a moment |
| 01:36.34 | Greenlight | You could always comment out the dial bit, and use a NoOp to print, and that way you'd get some output to your file as well |
| 01:37.33 | kaushal | Greenlight: http://pastebin.ubuntu.com/738847/ |
| 01:37.44 | kaushal | please correct me if i am wrong ? |
| 01:39.44 | Greenlight | Som you probably want to have the new bit of code up inside the [outboundcalls] context, or else there is no way to get to it |
| 01:40.00 | Greenlight | And it would not be the "s" extension |
| 01:40.35 | kaushal | ok |
| 01:42.08 | kaushal | let me pastebin that line |
| 01:42.53 | kaushal | Greenlight: little confused there |
| 01:43.13 | kaushal | exten => _0XXXXXXXXXX,1,Dial(${LOCAL}/${EXTEN:1}) |
| 01:43.49 | kaushal | exten => s,n,System(echo "${EXTEN} : ${DIALSTATUS} : ${HANGUPCAUSE}" >> /tmp/outputfile) |
| 01:43.58 | kaushal | under [outboundcalls] context |
| 01:44.04 | kaushal | am i correct ? |
| 01:44.08 | Greenlight | Kinda |
| 01:44.19 | kaushal | let me check |
| 01:44.24 | Greenlight | It's not extension "s", it's _0XXXXXXXXXX |
| 01:44.32 | kaushal | ok |
| 01:44.33 | Greenlight | Or alternatively "same => " |
| 01:44.51 | Greenlight | Since you want the call to "flow" from one command to the next |
| 01:45.41 | kaushal | ok |
| 01:45.57 | kaushal | exten => _0XXXXXXXXXX,1,Dial(${LOCAL}/${EXTEN:1}) => s,n,System(echo "${EXTEN} : ${DIALSTATUS} : ${HANGUPCAUSE}" >> /tmp/outputfile) ? |
| 01:46.23 | Greenlight | Umm |
| 01:46.24 | Greenlight | no |
| 01:46.39 | Greenlight | exten => _0XXXXXXXXXX,n,System(echo "${EXTEN} : ${DIALSTATUS} : ${HANGUPCAUSE}" >> /tmp/outputfile) |
| 01:47.14 | kaushal | sorry |
| 01:47.18 | kaushal | got it now |
| 01:48.12 | Greenlight | Right - Am gonna get some sleep. Laters all! |
| 01:48.22 | kaushal | Greenlight: sure |
| 01:48.31 | kaushal | Greenlight: can i pvt message you now |
| 01:48.46 | Greenlight | I tend to not reply when I'm asleep :P |
| 01:48.53 | kaushal | :) |
| 01:49.22 | Greenlight | It's 2AM here almost, and I've work in morning lol |
| 01:49.28 | kaushal | sure |
| 01:49.32 | Greenlight | nn |
| 01:53.41 | WIMPy | It's only a matter of months. |
| 01:54.07 | Greenlight | Till what |
| 01:54.28 | carrar | till FAILURE OF THE SYSTEM |
| 01:54.50 | Greenlight | Well that doesn't sound good |
| 01:55.05 | carrar | no it doesn't |
| 01:55.14 | Greenlight | looks scared |
| 01:56.11 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-154-7.chyn.qwest.net) |
| 01:56.27 | WIMPy | Until kaushal gets his stoff going. |
| 01:56.46 | Greenlight | hehe |
| 01:57.56 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-154-7.chyn.qwest.net) |
| 01:58.15 | WIMPy | Might make sense to keep logs for him. |
| 01:59.37 | Greenlight | I pretty much did his solution for him, where can he go wrong |
| 02:00.15 | WIMPy | Oh, we've been all over this for weeks. |
| 02:00.21 | Greenlight | Ahh |
| 02:00.27 | Greenlight | lol |
| 02:00.33 | Greenlight | Not his first time in here then |
| 02:01.02 | WIMPy | Nope. But I do indeed not have a clue why he has issues with his telco. |
| 02:01.23 | Greenlight | What issues he been having ? |
| 02:02.15 | WIMPy | Protocoll errors. |
| 02:02.48 | Greenlight | He was ISDN30 wasn't he |
| 02:02.58 | WIMPy | Hmm. Now that I think about it, it might be the same issue I read about on Jira a few days ago. |
| 02:03.03 | WIMPy | 8x |
| 02:04.27 | WIMPy | Hmm. Bad luck I don't have his traces any more. |
| 02:04.43 | Greenlight | Sounds like a strange one |
| 02:05.10 | WIMPy | Yes, it's one of those libpri features again. |
| 02:05.12 | Greenlight | I had a nightmare with an install the other week - turned out to be either a dodgy card or server (we replaced both and issue went away) |
| 02:05.51 | WIMPy | Although I have to admit that it's acting to specification in that case, but that's not a good thing here. |
| 02:26.22 | *** join/#asterisk cmartin0 (~drl@unaffiliated/cmartin0) |
| 02:26.41 | *** join/#asterisk tyman (~tyman@173-12-219-189-Fresno.hfc.comcastbusiness.net) |
| 02:27.21 | cmartin0 | i am a looking at the source code to learn how to using doxygen in C. I am confused as to why some functions have documenting comments where as other functions do not. |
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| 02:39.11 | moos3 | does anyone know what the thrid number in the data field in queue_log is |
| 03:03.30 | *** join/#asterisk grandpapadot (~grandpapa@99.175.248.81) |
| 03:04.05 | grandpapadot | Yo, Greenlight, you still here? I figured out hte format_mp3 volume problem ... |
| 03:04.10 | grandpapadot | .. rather, google did |
| 03:04.15 | grandpapadot | .. but took some digging |
| 03:06.19 | grandpapadot | Well, just in case you care and/or come back ... in format_mp3.c, there is a constant called OUTSCALE, set it to 32768 and boom, your mp3's sound like they are supposed to. |
| 03:08.52 | grandpapadot | .. somebody changed it to 4096 to make moh sound lower when using native ... |
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| 03:46.12 | *** join/#asterisk master_of_master (~master_of@p57B55EF6.dip.t-dialin.net) |
| 03:49.16 | *** join/#asterisk ponyofdeath (~vladi@75.80.175.217) |
| 03:49.56 | ponyofdeath | hi, i have started up asterisk and am trying to join my sip phone but all i see on the interface is SIP register requests without any logs on the asterisk side |
| 03:50.04 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 03:54.15 | p3nguin | Pastebin your entire sip.conf |
| 04:02.43 | SeRi | waz up guys |
| 04:02.49 | SeRi | finaly home :) |
| 04:08.15 | SeRi | p3nguin, you avail? |
| 04:08.30 | p3nguin | I suppose. |
| 04:08.44 | SeRi | p3nguin, I am getting this: cdr.c:940 ast_cdr_end: CDR on channel 'Bridge/0x8970a98-input' has no answer time but is 'ANSWERED' |
| 04:08.51 | p3nguin | Okay. |
| 04:09.10 | SeRi | Is that a bug from confbridge on cdr? |
| 04:09.14 | p3nguin | Maybe. |
| 04:09.50 | SeRi | I see. I get that everyt so often... |
| 04:10.08 | p3nguin | I suppose you could put a NoCDR() before the Answer(). |
| 04:11.39 | p3nguin | I doubt you care about CDR on the conf, anyway. |
| 04:12.09 | SeRi | I dont :) |
| 04:12.44 | SeRi | putting the last touches on buddies. |
| 04:13.27 | p3nguin | Isn't it amazing how that works once you enable it? |
| 04:13.41 | SeRi | rofl |
| 04:13.49 | SeRi | I suppose :P |
| 04:14.46 | SeRi | Its bee a crazy day today. |
| 04:15.50 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 04:20.15 | LiuYan | hi, I'm testing hot-desk example in Relational Database Integration chapter of 《Asterisk™: The Definitive Guide》, |
| 04:20.26 | LiuYan | I have a problem, the following function takes about 15 minutes to execute (maybe first connection to database): Set(${E}_STATUS=${HOTDESK_INFO(status,${E})}), I'm using unixODBC-2.2.14, and isql response at once when connect to same database. |
| 04:21.59 | LiuYan | is there a way to figure out why it takes such long time to connect database? |
| 04:24.01 | *** join/#asterisk lovetide (~lovetide@211.154.128.135) |
| 04:25.58 | Naikrovek | LiuYan: does the system load do anything during that 15 minutes? |
| 04:26.40 | Naikrovek | also, i know that the .rpm versions of unixODBC can cause issues. did you get that version from packages or did you compile it yourself |
| 04:27.21 | LiuYan | the system load is less than 2%(CPU usage) |
| 04:27.44 | Naikrovek | okay so the DB probably isn't doing much |
| 04:27.59 | LiuYan | I downloaded unixODBC-2.2.14-linux-x86-64.tar.gz from sourceforge.net, not compiled myself |
| 04:28.00 | Naikrovek | immediately i'm suspecting unixodbc but i don't know how to tell you to find out what it's doing |
| 04:28.21 | SeRi | p3nguin, what do you think about this system http://www.frys.com/product/5929124?site=sr:SEARCH:MAIN_RSLT_PG |
| 04:30.05 | LiuYan | Naikrovek: I tested the data source by /usr/bin/isql, it can response at once, so maybe it should not caused by unixODBC? |
| 04:30.13 | p3nguin | Make sure it is Linux compatible. I don't know anything about the VIA Hardware Security Engine. |
| 04:30.15 | Naikrovek | maybe not. |
| 04:30.34 | Naikrovek | check the asterisk log for the entire duration of the delay, i guess. |
| 04:30.53 | Naikrovek | also, if you can, see if you can find out what mysql is doing during that time. there must be a way to trace what it's doing |
| 04:31.12 | SeRi | p3nguin, Mhhhh good point. They claim they are Linux compatible. I will be loading Arch on it. |
| 04:31.35 | SeRi | BUT I had a Pico-ITX from VIA in the past and it was problematic. |
| 04:31.56 | LiuYan | Naikrovek: ok, thank, i'll check them, btw, i'm using PostgreSQL 9.1.1 for windows. |
| 04:32.25 | Naikrovek | i don't understand why asterisk even uses odbc... i understand the need for a common commandset for all major databases, but odbc? |
| 04:32.48 | SeRi | p3nguin, http://www.trustedreviews.com/VIA-ARTiGO-A2000-Barebones-Storage-Server_Peripheral_review |
| 04:32.56 | Naikrovek | LiuYan: ok. i'm pretty sure you can trace postgres and find out what it's doing. see if it receives an SQL command before that 15 minute wait starts |
| 04:34.02 | SeRi | p3nguin, are you running your Arch setup on HDD or Solid State IE: CF. |
| 04:34.57 | p3nguin | DoM |
| 04:35.11 | p3nguin | But I also run Arch on my laptop and my desktop systems. |
| 04:35.48 | p3nguin | I see that they say that box will run a few Linuxen. |
| 04:36.01 | p3nguin | It looks like a nice little box. |
| 04:36.39 | SeRi | p3nguin, I think I am going to roll the dice and pick one up tomorrow at my Local Fry's |
| 04:37.10 | SeRi | Migrate away from my alix to a better hardware. |
| 04:37.11 | p3nguin | If you're after a full-size SATA hard drive or two, that fits the bill... but I've had my eye on an Atom box which is intended to be used with laptop hard drive or SSD. |
| 04:37.33 | SeRi | p3nguin, well I just find it cheap. |
| 04:37.46 | SeRi | for 189.00 It just a steal I think |
| 04:37.58 | p3nguin | The Atom system is $179.something |
| 04:38.08 | SeRi | Where? |
| 04:38.13 | p3nguin | ebay |
| 04:38.31 | SeRi | Mhhhhh |
| 04:38.33 | SeRi | is looking |
| 04:39.54 | *** join/#asterisk radic (~radic@dslb-094-216-243-079.pools.arcor-ip.net) |
| 04:40.57 | p3nguin | I like the design of this VIA box. |
| 04:41.01 | SeRi | p3nguin, this? http://www.ebay.com/itm/Shuttle-XS35V2-Barebone-system-Atom-D525-NEW-/260834478771?pt=Desktop_PCs&hash=item3cbaf24ab3#ht_4985wt_1165 |
| 04:42.25 | p3nguin | It's not the exact one, but that's the type of system I was referring to. |
| 04:42.31 | SeRi | I see. |
| 04:42.45 | SeRi | Yea I like the design of the VIA box as well. |
| 04:43.02 | SeRi | But I think it only boots from solid state |
| 04:43.20 | SeRi | Not a problem with me though I have plenty of CF's. |
| 04:43.26 | SeRi | decisions! |
| 04:45.25 | p3nguin | Here's another: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=230688142048 |
| 04:46.46 | SeRi | Thats nice. Though I am not sure if Zotac supports linux. I all ways get the impression they are windows driven... |
| 04:50.40 | p3nguin | Also http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=320779012525 and http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=260687359643 |
| 04:53.32 | p3nguin | I kind of like the 320779012525 |
| 04:54.20 | SeRi | is looking |
| 04:56.02 | SeRi | d00d that one si nice! |
| 04:56.07 | SeRi | I like that too.http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=260687359643 |
| 04:56.18 | SeRi | no not that one |
| 04:56.34 | SeRi | That one is to bug. I dont have a big enough rack. |
| 04:56.39 | SeRi | big* |
| 04:56.42 | SeRi | I like this one |
| 04:56.43 | SeRi | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=320779012525 |
| 04:56.50 | SeRi | Perfect! and the price is awesome |
| 04:56.54 | p3nguin | For just $135.99+$20.24, it's not bad. |
| 04:57.02 | p3nguin | It's a single core hyper-thread Atom. |
| 04:58.06 | p3nguin | But if you need gigabit Ethernet, consider another model. |
| 04:58.13 | SeRi | I just saw that |
| 04:58.15 | SeRi | :/ |
| 04:58.31 | SeRi | gigE here |
| 04:59.07 | p3nguin | Might want to go for the Supermicro rack mount. |
| 04:59.59 | SeRi | It looks nice. and I like the specs. It wont fit though. You going to buy it? |
| 05:02.34 | p3nguin | This one is a little more expensive, but I like it a lot: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=110629553666 |
| 05:04.39 | SeRi | Indeed very nice! |
| 05:04.48 | p3nguin | I guess it's really the Morex 5677 industrial Mini-ITX enclosure that attracted me. |
| 05:05.25 | p3nguin | I should just buy the enclosure and be satisfied. :) |
| 05:05.31 | SeRi | LOL |
| 05:05.36 | SeRi | The case does look sick. |
| 05:05.40 | SeRi | brb |
| 05:06.59 | p3nguin | If I needed another computer, I might buy the enclosure to put something in. |
| 05:12.09 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 05:19.25 | SeRi | I have the Corsair Graphite 600 case and is the worst case ever, |
| 05:19.34 | SeRi | the whole damn thing makes noises |
| 05:19.46 | SeRi | I FUCKING HATE IT! |
| 05:20.04 | SeRi | I feel better :) |
| 05:21.07 | p3nguin | When metal things make noises that they shouldn't, there's always hot glue. |
| 05:21.51 | SeRi | lol nice idea |
| 05:23.18 | SeRi | I see the more expensive sip phones do have led notification for buddies. |
| 05:23.28 | p3nguin | A friend of mine had a pretty loud stereo system in his Cutlass, and every fucking thing on the car vibrated with the bass. He hot glued metal trim, tail lights, door handles, license plate, license plate light, and probably some stuff I didn't know a car had. |
| 05:24.22 | SeRi | hahahahaha! |
| 05:24.24 | SeRi | lmao |
| 05:24.40 | SeRi | gethorigit! |
| 05:24.42 | p3nguin | It worked, so I couldn't laugh too much. |
| 05:24.49 | SeRi | LMAO |
| 05:25.12 | SeRi | I can see the expression... wtf? :/ |
| 05:25.17 | SeRi | lol |
| 05:25.27 | p3nguin | He turned up the music and tracked down everything that vibrated until the outside of the car stopped buzzing. |
| 05:25.49 | p3nguin | He was a pretty funny guy. |
| 05:25.54 | SeRi | hahahahahaha that's hilarious! |
| 05:26.57 | p3nguin | He used to be a really heavy sleeper, and his alarm clock wouldn't wake him up to go to work, so he asked me to wire the alarm clock's speaker wires to a car stereo amp which was powered by a DC power supply. |
| 05:27.16 | p3nguin | So he ended up with a 10" 100W alarm clock. |
| 05:27.20 | SeRi | wtf! lol! |
| 05:27.35 | SeRi | hahaha damn! |
| 05:27.37 | SeRi | lol |
| 05:27.45 | SeRi | man thats just crazy lol |
| 05:27.52 | SeRi | good idea though |
| 05:27.53 | SeRi | lol |
| 05:29.25 | p3nguin | I'm trying to think of some of the other odd things he came up with. |
| 05:30.14 | SeRi | lol |
| 05:31.52 | p3nguin | If you think about it, the ideas are pretty ingenious, no matter how weird they are. |
| 05:33.30 | SeRi | They sure are. never ever in my life thought about hot glue. I can see how the hot glue would help as I am sure it can absorb the vibration |
| 05:35.19 | p3nguin | His best friend's car had something wrong with the windshield wipers... like the piece that connects to the motor came apart or something, so the wipers would not work by turning on the switch. The wipers were able to be moved back and forth by hand, though, so he tied a string to the tip of one wiper, ran it around and inside the car and back out the other window and tied it to the other wiper. So now they could control the ... |
| 05:35.25 | p3nguin | ... wipers from inside the car. The driver could pull one way and the passenger could pull the other way and it would wipe the windshield. |
| 05:36.31 | SeRi | hahahahahaha! |
| 05:36.36 | SeRi | Thats clasic! |
| 05:38.27 | *** join/#asterisk mattsl (~user@68.169.165.190) |
| 05:38.45 | SeRi | Thats just plain funny! lol! |
| 05:39.20 | mattsl | I'm having a couple, seemingly unrelated, problems. Anyone willing to try to help? |
| 05:39.26 | p3nguin | They did it because the dozen coats of Rain-X didn't help very much. |
| 05:39.35 | SeRi | hahahahaha! |
| 05:39.39 | SeRi | lol |
| 05:39.55 | p3nguin | They originally thought they'd just Rain-X the hell out of it and it wouldn't matter if they had wipers or not. |
| 05:41.09 | SeRi | I bet |
| 05:41.12 | p3nguin | A really funny thing was if he drove really fast, the wipers would fly straight upward. You could always tell when he's show up if he had obeyed the speed limit or not because the wipers would either be down or up. |
| 05:41.26 | SeRi | HAHAHAHA LOL! |
| 05:42.23 | mattsl | I can't get any SIP connections other than from my phones on my local network. I've tried allowing incoming anonymous SIP, setting up extensions that have Allow rules explicitly designed to allow the off-network IPs, and I've verified via tcpdump that the data is arriving at my box. My top suspect is NAT, but I've done everything I've been able to find to do that should let things work. However, it's quite possible I've missed something. |
| 05:43.05 | p3nguin | Did you forward the proper ports at the firewall? |
| 05:43.10 | mattsl | Yep |
| 05:43.25 | mattsl | 5060-5080 and 10000-20000 |
| 05:43.26 | p3nguin | UDP 5060 and whatever UDP range defined in rtp.conf? |
| 05:43.49 | mattsl | I double checked with tcpdump that the packets were making it to the server |
| 05:43.55 | p3nguin | SIP is just 5060, not 5061-5080. |
| 05:44.17 | p3nguin | Did you configure Asterisk to work behind NAT? |
| 05:44.19 | mattsl | Yeah. I read something someone put somewhere about the other ports, so I added them just in case |
| 05:44.25 | mattsl | I tried |
| 05:44.38 | mattsl | I'm guessing I missed something somewhere though |
| 05:45.01 | p3nguin | In the general section, nat=yes, directmedia=no, define a localnet and externhost or externaddr. |
| 05:45.46 | p3nguin | And then in each peer entry, if the device it behind a different NAT, you need to specify nat=yes and directmedia=no. |
| 05:45.49 | mattsl | I'm using FreePBX/PIAF on top. What is the best way to set those without screwing with the auto generated files? |
| 05:45.58 | p3nguin | ~freepbx |
| 05:45.58 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 05:46.35 | p3nguin | In other words, I have no flippin' idea. |
| 05:46.44 | mattsl | :-) |
| 05:46.46 | mattsl | Ok |
| 05:47.02 | mattsl | Thanks for the help. I |
| 05:47.31 | mattsl | I'll shoot over to the freepbx channel and if I can't find it, probably be back here to try to just manually change it in the conf |
| 05:47.59 | p3nguin | And then after you do, the first time you press the big orange Apply button, all your changes will be flushed away. |
| 05:48.10 | SeRi | ^^^ |
| 05:49.11 | mattsl | I know |
| 05:49.47 | mattsl | But if I can at least confirm that is the issue, then I can spend however long it takes to find out the "easy" way to do it. |
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| 05:57.08 | p3nguin | How's the success rate with D-Link plastic routers and SIP? |
| 05:57.39 | SeRi | I have no clue. Never used plastic routers :/ |
| 06:00.53 | p3nguin | What do you have? |
| 06:01.20 | p3nguin | I know you don't have Vyatta yet! |
| 06:01.35 | SeRi | I use pfsense |
| 06:01.58 | p3nguin | Oh yeah. |
| 06:02.03 | SeRi | I have not found the time to dig on to vyatta |
| 06:02.15 | SeRi | in* |
| 06:02.28 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-154-7.chyn.qwest.net) |
| 06:02.31 | SeRi | But its on my *to do* list |
| 06:02.52 | p3nguin | My to do list mysteriously got lost. |
| 06:03.23 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-asygfscntzmrfxoh) |
| 06:03.23 | SeRi | :/ |
| 06:03.26 | p3nguin | Now I don't do anything unless it's a spontaneous thought. |
| 06:03.29 | SeRi | lol |
| 06:03.41 | SeRi | I am like that about hardware |
| 06:03.58 | SeRi | as of lately I want to go away from mid towers to mini itx |
| 06:04.50 | p3nguin | Why the heck is it so bloody hot in here?! |
| 06:04.55 | p3nguin | I'm melting. |
| 06:05.08 | p3nguin | I have the AC set to 72. |
| 06:05.15 | SeRi | open the window. isnt it like 40 where you at? |
| 06:05.19 | p3nguin | Maybe I should lower it to 65. |
| 06:05.35 | SeRi | lol damn thats cold |
| 06:05.40 | p3nguin | It's in the 60s, but we have rain and high humidity right now. |
| 06:05.52 | SeRi | where you at? |
| 06:06.01 | p3nguin | southern IL |
| 06:06.04 | SeRi | It was the same over here |
| 06:06.07 | SeRi | ah! |
| 06:07.05 | SeRi | cool |
| 06:07.50 | p3nguin | I need to head out, anyway. |
| 06:08.51 | SeRi | cool. cya. ill be around for a bit more. working on my dial plan. |
| 06:09.15 | SeRi | and have a g/n |
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| 07:20.41 | schmidts | good morning |
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| 08:13.14 | wdoekes2 | good morning |
| 08:15.12 | irroot | wdoekes2 o/ |
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| 08:28.13 | schmidts | morning wdoekes2 |
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| 09:20.50 | Sakuranbo | hey guys ! |
| 09:21.27 | *** join/#asterisk _N1x (~n1x@mail.orient-logic.com) |
| 09:22.51 | _N1x | hi guys , i have one question , im testing my astrerisk with sipp stress test. and when i'm finishing my test and turn off sipp tool , i see in asterisk here is many sip channels , how i can fix dead sip channels? i have 1.6.2.9 version , thanks |
| 09:25.27 | schmidts | _N1x these sip channels should disappear after 32 seconds after the calls is hung up |
| 09:26.26 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 09:27.46 | _N1x | schmidts: im waited about 30 min |
| 09:27.51 | _N1x | but sip channels is still here |
| 09:30.07 | schmidts | _N1x try a sip show objects |
| 09:30.32 | schmidts | and look for the dialob objects and their refcounter values, is there one with a higher value than 3 or 4? |
| 09:30.54 | *** join/#asterisk hariom (~hariom@122.169.164.105) |
| 09:30.58 | _N1x | schmidts: now refcount is one |
| 09:31.06 | _N1x | but i have not sipp channels right now |
| 09:31.41 | schmidts | but this sounds like a refcounter problem cause it should be zero when the call is allready dead |
| 09:31.46 | schmidts | thats why its not removed |
| 09:32.07 | schmidts | why do you use such an old asterisk version? |
| 09:32.13 | hariom | I am testing a scenario where multiple sip calls comes to Asterisk and all incoming calls are dialled to the same sip id. How would asterisk handle that? Can asterisk handle all sip calls in parallel? |
| 09:32.23 | _N1x | schmidts: i use 1.6.2 version it is old? |
| 09:32.41 | schmidts | around 4 years i guess and btw its quit a mess |
| 09:32.58 | schmidts | you should try 1.8 atleast or 10 if you want to have some fun |
| 09:33.05 | _N1x | schmidts: and which is recommended? |
| 09:33.14 | _N1x | 1.5 is stable? |
| 09:33.23 | _N1x | *1.8 sry |
| 09:33.43 | hariom | To make my query more clear : [call 1, call 2, ... , call n] are dialed to the same sip id; say sip://1111@localhost is it possible to handle all the n calls in parallel? |
| 09:33.46 | schmidts | there is no "stable" asterisk version, only long term support releases like 1.8 but this has nothing to do with stable or unstalbe |
| 09:34.08 | schmidts | hariom how many calls do you want to have and what is on the other side |
| 09:34.22 | _N1x | schmidts: ok thanks , i'll install 1.8 can i update current version without uninstall install? |
| 09:34.25 | schmidts | _N1x 1.8 now is ready for production if this will answer your question |
| 09:34.50 | _N1x | schmidts: if this bug is fixed |
| 09:34.53 | _N1x | im ready to use 1.8 |
| 09:35.06 | hariom | schmidts: One side is n number of sip softphones dialing to the same sip id registered in sip.conf in asterisk. (n sip phones dialing the same sip id -> Asterisk) |
| 09:35.10 | schmidts | _N1x it depends on the OS but normally with linux you dont have to uninstall anything, just get the source of 1.8, compile it and when you do make install then the old version is removed |
| 09:35.43 | _N1x | ok |
| 09:36.16 | schmidts | hariom do they all want to dial the same extension or different extension on the same endpoint? and what should happen there? |
| 09:37.15 | irroot | hariom each call is in a new thread so it will be "parrellel" |
| 09:38.06 | hariom | schmidts: All dial to the same number registered on the asterisk server. When a new call comes, I want to play 'Hello World' to all of them. |
| 09:39.57 | hariom | irroot: thanks for the information |
| 09:41.06 | *** join/#asterisk garymc (~chatzilla@host81-139-152-192.in-addr.btopenworld.com) |
| 09:41.25 | irroot | schmidts _N1x i modify the Makefile to append -1.8/-10/-trunk onto the modules dir and edit asterisk.conf |
| 09:42.13 | hariom | irroot, schmidts: so in order to allow outsiders to call to my server and listen this hello world, do I need to configure for annoymous sip calls? |
| 09:45.51 | the_5th_wheel | Morning. I seem to have a strange issue with some AGI scripts. I am using asterisk 1.6.2. I know dead agi has been deprecated, so not sure if I need to change something for that to work nicely. What happens is when the far side puts the call down first, all is fine. When local side puts down, it doesnt finish off the calls. Anyone know how I can get past this? I thought perhaps it was a pcntl issue, but its compiled in and working. |
| 09:46.03 | irroot | hariom the term outsider implies untrusted you will need a default anon this is fine if you set up a context that cant leak billable or sensitive calls |
| 09:47.42 | hariom | irroot: ok |
| 09:58.06 | _N1x | schmidts: how i can install addons on asterisk 1.8? |
| 09:58.23 | _N1x | i can't see on the asterisk.org |
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| 10:02.26 | irroot | _N1x addons is included in the main package |
| 10:02.41 | irroot | make menuselect to enable / disable them |
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| 10:07.25 | *** join/#asterisk kjs (~kjs@fedora/kjs) |
| 10:08.01 | kjs | hi guys, is there a simple Asterisk client that will allow IM / Show me when another user is on a call? |
| 10:08.10 | kjs | and also act as a sip client if possible. |
| 10:09.49 | _N1x | irroot: hm i have enabled but cant see in module show |
| 10:10.16 | irroot | you built installed and checked modules.conf |
| 10:10.36 | irroot | ls /usr/lib/asterisk/modules/<module>.so |
| 10:10.58 | irroot | _N1x ^^ |
| 10:11.39 | _N1x | irroot: cdr_addon_mysql.so |
| 10:11.55 | irroot | _N1x its there ? |
| 10:12.16 | _N1x | irroot: yes , i need add load => cdr_addon_mysql.so in modules.conf |
| 10:12.18 | _N1x | yes? |
| 10:12.43 | irroot | _N1x if i could encourage you to use cdr_odbc_adaptive instead ?? its far better and can use mysql via ODBC |
| 10:13.05 | _N1x | irroot: what you menas? |
| 10:13.07 | _N1x | mean* |
| 10:13.24 | irroot | _N1x and if you "module load cdr_addon_mysql.so" on cli |
| 10:13.42 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 10:14.00 | irroot | _N1x the cdr_mysql is not so great and use via ODBC is encouraged |
| 10:14.09 | _N1x | irroot: http://pastie.org/2866300 |
| 10:14.10 | rmacd | This is probably a pretty dumb question - but I've got an in-only service attempting to connect to my asterisk instance. A quick tcpdump shows that asterisk's replying with a 404, cutting the call. Surely there's somewhere I can add the 'user' (that's what you call it?) for asterisk to proceed through a context? - apologies, just compiled and installed this afternoon. |
| 10:14.21 | irroot | it may not be cdr_addon_mysql could be cdr_mysql |
| 10:15.25 | irroot | _N1x that error is probably a indication that its the old version of the module |
| 10:15.46 | _N1x | irroot: i see in /usr/lib/asterisk/modules directory and cdr_addon_mysql.so is there |
| 10:15.56 | irroot | indeed its just "cdr_mysql.so" |
| 10:15.58 | ollii | is anyone here using t38modem ? |
| 10:16.06 | irroot | ollii indeed |
| 10:16.22 | irroot | rmacd default anon user ?? |
| 10:16.27 | *** join/#asterisk TimeRider (~steve@host213-122-111-242.range213-122.btcentralplus.com) |
| 10:16.55 | irroot | <PROTECTED> |
| 10:17.27 | ollii | stock asterisk 1.8 is not enough? |
| 10:17.31 | irroot | _N1x change to just cdr_mysql |
| 10:18.10 | ollii | but my problem occurs far earlier ;) are you using t38modem with ptlib and opal from source? |
| 10:18.28 | _N1x | irroot: No such file or directory |
| 10:18.34 | _N1x | hmm |
| 10:18.50 | irroot | ollii ptlib |
| 10:18.54 | irroot | not opal |
| 10:19.05 | _N1x | i tried cdr_mysql.so also but |
| 10:19.08 | _N1x | cant |
| 10:19.25 | irroot | _N1x check the source addons/ folder |
| 10:19.25 | *** join/#asterisk BuenGenio (~Gene@203.145.92.184) |
| 10:19.42 | irroot | ls addons/ |
| 10:19.46 | ollii | ah okay you use chan_ooh323 |
| 10:20.27 | ollii | which version of t38modem are you using? |
| 10:20.48 | irroot | ollii 2.0.0 |
| 10:21.14 | ollii | okay...so my is problem is that i want to use opal |
| 10:21.33 | _N1x | irroot: addons where? |
| 10:22.09 | irroot | i used it with h.323 during the work on t38 gateway to test ooh323 and fixup where needed |
| 10:22.25 | irroot | _N1x in the source folder ./addons |
| 10:23.18 | _N1x | irroot: here is cdr_mysql.o and cdr_mysql.so |
| 10:23.51 | _N1x | and app_mysql.so etc.. |
| 10:25.31 | irroot | but yet not in the /usr/lib/asterisk/modules |
| 10:25.36 | irroot | make install |
| 10:26.19 | irroot | _N1x should load once they in the folder either via modules.conf or manual take note of the name changes |
| 10:27.14 | *** join/#asterisk BuenGenio (~Gene@203.145.92.184) |
| 10:29.35 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 10:29.58 | *** join/#asterisk BuenGenio (~Gene@203.145.92.184) |
| 10:30.36 | ilj | hi! I'm trying to setup things in such a way that if no one answers the call in a queue a caller will be redirected to voicemail. Here's how I do it: http://pastebin.com/pqgxKq1q but the queue keeps ringing on an on, it would seem endlessly, even if I set Queue(myq,r,30) or Queue(myq,r,,,30) -- ... |
| 10:30.41 | ilj | ... I'm not sure whichever syntax is correct, but I tried both (what are those triple commas for anyway?). It'd be really great if someone experienced took a quick look at that paste. Maybe it's something obvious that I'm missing. |
| 10:32.00 | kaldemar | ilj: "core show application Queue" in CLI will give you the syntax. |
| 10:32.07 | irroot | ilj "core show application queue" |
| 10:32.32 | irroot | kaldemar ;) |
| 10:32.34 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 10:33.07 | kaldemar | irroot: it often echoes in here. :) |
| 10:33.46 | ilj | irroot, it's Queue after all, not queue ;) |
| 10:38.17 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 10:40.00 | kaldemar | ilj: remove the timeout and retry parameters from queues.conf |
| 10:45.09 | *** join/#asterisk _dmb (dmb@gateway/shell/yenn.ulegend.net/x-luoquknutkvfdinh) |
| 10:46.44 | _N1x | irroot: i do but ( No application 'MYSQL' for extension ) |
| 10:46.56 | _N1x | isee cdr_mysql.so in modules |
| 10:46.57 | _N1x | asterisk |
| 10:54.39 | _N1x | schmidts: i fixed and installed 1.8 but i have same problem , i have also dead sip channels when i'm turning of sipp tools |
| 10:54.44 | _N1x | sip channels is up |
| 10:54.50 | _N1x | and refcount is 2 each channel |
| 10:55.18 | _N1x | how i can fix it without restart asterisk everytime? :( |
| 10:55.56 | *** join/#asterisk kilgorex (~rob@82-69-63-23.dsl.in-addr.zen.co.uk) |
| 10:59.20 | schmidts | _N1x even after 30 seconds? |
| 11:01.22 | _N1x | schmidts: 443 active calls |
| 11:01.24 | _N1x | wait 30 sec |
| 11:03.00 | tompaw | Hey, I'm trying to install ztdummy module for my 1.8, but I'm getting confused: is it now a part of DAHDI project? |
| 11:03.28 | _N1x | schmidts: nothing happened 443 active channels |
| 11:03.29 | irroot | tompaw dahdi is all you need no more seperate dummy |
| 11:03.30 | schmidts | tompaw you will not need a ztdummy with 1.8 anymore, you can use timerfd instead |
| 11:03.30 | _N1x | is here |
| 11:04.02 | tompaw | schmidts: really? and this: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO will work, too? |
| 11:04.09 | irroot | tompaw if you just modprobe dahdi it will be there |
| 11:04.11 | schmidts | _N1x this looks like another problem if even the whole calls stay connected |
| 11:04.40 | irroot | _N1x using rtp timeout ? |
| 11:04.42 | schmidts | tompaw meetme can now run without dahdi dummy |
| 11:05.11 | _N1x | irroot: no i have not configed it , how to do? |
| 11:05.37 | _N1x | you |
| 11:05.38 | irroot | _N1x in sip.conf if the RTP dies the call will die |
| 11:05.44 | tompaw | schmidts: great news indeed, so I just build 1.8 and that's it? |
| 11:06.03 | _N1x | irroot: in detail? |
| 11:06.04 | schmidts | tompaw and you have to use confbridge instead of meetme but its nearly the same |
| 11:06.17 | schmidts | _N1x rtptimeout=10 for example |
| 11:06.58 | _N1x | schmidts: 10 is second or milliseconds? |
| 11:06.59 | *** join/#asterisk coppice (~chatzilla@14.63.101.23) |
| 11:07.12 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 11:07.43 | schmidts | AFAIK seconds |
| 11:08.42 | tompaw | schmidts: sounds neat! |
| 11:09.14 | irroot | indeed rtptimeout is seconds |
| 11:09.38 | schmidts | tompaw but using timerfd could be a load killer so if you expect to have a lot of concurrent calls you will get high load and also some locks ... |
| 11:10.31 | tompaw | schmidts: I'm planning to have no more than 20-30 3-way calls |
| 11:10.35 | _N1x | irroot: schmidts i do but not fixed ... |
| 11:10.46 | _N1x | what is real problem?... i read this is old asterisks bug |
| 11:10.55 | _N1x | but i see not fixed in 1.8 |
| 11:10.57 | tompaw | and it's a bi-xeon e560x something something with 48 gigs of ram |
| 11:11.08 | irroot | schmidts tompaw i found dahdi timer [dummy even] to be more reliable for sending faxes |
| 11:11.40 | tompaw | :-) no faxes needed, just the capability to turn a 2-way call into 3-way and back |
| 11:12.05 | schmidts | irroot yes thats for sure ;) |
| 11:12.26 | coppice | I didn't knwo some xeons were bi :-\ |
| 11:12.26 | schmidts | tompaw with 20 to 30 you will be fine, the problems start after 200 concurrent calls |
| 11:13.05 | tompaw | schmidts: the plan is to have 1500 no-transcode calls on freeswitch + 20-30 3-way conferences on asterisk |
| 11:14.04 | tompaw | I wonder if there are any examples of 2-way-to-3-way with confbridge out there... |
| 11:15.37 | rmacd | Any ideas as to why iax firmware would be 'unavailable'? What's that supposed to mean? "Error opening firmware directory '/opt/local/var/lib/asterisk/firmware/iax': No such file or directory" |
| 11:16.22 | coppice | tompaw: why do you want to mix Freeswitch and Asterisk? |
| 11:17.37 | tompaw | coppice: I want to have a predictive dialer which simply calls a number and plays g729 stream with no transcoding, for which I use fs with their native module |
| 11:18.05 | tompaw | coppice: as soon as receiving party presses a dtmf, it will bridge it to asterisk, which can handle queues and (hopefully) 3-way calls |
| 11:18.47 | coppice | tompaw: what's wrong with the queues and conferences in Freeswitch? |
| 11:20.20 | tompaw | coppice: to be honest I couldn't work out how to perform this 2-way-to-3-way action with FS, I "felt" it would be more natural for asterisk, since it's a pbx. |
| 11:25.27 | tompaw | Is there an equivalent of MEETME_EXIT_CONTEXT with confbridge? |
| 11:53.20 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 11:54.20 | elliot98 | inside the dialplan, how does one obtain the ip and port from an incoming SIP invite that has not been previously registered? |
| 11:56.39 | *** join/#asterisk _N1x (~n1x@mail.orient-logic.com) |
| 11:57.13 | _N1x | guys , can i use callweaver instead asterisk for fixed dead sip channels purpose? |
| 11:57.31 | _N1x | i use 1.8 but dont fixed dead sip channels |
| 11:58.02 | kaldemar | elliot98: core show function CHANNEL |
| 12:02.14 | *** join/#asterisk BuenGenio (~Gene@203.145.92.185) |
| 12:02.55 | elliot98 | kaldemar: thanks, but that only shows the ip address, how does one get the port as well? |
| 12:03.43 | irroot | _N1x what you mean "dead sip" what version you using i know of a issue with pickup earlier in 1.8 |
| 12:05.38 | *** part/#asterisk the_5th_wheel (~edd@tcs-gw.bulwer.thusa.net) |
| 12:05.47 | _N1x | irroot: im testing asterisk with sipp stress tool and when i making 1000 concurrent calls and when i'm stoping sipp tools to finish testing , i see in asterisk here is many sip (up) channels |
| 12:06.02 | _N1x | maybe 350 maybe 450 it's random |
| 12:06.14 | kaldemar | elliot98: it shows the port too in the uri field. use func CUT to get it. |
| 12:06.41 | _N1x | irroot: and i use 1.8 also |
| 12:06.45 | irroot | _N1x and are these calls bridged in a thread ? |
| 12:06.52 | irroot | 1.8.X ?? |
| 12:07.03 | _N1x | irroot: 1.8.7.1 |
| 12:07.30 | rmacd | Just trying to get a hang of the terminology here at the same time as put together a demo for a presentation this afternoon - bloody madness. I'm a fool. Trying to get some sort of bog-standard audio out to a stream, fifo or otherwise. Has to be picked up by a stdin. Any ideas? |
| 12:07.55 | _N1x | irroot: what you mean bridged in a thread? |
| 12:08.08 | irroot | _N1x show channel XXXX |
| 12:08.17 | irroot | of a dead channel |
| 12:08.17 | elliot98 | kaldemar: the URI may not be accurate if behind NAT |
| 12:08.25 | irroot | and see where in the dialplan it is |
| 12:08.34 | _N1x | irroot: yes i see from asterisk cli , core show calls and core show channels |
| 12:08.44 | irroot | what are these "hung" channels doing |
| 12:08.56 | _N1x | irroot: musiconhol |
| 12:08.57 | _N1x | d |
| 12:09.21 | irroot | music on hold will hang arround for ever |
| 12:09.25 | kaldemar | elliot98: where do you see the information? |
| 12:09.38 | irroot | soft hangup SIP/XXXXX |
| 12:09.55 | _N1x | irroot: yes i know but when im turning off my phone from power supply |
| 12:09.59 | irroot | one of the channels it will hangup therefore it is not a dead hung channel |
| 12:10.03 | _N1x | musiconhold is on forever? |
| 12:10.43 | elliot98 | kaldemar: what do you mean? |
| 12:11.05 | kaldemar | elliot98: where is the reliable information then? |
| 12:11.08 | irroot | it can be depending on the circumstance change the dialplan to use wait(20) not musiconhold |
| 12:11.32 | irroot | and see after 20s they will be gone if so the problem is with the test method |
| 12:12.32 | _N1x | irroot: maybe yes but if i need musiconhold in real , what can i do? |
| 12:12.43 | _N1x | i dont need dead sip channels in my asterisk |
| 12:12.46 | _N1x | for performance |
| 12:12.47 | elliot98 | kaldemar: on the UDP packet |
| 12:13.04 | irroot | well when a channel is on hold and the user hangs up the channel stops |
| 12:13.17 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-228-247-87.inter.net.il) |
| 12:13.21 | elliot98 | kaldemar: a device behind NAT needs to use the UDP ip/port |
| 12:14.01 | _N1x | irroot: yes should channel stops |
| 12:14.04 | irroot | testing it the way you are is not a realistic situation its a stress situation |
| 12:14.06 | _N1x | but i see dont stops |
| 12:14.44 | irroot | you need to ensure your test case is valid and that the hangup is been sent from the testing |
| 12:15.21 | _N1x | irroot: i need simulate when hangup is dont sended |
| 12:16.00 | file | then you need to enable RTP timeout in sip.conf so when no RTP is received Asterisk will hang up the channel or if all your parties support it then you can enable SIP session timers and rely on that |
| 12:16.04 | irroot | but then make sure the test is not influenced by the method |
| 12:17.00 | irroot | _N1x ^^ file ^ n |
| 12:17.17 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 12:17.30 | _N1x | irroot: what you mean? |
| 12:17.41 | file | _N1x, what I said is for you |
| 12:17.55 | irroot | see what file says above |
| 12:18.14 | kaldemar | elliot98: i guess you're out of luck then. the port in the Contact or Via header is all you get. |
| 12:18.22 | _N1x | file: i have rtptimeout=10 in |
| 12:18.27 | _N1x | sip.conf but.... |
| 12:19.38 | irroot | _N1x also a use the duration option on MusicOnHold its only polite after some time to handle it in a IVR |
| 12:21.46 | elliot98 | kaldemar: there must be a way, since the NAT function is able to retrieve the values |
| 12:33.00 | *** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-228-247-87.inter.net.il) |
| 12:33.18 | kaldemar | elliot98: all information that the channel driver has is not necessarily available in the dialplan. |
| 12:38.11 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 12:47.10 | tompaw | Guys, is anyone using this: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ? |
| 12:48.06 | tompaw | I changed meetme to confbridge and everything works fine apart from one thing - at the end of [macro-nway-start] there is a Goto, which pretty much kills the A-leg of the call (C and B are happily chatting) |
| 12:55.18 | *** join/#asterisk francisvgarcia (~francis.g@190.80.239.124) |
| 13:03.34 | tompaw | Is there a way to have DYNAMIC_FEATURES on Agent calls? |
| 13:03.59 | tompaw | For example: I want an Agent who's on a queue call to be able to perform DTMF actions defined in DYNAMIC_FEATURES, is it possible? |
| 13:08.07 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 13:10.52 | *** join/#asterisk AmirBehzad (~behzad@31.184.187.1) |
| 13:11.35 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:14.21 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:19.07 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 13:29.25 | *** part/#asterisk catphish (~catphish@2001:9d8:2005:11:222:15ff:fe88:aae2) |
| 13:31.01 | ilj | kaldemar, sorry, I was occupied with other things... removing retry and timeout from queues.config didn't help: http://pastebin.com/LU8jTQu3 it just keeps ringing on and on until I hang up (to be honest I didn't wait for very long, not even 5 minutes but real callers won't wait that long either ... |
| 13:31.06 | ilj | ... so it wouldn't be of help if queue returned to dialplan after 5+ minutes anyway) |
| 13:32.06 | kaldemar | ilj: you have no timeout there. |
| 13:32.48 | [TK]D-Fender | ilj, "core show application timeout" |
| 13:33.10 | [TK]D-Fender | ilj, Your queues.conf is for inter-agent timeo, not "total time you can sit around waiting to get answered" |
| 13:33.35 | [TK]D-Fender | ilj, you set the max time in your dialplan app call |
| 13:35.27 | ilj | [TK]D-Fender, like this: Queue(somequeue,r,,,30) right? |
| 13:35.39 | [TK]D-Fender | ilj, In there somewhere... |
| 13:37.43 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 13:38.55 | kaldemar | ilj: yes, it is right as "core show application Queue" will tell you. |
| 13:41.08 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 13:41.18 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 13:41.32 | elliot98 | is it possible to register a peer from within Asterisk without the client needing to do so? |
| 13:44.13 | irroot | elliot98 that does not make sense perhaps you refering to accepting calls with out registration this can be done if the client is on a fixed ip address and this is configured in sip.conf |
| 13:44.31 | [TK]D-Fender | elliot98, No. |
| 13:45.09 | [TK]D-Fender | elliot98, But as irroot said you could just leave it as a fixed IP. |
| 13:48.29 | ilj | kaldemar, [TK]D-Fender damn, I think I know why it failed on me before. I think I've left Answer out and expected Voicemail to step in right after Queue. It works now, doesn't matter if I have retry and timeout options in queues.conf. Thanks a lot guys. |
| 13:49.21 | *** join/#asterisk jkroon (~jkroon@dsl-242-6-113.telkomadsl.co.za) |
| 13:49.27 | elliot98 | [TK]D-Fender: the reason is that a particular ip does not register, but needs to be qualified |
| 13:49.28 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-lshkhlezwvuslohn) |
| 13:49.34 | [TK]D-Fender | ilj, IIRC you do need a timeout (not sure if there is a default). should be that it will only check the overall Queue timeout between agent dials. |
| 13:49.36 | jkroon | hi guys, how long can I rasonably expect sip to take to become usable on a fresh restart? |
| 13:49.50 | elliot98 | [TK]D-Fender: it seems without registration, Asterisk does not send OPTIONS packets |
| 13:49.57 | [TK]D-Fender | elliot98, Incorrect |
| 13:50.09 | [TK]D-Fender | elliot98, If * doesn't have a target it won't |
| 13:50.10 | [TK]D-Fender | ^ |
| 13:50.26 | elliot98 | [TK]D-Fender: so by setting the host and port, things should be ok |
| 13:50.48 | [TK]D-Fender | elliot98, Depending on your client. |
| 13:51.20 | elliot98 | [TK]D-Fender: tcpdump doesn't show any OPTIONS packets going through |
| 13:51.50 | [TK]D-Fender | * SIP debug only, not external. |
| 13:51.54 | [TK]D-Fender | elliot98, Show us all of it |
| 13:52.57 | elliot98 | the system is using realtime, btw |
| 13:53.46 | elliot98 | what's the address for pastie? |
| 13:54.01 | kaldemar | ~pb |
| 13:54.01 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:55.10 | SeRi | g/m all |
| 13:56.57 | elliot98 | is it necessary to do a sip reload with realtime? |
| 13:57.12 | bulkorok | no |
| 13:57.27 | bulkorok | it's called realtime, isn't it!? |
| 13:57.55 | [TK]D-Fender | Somehow I doubt it will pick up changes on the fly for outbound bits like that |
| 13:58.07 | [TK]D-Fender | It's have to poll constantly. Sounds insane |
| 13:58.13 | [TK]D-Fender | it'd |
| 13:58.59 | ilj | [TK]D-Fender, okay, I'll test that. |
| 13:59.49 | jkroon | ok, so asterisk restart clocked in at 22 minutes (seemingly SIP getting stuck somewhere) |
| 13:59.54 | elliot98 | bulkorok: so why do things not seem to update until a sip reload is done? |
| 14:00.35 | irroot | jkroon realtime ?? |
| 14:00.45 | jkroon | irroot, nope. |
| 14:00.47 | irroot | dns lookups ? |
| 14:00.55 | bulkorok | elliot98: if you change settings in [general] of sip.conf you surely need a sip reload. |
| 14:01.01 | jkroon | i hope not, but that would make a strange amount of sense. |
| 14:01.34 | [TK]D-Fender | bulkorok, We aren't talking about [general] |
| 14:01.44 | bulkorok | just a guess |
| 14:01.50 | elliot98 | bulkorok: I'm talking about changes in the database |
| 14:01.55 | bulkorok | k |
| 14:01.57 | jkroon | irroot, got three explicit host= lines in sip.conf, one of which is a DNS address, and it returns immediately with NXDOMAIN. |
| 14:02.43 | irroot | jkroon 22min is exesive and strange |
| 14:02.52 | jkroon | indeed. |
| 14:03.36 | jkroon | and a significant amount of lost revenue and bad name. |
| 14:03.39 | irroot | jkroon a asterisk -cv....d..... |
| 14:04.06 | irroot | any clues as to where it sticking maybe work on it after beer 'o clock |
| 14:04.24 | jkroon | how do I go about finding where? |
| 14:04.47 | jkroon | quessing a LATE NIGHT restart and seeing if the same symptoms crop up again is an option ... and then grabbing core dumps. |
| 14:04.54 | jkroon | restart was initiated by a crash anyway. |
| 14:05.15 | irroot | it quite possibly is not sip look at the debug messages till they stop and work back from there also minimalise the config ... |
| 14:05.34 | irroot | add all known hosts to /etc/hosts as a rule |
| 14:05.57 | jkroon | i believe in minimalistic configs to begin with, but this monster is getting to a few thousand lines now. |
| 14:08.23 | ilj | holly molly! jkroon a few thousand lines? what is that PBX for? |
| 14:10.55 | elliot98 | [TK]D-Fender: here is the peer settings http://pastebin.com/HkVRxEde |
| 14:11.06 | elliot98 | [TK]D-Fender: sip debug doesn't state any OPTIONS packest |
| 14:11.19 | elliot98 | [TK]D-Fender: again, this is a realtime peer |
| 14:11.57 | [TK]D-Fender | Status : UNREACHABLE |
| 14:12.29 | [TK]D-Fender | elliot98, What is your target and where is it precisely? |
| 14:12.54 | [TK]D-Fender | elliot98, and if you have a host why do you need qualify at all? |
| 14:13.05 | elliot98 | it is behind nat, so to keep to port opened |
| 14:13.20 | jkroon | ilj, it's a voice switch. |
| 14:13.22 | [TK]D-Fender | elliot98, This has and will fail. Hard. Every time |
| 14:13.30 | [TK]D-Fender | elliot98, This will not punch a new hole open |
| 14:13.44 | elliot98 | a hole is already opened |
| 14:13.45 | jkroon | <PROTECTED> |
| 14:13.47 | [TK]D-Fender | elliot98, it can only keep one alive. |
| 14:13.51 | [TK]D-Fender | elliot98, Apparently not |
| 14:14.00 | [TK]D-Fender | elliot98, What is the client? |
| 14:14.04 | jkroon | sip show peers => 411 sip peers |
| 14:14.10 | [TK]D-Fender | elliot98, And why can't it register like normal? |
| 14:14.23 | elliot98 | but why doesn't sip debug show any OPTIONS? |
| 14:14.30 | [TK]D-Fender | elliot98, IT GAVE UP |
| 14:14.49 | elliot98 | is there any way to make it not give up |
| 14:15.07 | elliot98 | basically try until a hole opens up again |
| 14:15.26 | [TK]D-Fender | elliot98, Fix your client. forward ports on the remote. Pick one. |
| 14:16.52 | elliot98 | it's a bit of a complicated setup, Asterisk would need to qualify on its own |
| 14:17.13 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 14:17.29 | elliot98 | so Asterisk stops sending if UNREACHABLE |
| 14:18.01 | jkroon | elliot98, there is a setting that affects how many times asterisk tries to reregister, and it stops doing OPTIONS when the registration expires (ie, UNREACHABLE, otherwise it'll simply show LAGGED) |
| 14:18.10 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 14:18.29 | elliot98 | where is the setting? |
| 14:19.50 | elliot98 | is it compile time setting or in a conf file? |
| 14:20.05 | jkroon | in sip.conf: registerattempts=0 |
| 14:21.31 | elliot98 | registerattempts is for outpound regisration, no? |
| 14:21.39 | jkroon | that's what I said. |
| 14:21.45 | *** join/#asterisk LiuYan1 (~LiuYan@222.125.132.191) |
| 14:21.51 | elliot98 | the issue is that asterisk needs to qualify a device |
| 14:22.08 | elliot98 | ie. send OPTIONS to a particular device, even if the device has not registered |
| 14:22.24 | jkroon | then you need to do host= in the peer definition. |
| 14:29.22 | *** join/#asterisk Katty (~Katty@mail.copi-rite.com) |
| 14:29.57 | Katty | it's TUESDAY! |
| 14:30.36 | jkroon | ... still three more to go |
| 14:30.39 | elliot98 | jkroon: yes, but when host is set, eventually, it states UNREACHABLE |
| 14:30.50 | elliot98 | jkroon: and doesn't send OPTIONS anymore |
| 14:31.02 | jkroon | elliot98, that's weird. that shouldn't happen. are you sure the frequency doesn't just reduce? |
| 14:31.42 | [TK]D-Fender | elliot98, Fix your client. Forward ports. Pick one. |
| 14:34.53 | SeRi | [TK]D-Fender, I was reading on the Polycom SoundPoint Devices and to what they say BLF should turn on the red led on the top my phone.... but mine does not. bw is working and display all correct presences without and issue but the led does not light up.... Is this true? |
| 14:35.11 | SeRi | an* |
| 14:35.17 | [TK]D-Fender | serNo, MWI = MWI, not BLF |
| 14:35.24 | [TK]D-Fender | SeRi, ^ |
| 14:35.26 | Naikrovek | "on the top of [your] phone" no. the lights next to the line buttons |
| 14:35.36 | SeRi | Ok. Thats what I thought. |
| 14:35.37 | *** join/#asterisk Ionic (ionic@ionic.de) |
| 14:35.39 | SeRi | Thanks guys |
| 14:35.46 | *** join/#asterisk grandpapadot (~grandpapa@99.175.248.81) |
| 14:35.51 | SeRi | I think this was for the newer models :) |
| 14:36.05 | SeRi | off to work. Thanks guys! |
| 14:36.38 | irroot | jkroon RT FTW |
| 14:37.34 | *** join/#asterisk catphish (~catphish@2001:9d8:2005:11:222:15ff:fe88:aae2) |
| 14:38.08 | jkroon | irroot, yea, realized i'll have to move there at some point. |
| 14:38.27 | jkroon | desember project though, not something I'll manage overnight. |
| 14:39.29 | jkroon | and realtime brings it's own issues for me as well, i often have one entry in the database for up to around 8 peers ending up in the world. |
| 14:40.15 | jkroon | which means I'll need views to map things sanely, and views aren't updatable, so a restart of asterisk causes all the peer information to go lost. |
| 14:48.09 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 14:48.51 | irroot | jkroon mmm there is a bit of work to be done |
| 14:51.27 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 14:51.29 | jkroon | yes, like hacking asterisk to store the realtime data in astdb instead of the source db, if possible. |
| 14:51.43 | catphish | is there anything that would cause a phone to give up registering after receiving a 401? |
| 14:52.01 | catphish | (instead of sending authentication details) |
| 14:52.16 | jkroon | a bug in the phone's firmware? |
| 14:52.44 | catphish | perhaps, its a SPA504G |
| 14:53.05 | catphish | i'm more inclined to blame a problem with data being modified by the nat at the moment |
| 14:53.09 | catphish | its an unusual problem though |
| 14:54.01 | jkroon | what's the NAT device? |
| 14:54.31 | catphish | Vigor 2830 |
| 14:54.51 | catphish | customer tells me ALG is disabled by defaukt |
| 14:55.41 | [TK]D-Fender | Better not be the VN model.. |
| 14:57.12 | *** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk) |
| 14:57.51 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:03.16 | catphish | i'm not sure exactly what's happening, i appear to receive an unmolested REGISTER from the handset, send back a 401 with a nonce, then nothing |
| 15:04.35 | [TK]D-Fender | catphish, pastebin the complete SIP debug and your peer masking only the secret |
| 15:04.42 | catphish | sure |
| 15:05.05 | catphish | is it possible to sip debug an individual peer? |
| 15:05.21 | catphish | oh yeah |
| 15:05.22 | catphish | got it |
| 15:05.41 | catphish | hmm, can't because asterisk doesn't know the IP of the peer |
| 15:09.38 | catphish | [TK]D-Fender: http://paste.codebasehq.com/pastes/8qxo6knhju46i34ixb |
| 15:09.44 | [TK]D-Fender | It's not supposed to.. that's why it's registering. |
| 15:09.51 | catphish | indeed |
| 15:10.09 | catphish | there's a packet trace anyway, afraid it's not direction coded |
| 15:10.19 | [TK]D-Fender | Where's the peer? |
| 15:10.34 | catphish | what do you mean? |
| 15:10.38 | [TK]D-Fender | SIP.CONF |
| 15:10.43 | [TK]D-Fender | and please use * SIP debug. |
| 15:10.47 | [TK]D-Fender | not an external dump |
| 15:10.54 | catphish | it's realtime |
| 15:10.56 | [TK]D-Fender | new PB with the proper bits... |
| 15:11.10 | [TK]D-Fender | dump it |
| 15:13.19 | catphish | http://paste.codebasehq.com/pastes/0iznty3c8v05gk0md3 |
| 15:14.09 | catphish | http://paste.codebasehq.com/pastes/rktoi1ksonbt67zyoc |
| 15:14.56 | catphish | ooh, you can sip debug by IP :D |
| 15:16.20 | catphish | though the output is no more useful than my wireshark dump sadly |
| 15:16.46 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 15:18.27 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 15:20.32 | tompaw | Hey, can DYNAMIC_FEATURES be used inside a ConfBridge? |
| 15:22.48 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:26.14 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 15:28.47 | [TK]D-Fender | tomnest the call inside a Local channel dial |
| 15:28.49 | *** join/#asterisk sbszulu (~dundubala@41.52.155.203) |
| 15:28.56 | [TK]D-Fender | tompaw, nest the call inside a Local channel dial |
| 15:29.08 | [TK]D-Fender | catphish, Odd... |
| 15:30.18 | catphish | the phone is using STUN, hence the correct external IP and port in the Via header |
| 15:30.47 | tompaw | [TK]D-Fender: excuse me, but I don't think I follow... I checked the logs and once a participant is inside the conference it looks like it doesn't understand the DTMFs anymore, or at least doesn't parse them via applicationmap layer. |
| 15:31.18 | [TK]D-Fender | tompaw, It doesn't That's why I told you to nest it inside a Local channel Dial. |
| 15:31.34 | [TK]D-Fender | catphish, * SIP debug if you would.... |
| 15:31.50 | catphish | sure |
| 15:32.49 | vader-- | can you guys recommend a half height pci card that has 2-3 FXO and 1 FXS or a USB channel bank that does... Something that is hopefully in the 100$ range? |
| 15:32.51 | catphish | [TK]D-Fender: http://paste.codebasehq.com/pastes/cztq4p4kqabm6v8bom |
| 15:33.26 | [TK]D-Fender | vader--, You had us... until "$100" |
| 15:33.37 | catphish | my theory is that the customer is mistaken about ALG and his router is trying to un-nat the 401 response |
| 15:33.48 | tompaw | [TK]D-Fender: do you mean at the moment of joining the conference by myself? I do it by ChannelRedirect(${CHANNEL},dynamic-nway,${CONFNO},1) - is that the place where I should do the wrapping? |
| 15:34.13 | [TK]D-Fender | catphish, the NAT bit doesn't seem to be an issue. The return = the contact anyway... |
| 15:34.14 | tompaw | (dynamic-nway is a context where I am ConfBridge'd) |
| 15:34.21 | *** join/#asterisk highvoltz (rogers@bling.bling.org) |
| 15:34.49 | [TK]D-Fender | catphish, Small thought : change the type to peer. |
| 15:35.44 | [TK]D-Fender | tompaw, do an inner dial to the Confbridge. |
| 15:35.49 | catphish | apparantly this phone works for about 18 hours, then finds its way into this state until somebody gives it a new internal IP |
| 15:36.34 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 15:36.35 | wcselby | o/ |
| 15:36.44 | [TK]D-Fender | \o |
| 15:37.01 | catphish | lol |
| 15:37.05 | wcselby | what's going on? |
| 15:37.07 | tompaw | [TK]D-Fender: so Dial(extension_with_confbridge) instead of ConfBridge? |
| 15:37.10 | catphish | i mean \o/ |
| 15:37.23 | [TK]D-Fender | tompaw, not "instead of", "in addition to" |
| 15:37.31 | tompaw | well of course :-) |
| 15:37.33 | tompaw | thanks, will try that now |
| 15:39.22 | vader-- | so what would i be looking at for a FXO/FXS card? |
| 15:39.42 | catphish | [TK]D-Fender: i've run into all kinds of SIP nastiness recently, so far all resolvable by having customers disable ALG |
| 15:39.54 | Qwell | ALG should always be disabled |
| 15:40.01 | catphish | indeed |
| 15:40.05 | *** join/#asterisk fIorz (nobody@2001:1a50:503c::1) |
| 15:40.07 | *** join/#asterisk ks3 (~ks3@74.203.195.1) |
| 15:40.11 | Qwell | it's broken by design |
| 15:40.17 | catphish | sadly lots of devices have it enabled by default |
| 15:40.31 | catphish | Qwell: that rather depends on what the ALG is doing |
| 15:40.33 | [TK]D-Fender | vader--, Sangoma B600d |
| 15:40.38 | catphish | but yes, most appear to be horribly broken |
| 15:41.17 | vader-- | ouch too much |
| 15:44.00 | [TK]D-Fender | vader--, http://www.voipsupply.com/sangoma-usbfxo + ATA |
| 15:44.27 | tompaw | do we have a pastebin? |
| 15:44.48 | [TK]D-Fender | ~pb |
| 15:44.48 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 15:45.35 | vader-- | TK thanks for the link. Can more than one of these be used in a system at a time? |
| 15:45.53 | [TK]D-Fender | vader--, Please read the link.. |
| 15:45.57 | [TK]D-Fender | And product specs... |
| 15:46.26 | vader-- | i did |
| 15:46.30 | [TK]D-Fender | "The Sangoma U1-- is a low-cost development tool kit, ideal for those just getting started with Asterisk® or those who just want a simple external solution. The USBfxo is easy to install. There is no need to open up the computer to insert an interface card. Simply plug it in, and set up your Asterisk system; configure Sangoma's Wanpipe drivers, as you would with any Sangoma card, and you're set to go. You can even connect multiple USBfxos in one |
| 15:46.30 | [TK]D-Fender | <PROTECTED> |
| 15:46.34 | [TK]D-Fender | vader--, Apparently not |
| 15:46.37 | vader-- | mentions nothing about stack them |
| 15:46.45 | [TK]D-Fender | "You can even connect multiple USBfxos in one system. |
| 15:46.45 | [TK]D-Fender | " <- |
| 15:46.59 | vader-- | :-/ |
| 15:47.48 | wcselby | lol |
| 15:48.04 | wcselby | [TK]D-Fender reading all the way to the end is for suckers |
| 15:48.25 | wcselby | ;) |
| 15:48.30 | [TK]D-Fender | wcselby, It was right in the MIDDLE |
| 15:48.37 | wcselby | :) |
| 15:49.43 | wcselby | completely off topic, but anyone in here familiar with BGP? |
| 15:50.13 | ks3 | wcselby: Yes |
| 15:50.52 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:50.53 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:51.51 | *** join/#asterisk bitbandit (~taggmcd@c-98-202-116-168.hsd1.ut.comcast.net) |
| 15:53.50 | catphish | [TK]D-Fender: that natted device definitely has no ALG running :| |
| 15:54.14 | catphish | so i'm clinically certified confused |
| 15:54.41 | [TK]D-Fender | catphish, did you change they type as I suggested? do you have multiple devices behind that router? |
| 15:54.50 | catphish | they do, yes |
| 15:55.13 | catphish | and no, i havent tried that |
| 15:55.48 | [TK]D-Fender | catphish, my suggestion might actually break things in that case, but give it a shot. Next suggestion, run a small proxy on-site |
| 15:57.09 | catphish | [TK]D-Fender: we have a large number of low value customers connecting from NATs, so unfortunately proxying isn't an option, though a phone vpn client may be an option |
| 15:57.55 | [TK]D-Fender | catphish, Are they all as problematic as this one? |
| 15:58.22 | catphish | no, but a lot have run into problems |
| 15:58.29 | catphish | mostly solved by disabling ALG |
| 15:58.42 | catphish | but that's still an irritating support process |
| 15:59.03 | catphish | i'm really looking for the myththical nat solve-all |
| 16:00.19 | [TK]D-Fender | "Don't Do It" |
| 16:00.23 | [TK]D-Fender | :) |
| 16:01.28 | catphish | i'm genuinely confused as to why nobody has replaced sip with a simple symmetric trunk protocol |
| 16:01.50 | catphish | a la IAX |
| 16:02.01 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 16:02.27 | p3nguin | Some of us have. |
| 16:03.52 | catphish | is there any hardware support for anything like that? |
| 16:04.04 | catphish | or a vpn even? |
| 16:04.23 | p3nguin | There's nothing stopping you from running any VoIP protocol over a VPN. |
| 16:04.52 | catphish | i know, my question was one of hardware support |
| 16:05.36 | wcselby | some snom phones come with built-in openvpn support |
| 16:05.58 | wcselby | or you could find some cheap ipsec router and send that to the clients and tell them to use that to tunnel back into your network, place all the phone behind the vpn router, etc |
| 16:06.11 | catphish | thats true |
| 16:06.51 | catphish | i'm so puzzled by why this phone appears to ignore a friendly 401 |
| 16:08.43 | catphish | is there any reason why multiple devices behind the same nat causes issues? |
| 16:10.11 | p3nguin | You could deploy Vyatta all over the place. It supports ipsec, l2tp, and pptp style vpns. |
| 16:10.24 | catphish | i'm contemplating listening on not 5060, as that will disable ALG swiftly |
| 16:10.40 | p3nguin | A common alternate port is 5080. Go for it. |
| 16:10.48 | *** join/#asterisk furlonda (~crash@194.193.169.242) |
| 16:11.37 | furlonda | anyone know what the new dialplan application is for 1.8 that was "AgentCallbackLogin" in 1.4 and 1.6 i know its not in 1.8 so how do i allow users to log into roaming extensions |
| 16:11.53 | furlonda | 1.4 not 1.6 sorry* |
| 16:12.05 | Qwell | furlonda: it's all done in dialplan now |
| 16:12.10 | Qwell | it's documented somewhere.. hang on |
| 16:12.25 | furlonda | thats what i suspected :( |
| 16:12.48 | Qwell | the new way is *vastly* superior to the garbage that was agentcallbacklogin |
| 16:13.07 | furlonda | thank god lol |
| 16:13.34 | p3nguin | Superior doesn't necessarily mean easier to set up. |
| 16:13.37 | wcselby | https://wiki.asterisk.org/wiki/display/AST/Application_AddQueueMember |
| 16:13.41 | wcselby | furlonda ^^ |
| 16:14.19 | irroot | cheers folks hometime BBL |
| 16:14.32 | furlonda | thanks for the link wcselby |
| 16:14.48 | wcselby | np |
| 16:15.16 | wcselby | if you search leifmadsen 's blog, he's got some configuration examples of ways to use it, but I think that was prior to 1.8 where they added some extra options to the application that makes things easier now |
| 16:16.00 | [TK]D-Fender | LONG ago |
| 16:16.14 | furlonda | so by using this in the dialplan i can allow a user to log in say 420 and password of 1111 then put the phone down another user of the system can call 420 not like "AgentLogin" feature in 1.8 where the music on hold music is played |
| 16:16.48 | wcselby | furlonda right, this app is sort of like a dynamic agentcallbacklogin, but without using chan_agent |
| 16:17.02 | furlonda | i see |
| 16:17.34 | wcselby | this app allows you setup one login extension, the user can enter whatever their username / password should be, input those into AQM as variables, and you're good |
| 16:18.01 | [TK]D-Fender | furlonda, that is pure dialplan then. Chan_agent = toast |
| 16:18.44 | wcselby | so it's more than just one line of dialplan |
| 16:18.54 | wcselby | you'll have to Read() in variables and stuff |
| 16:18.55 | [TK]D-Fender | furlonda, make your own login scripts and store the extension > device map in something like AstDB, etc |
| 16:19.16 | [TK]D-Fender | furlonda, And your extensions will look up the associated device |
| 16:20.27 | furlonda | has anyone got an example of this in dialplan form just so a user dials 2199 to login is asked for extension number and password and then is logged in then 2198 to logoff |
| 16:20.38 | tompaw | [TK]D-Fender: your idea worked :-) Now the question is: how do I add another party to the conference from inside? |
| 16:21.00 | wcselby | furlonda i don't have one done but I could make one up for you quick and dirty |
| 16:21.24 | tompaw | I can run a macro via applicationmap, but 1) I need to somehow pass the ${EXTEN} to that macro, no idea how. 2) How do I invite a remote party from a macro without leaving the conference? |
| 16:21.38 | furlonda | its just something to start from then i can play im so used to 1.4 that we're running at the moment |
| 16:24.07 | *** part/#asterisk AmirBehzad (~behzad@31.184.187.1) |
| 16:24.08 | p3nguin | tompaw: ${MACRO_EXTEN} and ${ARG1} |
| 16:24.35 | p3nguin | (the answer to part 1) |
| 16:25.04 | [TK]D-Fender | tompaw, Launch the invite from another call or quit the conf to do it and rejoin after |
| 16:25.42 | wcselby | ~pb |
| 16:25.42 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 16:25.43 | tompaw | [TK]D-Fender: since the whole thing is web based, I was thinking about using the manager for these "external" invites. |
| 16:25.58 | tompaw | "quit the conf and rejoin" sounds scary :F |
| 16:25.59 | [TK]D-Fender | tompaw, Sure |
| 16:26.15 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 16:26.39 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 16:27.20 | tompaw | p3nguin: will ${MACRO_EXTEN} survive applicationmap? |
| 16:27.49 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 16:27.55 | wcselby | furlonda here's a rough outline, you'll need to clean it up and confirm channel variables and the like - http://pastebin.com/gCDs8ynM |
| 16:27.58 | p3nguin | No clue. Pass it as ${ARG1} to the macro. |
| 16:27.59 | [TK]D-Fender | tompaw, Not sure it has a chance of besing set in there at all |
| 16:28.13 | wcselby | i'm pretty sure I put the wrong variable in there on the interface option, but you can play around with it |
| 16:29.09 | [TK]D-Fender | wcselby, I don't think it has registered with you. HotDesking. Not queues. |
| 16:29.31 | wcselby | [TK]D-Fender I'm sorry I thought he was talking about dynamic agents |
| 16:29.37 | wcselby | dynamic agents for queues |
| 16:29.45 | wcselby | if that's not what he meant, then my example totally doesn't work |
| 16:29.49 | *** join/#asterisk bmint (~bmint@h146.129.90.75.dynamic.ip.windstream.net) |
| 16:29.52 | [TK]D-Fender | furlonda, And there is a chapter in the boko for this. |
| 16:29.53 | [TK]D-Fender | ~book |
| 16:29.54 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 16:30.12 | [TK]D-Fender | Coincidentally. |
| 16:30.22 | tompaw | it doesn't survive... |
| 16:30.30 | bmint | For some reason Asterisk all of a sudden stopped announcing the parking lot extension when parking a call. How can I debug this? |
| 16:30.45 | p3nguin | I'd imagine it was not used correctly if it didn't make it. |
| 16:30.57 | tompaw | Set("SIP/2001-00000003", "MACRO_EXTEN=59123") in new stack ... -- Executing [s@macro-nway-invite1:1] :-( |
| 16:30.57 | p3nguin | ARG1 will carry the MACRO_EXTEN into the macro. |
| 16:31.36 | p3nguin | From there, you could do anything you want with the data in ARG1, including set it to another variable or even a database key. |
| 16:32.04 | tompaw | [TK]D-Fender: if I leave the conference and this action is triggered by an applicationmap, how do I join back? :-) |
| 16:32.17 | tompaw | I mean, I need to store the conference ID somewhere. |
| 16:35.50 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:36.57 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 16:40.24 | tompaw | I need some light web-based sip phone that can be controlled via JS. So far I found only flashphoner.com - does anyone have any experience in this field? |
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| 17:02.58 | *** join/#asterisk skirmisha (~vk@46.10.5.42) |
| 17:03.06 | skirmisha | hello |
| 17:03.38 | skirmisha | can someone help me with SS7 |
| 17:03.57 | WIMPy | ~ask |
| 17:03.57 | 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. |
| 17:04.07 | skirmisha | how do i assign point codes to servers, means do i need to generate them or i need to request them from the telco |
| 17:04.25 | skirmisha | i am talking for point codes for internal use |
| 17:05.33 | WIMPy | Internal? Sounds interesting. |
| 17:05.45 | skirmisha | i've been reading how to get point code, but can not find anything about point code assignments |
| 17:06.03 | skirmisha | lets say i want to connect several servers within my network |
| 17:06.08 | skirmisha | they don't have public access |
| 17:06.20 | skirmisha | like database etc... |
| 17:06.24 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:06.40 | skirmisha | i have ITU point code for my gateway which is fine |
| 17:07.02 | skirmisha | but i am not sure whether i need to use ANSI variant for internal or ITU and generate random point code |
| 17:07.16 | skirmisha | http://www.ss7-training.net/sigtran-training/ch07lev1sec1.html |
| 17:10.58 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 17:11.29 | WIMPy | T thing the correct way would be to set up an independant local network. |
| 17:11.41 | WIMPy | I think ... |
| 17:14.45 | r0m|u | o why o why does comcast suck so bad |
| 17:15.06 | p3nguin | Because no one has the power to stop them. |
| 17:15.19 | r0m|u | so true :( |
| 17:16.14 | WIMPy | skirmisha: http://rfc-ref.org/RFC-TEXTS/4666/chapter1.html from section 1.4 on might be helpful |
| 17:16.36 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
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| 17:17.39 | wcselby | r0m|u what did they do today? |
| 17:19.18 | r0m|u | They cant explain to me whats going on with my telephone service. I thinks is a conspiracy. Again my phone was interrupted and sent to disconnections today :( |
| 17:19.29 | wcselby | r0m|u lol |
| 17:19.57 | r0m|u | :P |
| 17:20.22 | wcselby | i mean |
| 17:20.24 | wcselby | sorry to hear that |
| 17:20.26 | wcselby | :P |
| 17:20.37 | wcselby | can you get twtelecom in that area? or at&t? |
| 17:20.48 | wcselby | i'm guessing you're on comcast business class? |
| 17:21.04 | *** join/#asterisk nix8n82-phone (~AndChat@75-174-154-7.chyn.qwest.net) |
| 17:22.00 | p3nguin | So this is the THIRD time, now? |
| 17:22.20 | r0m|u | wcselby, Yes I can get ATT which I might do. I have two comcast lines. one is residential and one is business. Business is my office and no issues there. personal is for home and ALL KINDS OF ISSUES THERE! lol |
| 17:22.27 | skirmisha | nothing about point code generation in that rfc |
| 17:22.48 | wcselby | r0m|u i would start seeing what it would take to transition your personal stuff away ASAP |
| 17:22.53 | r0m|u | I might just drop them all together if they cant get it resolved. |
| 17:23.30 | p3nguin | Did you explain to wcselby what it is they are doing? |
| 17:23.45 | r0m|u | wcselby, I am looking in to it. I gave them there ultimatum today. so Far corporate has been "ok" they gave me one month credit. |
| 17:24.20 | r0m|u | p3nguin, yes I think I did. did I wcselby? Do you know they try to port my voip.ms number in to there system? |
| 17:24.35 | r0m|u | without my authorization. |
| 17:24.39 | p3nguin | It's like some sort of number theft. |
| 17:24.52 | p3nguin | MCI got in a heap of shit for doing that. |
| 17:25.11 | wcselby | r0m|u yeah I was here during that initial conversation, I was the one that brought up going to the FCC |
| 17:25.15 | r0m|u | p3nguin, remember those days... |
| 17:25.26 | r0m|u | wcselby, ah yes! |
| 17:25.42 | r0m|u | well they gave me my did back. but the issue just tumble down from there |
| 17:26.02 | p3nguin | I just don't see how Comcast could port your DID out of VoIP.ms without a porting request. You bought the DID through VoIP.ms! |
| 17:26.07 | r0m|u | almost every day "3" times this week my number has some how been put in for disconnections |
| 17:27.04 | p3nguin | What has VoIP.ms support said about it? |
| 17:27.34 | WIMPy | skirmisha: Nope, but about scopes. Seemed helpful to me. |
| 17:27.35 | r0m|u | p3nguin, Let me explain. They did not fully port the number because the request was put in just that day. I caught it because my line went to shit as soon as the request was put in. |
| 17:27.53 | r0m|u | SO |
| 17:28.14 | r0m|u | I contacted both CC and voip.ms and put everything on a hold ASAP! |
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| 17:28.34 | *** join/#asterisk irroot (~irroot@197.109.111.39) |
| 17:28.36 | wcselby | p3nguin if voip.ms isn't the LEC, then they honestly have no say over the matter, it's not them handling the port request in the end |
| 17:29.16 | p3nguin | r0m|u: Did you determine who the carrier is that VoIP.ms got the number from? Was it from Comcast? |
| 17:30.02 | r0m|u | p3nguin, I put in the request. I am waiting for the reply. |
| 17:30.28 | p3nguin | After the second time they messed with my stuff, I would have called the FCC. |
| 17:32.17 | r0m|u | wcselby, Thats exactly what voip.ms said but they said that if I did not put the request there is something called "snapback" that allows them to get that number back without questions right away. |
| 17:32.40 | wcselby | nice |
| 17:32.46 | r0m|u | All they need to know is that I did not put in the request and the will do that method.... |
| 17:32.55 | r0m|u | which implies that this has happen in the past |
| 17:33.02 | r0m|u | at least I think. |
| 17:34.05 | p3nguin | I'm sure it happens frequently. |
| 17:34.18 | r0m|u | p3nguin, I agree. |
| 17:34.25 | wcselby | it does |
| 17:34.58 | r0m|u | p3nguin, I am tempted to get the FFC involved but I just dont want to make the shit stinks more than what it does. |
| 17:35.25 | p3nguin | You'll have to draw the line somewhere. You said yourself that this is the third time. |
| 17:35.47 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 17:35.52 | r0m|u | I agree. |
| 17:36.15 | wcselby | r0m|u at this point, either finalize moving to a new provider, or go to the FCC if you want to stick it out with Comcast |
| 17:36.22 | wcselby | but I personally would just go to the new provider |
| 17:37.05 | r0m|u | Agreed. enough complaining I guess. :) |
| 17:44.27 | vader-- | thoughts Polycom IP 335 vs IP 450? for a cheap entry level starter phone |
| 17:46.39 | wcselby | 335 |
| 17:46.42 | wcselby | IMO |
| 17:46.47 | vader-- | have you used them? |
| 17:46.52 | wcselby | unless they want a larger screen , then the 450 is good |
| 17:46.53 | wcselby | yeah |
| 17:46.57 | wcselby | i like them |
| 17:47.00 | wcselby | small screen though |
| 17:47.02 | wcselby | two lines |
| 17:47.07 | wcselby | two lines of screen I mean |
| 17:47.16 | vader-- | i need a good entry phone for a small business |
| 17:47.20 | wcselby | 335 |
| 17:47.23 | vader-- | im going to setup a small freepbx system for them |
| 17:47.24 | wcselby | it's like an 80 dollar phone |
| 17:47.31 | vader-- | they want to spend about 100$ a phone |
| 17:47.42 | wcselby | how many phones? |
| 17:47.56 | vader-- | they want to be able to put a call on hold, walk to the other side of the store and pick it up |
| 17:47.59 | vader-- | 5-7 |
| 17:48.16 | wcselby | that's parking lot |
| 17:48.26 | wcselby | not a function of the phone itself |
| 17:48.35 | wcselby | i think you can get 335's for around 100 |
| 17:48.47 | wcselby | they may not be 80 dollar phones, I was thinking of the next step down |
| 17:48.52 | wcselby | just depends on your vendor |
| 17:49.29 | r0m|u | wants a cisco phone with back lit screen :) |
| 17:49.39 | r0m|u | or even a polycom |
| 17:49.42 | r0m|u | :) |
| 17:49.47 | wcselby | r0m|u have you looked at the new cisco 50x series? |
| 17:49.50 | wcselby | i think they may have backlit |
| 17:50.03 | wcselby | and if you want a polycom backlit, I know for sure the 550 has it (that's what I have at home) |
| 17:50.12 | wcselby | i think the 335 and 450 may as well |
| 17:50.14 | r0m|u | No I have not. maybe p3nguin can chime in some info. he is a cisco guy |
| 17:50.29 | r0m|u | 550 is like the 501? |
| 17:50.41 | wcselby | don't mention too much cisco in here , Qwell will get mad |
| 17:50.48 | wcselby | 550 is an upgraded 501, yeah |
| 17:50.49 | vader-- | 335 and 450 now have backlit |
| 17:51.06 | wcselby | 550 is around 225-ish? I think, depending on vendor |
| 17:51.14 | wcselby | maybe a bit less, I haven't priced phones in a while |
| 17:51.27 | wcselby | i love my 550, btw |
| 17:51.32 | wcselby | i use it at home |
| 17:51.44 | Qwell | wcselby: I won't get mad - I'll just laugh at people that buy Cisco. :) |
| 17:51.55 | wcselby | Qwell :) |
| 17:52.09 | Qwell | The non-79xx series are a little better though |
| 17:53.25 | citywok | i will forever :heart: my super easy to configure & auto-provision aastra phones |
| 17:53.27 | r0m|u | Qwell, do I dare ask.... whats wrong with cisco? Just curious :) |
| 17:53.38 | Qwell | ~cisco licensing |
| 17:53.40 | Qwell | ~ciscolicensing |
| 17:53.43 | Qwell | stupid bot |
| 17:53.46 | r0m|u | rofl |
| 17:53.49 | r0m|u | hahahaha |
| 17:53.53 | Qwell | http://ntbox.com/cisco-openletter.html |
| 17:53.56 | r0m|u | I see. |
| 17:54.05 | Qwell | feel free to skip down to the response |
| 17:54.12 | Qwell | it took them weeks to draft and run by legal |
| 17:55.54 | irroot | Qwell Cisco like Apple is a quasi religion |
| 17:56.10 | r0m|u | damn. wtf cisco.... :/ |
| 17:56.25 | r0m|u | so it is tru what I heard..... |
| 17:56.34 | r0m|u | You buy a cisco phone on ebay you are sol |
| 17:56.51 | wcselby | r0m|u if they decide to pursue the matter, yes |
| 17:56.58 | wcselby | but I have never had them try to do that |
| 17:57.11 | r0m|u | o wow. thats fucked up... |
| 17:57.14 | r0m|u | I see. |
| 17:57.17 | carrar | buy polycom |
| 17:57.33 | irroot | wcselby just cause they have not does not mean they wont |
| 17:57.41 | r0m|u | but polycom needs licenses to surten futures too right? |
| 17:58.08 | Naikrovek | with 4.0, very few features require the license |
| 17:58.13 | r0m|u | my phone is all ways looking for a license file.... |
| 17:58.17 | Naikrovek | i think only one, where previously there were 6 i think |
| 17:58.18 | irroot | carrar Polycom is going the same way hear they slacked off on some of the licenced modules but i still dont recomend them |
| 17:58.52 | carrar | what modules are we talking about? |
| 17:59.06 | r0m|u | 550? |
| 17:59.16 | r0m|u | I have a 501. |
| 17:59.17 | carrar | thats a model |
| 17:59.22 | carrar | not a module |
| 17:59.23 | r0m|u | I want to upgrade to "something" with a back lit |
| 17:59.27 | r0m|u | o sorry |
| 17:59.29 | r0m|u | lol |
| 17:59.39 | r0m|u | Miss read the sentence |
| 17:59.44 | irroot | Naikrovek talking about polycom v3... have a customer that sometimes they get MWI and clear msg but need to reboot phone as the "handset" mic/speaker does not work but speaker phone does .... |
| 18:00.17 | carrar | I don't use v.4 polycom |
| 18:00.37 | r0m|u | irroot, I use 3.1.7 never had that issue |
| 18:01.01 | r0m|u | well I never use the handset ether. I all ways use speaker. |
| 18:01.20 | r0m|u | so I guess I shouldn't say anything. :/ |
| 18:01.43 | irroot | yeah its a odd one i will be going there sometime to reproduce it and track it down |
| 18:02.54 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 18:03.09 | wcselby | afkish |
| 18:04.59 | Naikrovek | v4 just came out... most people wont' have it for a while |
| 18:06.37 | r0m|u | I need to trim my whole config files down on my polycom. I am still using the default file from polycom. Is taking the lines out of the config files the same as default options? |
| 18:07.05 | [TK]D-Fender | r0m|u, Not quite |
| 18:07.40 | r0m|u | [TK]D-Fender, so I have to specify the options in most cases even though I dont need them? |
| 18:08.00 | [TK]D-Fender | r0m|u, The phone remembers every option you set. When it loads configs effective what is in the configs overrides the "running storage". Which means leftovers can sit in the phone if not overridden. This poses issue on version jumping, etc. Also meakes things harder to debug |
| 18:08.12 | *** join/#asterisk francisvgarcia (~francis.g@190.6.137.113) |
| 18:09.00 | *** join/#asterisk pdtpatrick1 (~pdtpdt@12.249.4.226) |
| 18:09.24 | r0m|u | [TK]D-Fender, ah! very clear. Thanks [TK]D-Fender! |
| 18:09.39 | r0m|u | ~motorbooat |
| 18:09.44 | r0m|u | ~motorboat |
| 18:09.45 | infobot | <r0m|u> its all clear now. like boobs slaping you in the face! |
| 18:14.15 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 18:17.39 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 18:31.53 | r0m|u | I want to become an ITSP. |
| 18:32.05 | r0m|u | :/ |
| 18:32.42 | carrar | sounds great |
| 18:32.49 | carrar | make it happen |
| 18:33.19 | r0m|u | :) |
| 18:33.49 | r0m|u | I have a lot to read. lol |
| 18:36.01 | r0m|u | I once was idling at the freepbx chann and I mentioned that I wanted to do calling cards and become an ITSP as well. some dude told me I have no clue what the hell Ia m getting too. ITSP/Calling Card Business is full of cheating and deceptions and that I will be quickly burn out of the market. I quickly got discourage. |
| 18:36.52 | r0m|u | But the thought is still there. |
| 18:37.16 | p3nguin | There are plenty of ITSPs already, so becoming one isn't necessarily a wonderful idea. |
| 18:37.27 | p3nguin | It's like becoming a new web host. |
| 18:39.05 | r0m|u | p3nguin, If done properly been an ITSP does not incur much cost. dont you think? |
| 18:39.26 | p3nguin | Much cost is a relative term. |
| 18:39.47 | r0m|u | In terms of running it self beside support and uptime. |
| 18:40.01 | irroot | r0m|u it is a dodgy side of bussiness margins are low so you need to screw on quality ... been a niche provider as a partner to a ITSP is a great idea we work well with our ITSP and its more benificial |
| 18:40.27 | r0m|u | irroot, like reselling an itsp? |
| 18:41.27 | irroot | yes and back to back arrangements .... with the margin so low the risk is always someone does not pay and that is a real risk |
| 18:41.45 | r0m|u | irroot, I see. |
| 18:41.55 | r0m|u | I do that with voip.ms and again I was quickly discourage with "users should skip you and should go straight to the source" so I drop it. |
| 18:43.05 | irroot | we do asterisk solutions as our core bussiness and we hook these up to a a hosted platform at our ITSP we manage the accounts and value adds they route the traffic we also have our own DSL realm and range of private ip's linked to it for voice purposes coming in on a STM-1 |
| 18:43.14 | p3nguin | There's a correct way and an incorrect way to resell. |
| 18:43.17 | wcselby | r0m|u the goal is to be able to negotiate a wholesale contract with an upstream provider that will enable you to offer a lower rate than the current offerings out there |
| 18:43.21 | wcselby | or at least a competitive rate |
| 18:43.26 | *** join/#asterisk xpot-mobile (~james@dhcp69.emcb.utah.edu) |
| 18:43.27 | p3nguin | Actually, there's a bunch of incorrect ways. |
| 18:43.29 | wcselby | and for calling cards - that's a bitch of a market |
| 18:44.25 | r0m|u | wcselby, I see, Thanks for the info. That is very useful! |
| 18:44.38 | r0m|u | p3nguin, I am sure I was one of those doing it the incorrect way :P |
| 18:44.57 | wcselby | r0m|u and of course, as you're just getting started, you'll not have a lot of minutes / bandwidth, so you're not going to get very good wholesale rates |
| 18:45.11 | wcselby | but as your minutes / bandwidth go up, your wholesale rates can get negotiated lower, etc |
| 18:45.29 | r0m|u | Is flow rate a good upstream provider? |
| 18:45.36 | irroot | problem with calling cards they always refer to "millions" of minutes you need to put up the platform and infrastructure then they go in 3 months or a yeat |
| 18:46.06 | r0m|u | irroot, Yea I was told that was more like a mafia market LOL |
| 18:46.13 | wcselby | irroot I've developed a ccs system that doesn't require that, but we've also targeted a niche market with willing customers |
| 18:46.37 | irroot | wcselby +1 |
| 18:48.17 | wcselby | alright, time for me to go get some lunch I think |
| 18:48.18 | r0m|u | so here is my scenario. I have a set market and is outside the US. I want to get in the mix and provide voip service to them. I start by acquiring and negotiating a contract with upstream provider? |
| 18:48.22 | wcselby | and of course, it's just started to rain |
| 18:48.34 | r0m|u | wcselby, we have TS warning |
| 18:48.36 | wcselby | r0m|u you'll want to separate things |
| 18:48.52 | r0m|u | wcselby, msg me. |
| 18:48.52 | wcselby | get a good low-cost all in one origination provider |
| 18:48.59 | wcselby | okay one sec |
| 18:49.06 | r0m|u | I dont want to flod the chan with useless info. |
| 18:49.11 | wcselby | lol k |
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| 18:53.16 | r0m|u | is thinking of lunch. |
| 18:53.27 | r0m|u | p3nguin, you avail? |
| 18:56.05 | Qwell | Please don't eat p3nguin. |
| 18:57.31 | r0m|u | lmao |
| 18:57.40 | r0m|u | that didnt sound right :/ |
| 18:57.47 | r0m|u | lol |
| 18:58.04 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 18:58.25 | r0m|u | Qwell, interesting reading on the open letters... |
| 18:59.10 | r0m|u | So I was looking in to some reviews and found that a customer had some issues with voip.ms E911 routing while hes house was on fire.... |
| 18:59.29 | r0m|u | or "catching" fire |
| 19:00.15 | r0m|u | This goes back on my issue of having a "proper" way to test this E911 centers as customers. |
| 19:06.50 | p3nguin | Here. |
| 19:08.03 | *** join/#asterisk alrs (~lars@97.65.116.210) |
| 19:08.17 | r0m|u | p3nguin, msg me please. |
| 19:09.27 | p3nguin | .win 35 |
| 19:09.31 | p3nguin | sigh |
| 19:14.05 | r0m|u | Winchester? |
| 19:14.17 | r0m|u | cal .35 |
| 19:15.16 | r0m|u | Its been around for many years if not century's :) |
| 19:15.37 | p3nguin | Maybe centuries |
| 19:15.53 | p3nguin | since apostrophes are not used to make something plural. |
| 19:16.48 | r0m|u | ops. Thanks for the correction :) |
| 19:16.54 | p3nguin | But no, I intended to type /win 35 to see what you said to me, but I messed up. |
| 19:16.55 | [TK]D-Fender | I don't remember any significant cartridge that is 35 cal... |
| 19:17.55 | [TK]D-Fender | 9mm ~ .36". 36 cal was common ball size in cap&ball pistols in the US civil war era, etc.... |
| 19:18.00 | [TK]D-Fender | never a 35 though... |
| 19:19.14 | *** part/#asterisk SwK (~SwK@freeswitch/developer/swk) |
| 19:19.16 | r0m|u | [TK]D-Fender, http://www.leverguns.com/articles/model_1895.htm |
| 19:19.30 | r0m|u | also http://en.wikipedia.org/wiki/.35_Winchester |
| 19:19.51 | r0m|u | Maybe not significant but its been around for some time. |
| 19:20.15 | [TK]D-Fender | Other cartridges were introduced, as follows: the .35 WCF (1903), which is a grossly underrated |
| 19:20.21 | r0m|u | It did not take of in major popularity like the 10MM hasent. |
| 19:20.27 | [TK]D-Fender | r0m|u, A lot of shit was made long ago and quickly forgotten :p |
| 19:20.35 | r0m|u | lol so true |
| 19:21.14 | [TK]D-Fender | "So, I had a .270 rebored & rechambered to .35 Whelen. The Whelen is the .30-06 necked up to .35 caliber" <- THIS is kinda scary in the big-game dept. |
| 19:21.28 | *** join/#asterisk JustinCampbell (~justinCam@74-94-59-225-Philadelphia.hfc.comcastbusiness.net) |
| 19:21.34 | [TK]D-Fender | muzzle energy must be frightening on it |
| 19:21.51 | r0m|u | lol |
| 19:22.15 | JustinCampbell | I have an Avaya phone connected via SIP, and it hangs up every call after 1-2 seconds |
| 19:22.23 | JustinCampbell | is there some magic i can put in sip.conf to fix that? ;) |
| 19:22.48 | [TK]D-Fender | JustinCampbell, Got debug for us to look at? |
| 19:23.22 | JustinCampbell | [TK]D-Fender: i can grab sip packets |
| 19:23.28 | JustinCampbell | is there a better debug to get? |
| 19:25.04 | r0m|u | p3nguin, you come across a .35? |
| 19:25.17 | p3nguin | No, you made that up all by yourself. |
| 19:25.24 | r0m|u | came* |
| 19:26.00 | r0m|u | hu? ok. lol |
| 19:26.07 | [TK]D-Fender | X.35 would be far more common to hear around here :) |
| 19:26.16 | [TK]D-Fender | JustinCampbell, Not yet |
| 19:26.28 | p3nguin | (1316.53) <p3nguin> But no, I intended to type /win 35 to see what you said to me, but I messed up. |
| 19:26.37 | p3nguin | It has nothing to do with .35 anything. |
| 19:26.47 | r0m|u | p3nguin, I msg you back. Did you get it? |
| 19:26.51 | r0m|u | ooo lol |
| 19:26.54 | p3nguin | I saw it. |
| 19:27.01 | r0m|u | sorry. |
| 19:27.10 | p3nguin | I couldn't find your credit card on file, though. |
| 19:27.32 | r0m|u | I thought I put it in. |
| 19:27.58 | p3nguin | *shrug* |
| 19:28.01 | r0m|u | I change CC's that's probably why. |
| 19:28.11 | r0m|u | :P |
| 19:29.23 | r0m|u | curiosity is all. some question here go even further. which some times puzzle me.... |
| 19:30.13 | p3nguin | I understand your curiosity, and that's what consultants are for... |
| 19:30.29 | p3nguin | But when we don't get paid for our work, we get bitchy. |
| 19:30.52 | *** join/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust148.aztw.cable.virginmedia.com) |
| 19:31.23 | francisvgarcia | JustinCampbell tell us more about your problem and post your sip.conf |
| 19:31.25 | p3nguin | I give away far too much free help in this channel. |
| 19:31.41 | JustinCampbell | francisvgarcia: im working on getting a sip debug |
| 19:31.46 | JustinCampbell | will gist sip.conf first |
| 19:32.00 | *** part/#asterisk ChUbB (~IceChat7@cpc2-aztw12-0-0-cust148.aztw.cable.virginmedia.com) |
| 19:32.08 | r0m|u | Understood. and is why I am puzzled on how some people here just come in and get a whole infrastructure build for free... when I do nderstand most people here are consultants.... |
| 19:32.16 | r0m|u | p3nguin, ^^ |
| 19:33.42 | r0m|u | I mention this not to long ago about one user wanting to do a whole asterisk build for a "company" |
| 19:33.42 | p3nguin | I can tell you how they get it... we're fare too nice and have too much desire to help people. |
| 19:33.42 | p3nguin | s/fare/far/ |
| 19:33.59 | r0m|u | Understandable. |
| 19:35.01 | JustinCampbell | francisvgarcia: https://gist.github.com/3cb178329c5b6a00a5aa |
| 19:35.38 | r0m|u | p3nguin, we are all here for a reason. to help and to be helped... Understand that I am not here to take advantage of that. |
| 19:36.07 | p3nguin | I hope you don't think I'm accusing you of trying to take advantage. |
| 19:36.13 | r0m|u | Compared to others at least. get help and jet. |
| 19:36.23 | r0m|u | p3nguin, It felt like it. :/ |
| 19:36.24 | p3nguin | That's how a lot of them operate. |
| 19:36.40 | francisvgarcia | JustinCampbell which extensions is having the issue? |
| 19:36.55 | JustinCampbell | just 54009 |
| 19:37.05 | JustinCampbell | i have 54000-54009, but i ommitted them |
| 19:37.08 | JustinCampbell | theyre all identical |
| 19:37.14 | JustinCampbell | i have a softphone connected to 54008 |
| 19:37.15 | JustinCampbell | and that works fine |
| 19:37.20 | francisvgarcia | JustinCampbell are behind a nat? |
| 19:37.22 | JustinCampbell | its just the hardware phone thats having issues |
| 19:37.23 | JustinCampbell | yeah |
| 19:37.37 | r0m|u | Thanks for clarifying :) |
| 19:38.31 | p3nguin | Taking advantage of someone or something is driven by the intention of doing so. I think you simply have a desire of knowledge. |
| 19:38.55 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 19:39.02 | p3nguin | And having a desire of knowledge isn't necessarily a bad thing. |
| 19:39.05 | *** join/#asterisk xpot-mobile (~james@70-91-210-237-BusName-Utah.hfc.comcastbusiness.net) |
| 19:39.27 | JustinCampbell | francisvgarcia: incoming calls seem to work ok, although a hangup of the hardware phone doesnt tell asterisk that it hung up |
| 19:39.27 | francisvgarcia | JustinCampbell and what about the sip debuging |
| 19:39.32 | JustinCampbell | francisvgarcia: getting it now |
| 19:39.35 | r0m|u | Just to finish the sentence.... I recive help from the community and I give back what I can within my grasp and technical knowledge. this is why I stick around. Plus I like you guys. you guys are an awesome community!!!!!!!!! |
| 19:39.37 | francisvgarcia | ok |
| 19:40.06 | r0m|u | p3nguin, Exactly! :D |
| 19:41.55 | r0m|u | I like to understand things threwle which some times gets me in trouble :/ specially at work. lol |
| 19:41.57 | francisvgarcia | JustinCampbell to complete the situation the phones are working like this AVAYA---ADSLROUTER------INTERNET-----ASTERISK10.0.0.0/8 |
| 19:42.34 | JustinCampbell | francisvgarcia: yes, asterisk does have a public ip though that i have hidden |
| 19:42.41 | JustinCampbell | francisvgarcia: and other sip clients are working normally with it |
| 19:42.55 | r0m|u | My boss hates it when I ask questions... |
| 19:43.04 | francisvgarcia | JustinCampbell only it happens on outgoing calls |
| 19:43.24 | [TK]D-Fender | r0m|u, because your job is to answer his questions, not the other way around. |
| 19:43.39 | JustinCampbell | francisvgarcia: yes, and the other odd thing is that incoming calls, the avaya hangup doesnt tell any others that it hung up" |
| 19:43.57 | r0m|u | lmao. when he ask questions that need another question to be answered? |
| 19:43.58 | r0m|u | LOL |
| 19:44.33 | r0m|u | Thats what trips me out. My boss does not understand what I do and my position so he is at a lost most of the time |
| 19:44.41 | francisvgarcia | JustinCampbell ok, post your sip debug and let's see |
| 19:44.57 | r0m|u | He is a Past windows admin. :) |
| 19:45.11 | r0m|u | and hates Unix/Linux |
| 19:51.22 | r0m|u | I guess thats why I am here. :) |
| 19:52.39 | Naikrovek | it's a poor boss that doesn't at least have a precursory understanding of what his employees do |
| 19:52.46 | Naikrovek | his/her |
| 19:52.56 | r0m|u | Agreed. |
| 19:54.26 | JustinCampbell | francisvgarcia: https://mvpbx.s3.amazonaws.com/tcpdump1.pcap?AWSAccessKeyId=AKIAILIJMYMTDQWRAIJA&Expires=1321386870&Signature=RxriK1I1n4xvZgE1PRr5BaQ5HYw%3D |
| 19:55.25 | r0m|u | if my moh is from free streaming channel do I have to pay royalties? (US) |
| 19:55.51 | JustinCampbell | francisvgarcia: there were 2 calls, one to 54008, which isnt connected, and one to my cell phone which is out of voicepulse |
| 19:56.15 | JustinCampbell | my cell phone rang and continued to ring after the handset screen flashed that it was hungup |
| 19:56.59 | *** part/#asterisk alrs (~lars@97.65.116.210) |
| 19:57.59 | p3nguin | r0m|u: Legally, probably. |
| 19:58.26 | p3nguin | You can listen to streams legally all day long, but the minute to retransmit it to someone else, rules are different. |
| 19:58.45 | p3nguin | s/te to/te you/ |
| 19:59.25 | r0m|u | ah! |
| 19:59.27 | r0m|u | Interesting |
| 20:01.08 | r0m|u | p3nguin, Thanks. Thats what I was afraid of. I ask dijib last night how he can get away from it but he mention that in Canada there is no such law. So I was curious if streaming was different. |
| 20:01.20 | *** join/#asterisk bipolar (~bipolar@204.186.46.67) |
| 20:02.05 | p3nguin | It's just a matter of doing it and expecting no one to care. |
| 20:02.28 | r0m|u | ah. I see. |
| 20:03.50 | p3nguin | There are mp3s out there which are royalty free. |
| 20:04.57 | r0m|u | p3nguin, Interesting. Didnt know that. |
| 20:05.09 | r0m|u | is googling |
| 20:06.23 | p3nguin | http://www.royaltyfreemusic.com/music-on-hold.htm |
| 20:06.58 | r0m|u | Thanks! |
| 20:07.24 | Qwell | royalty free, costs money? |
| 20:07.36 | Qwell | one of these things is not like the other. |
| 20:08.17 | p3nguin | That's just an example. It's probably not the best choice, but it was the first thing I saw. |
| 20:09.03 | r0m|u | lol. I saw that. |
| 20:09.08 | Qwell | r0m|u: opsound.org |
| 20:09.24 | p3nguin | Basically, you buy a licensed track (or CD), and you can play it to whomever you want as often as you want, and are not required to pay royalties to the record label. |
| 20:10.12 | *** join/#asterisk sbszulu (~dundubala@197.107.122.164) |
| 20:10.32 | r0m|u | p3nguin, interesting. Not bad. I actually like this. http://www.royaltyfreemusic.com/background-music-on-hold-10.html |
| 20:10.40 | r0m|u | Qwell, Thanks checking itnow! |
| 20:11.02 | p3nguin | The prices are unreasonable, though. |
| 20:12.09 | r0m|u | Qwell, There streams are off line. |
| 20:12.25 | r0m|u | p3nguin, I have no idea what would be reasonable and what is not on that field. |
| 20:12.33 | r0m|u | Thanks for letting me know |
| 20:12.47 | r0m|u | I probable would of thought it was a good deal. :/ |
| 20:12.48 | francisvgarcia | JustinCampbell sorry I went out for a while |
| 20:12.52 | p3nguin | If you can get some free music, you'll be much better off. |
| 20:12.52 | francisvgarcia | I am actually working |
| 20:13.10 | JustinCampbell | francisvgarcia: me too! ;) |
| 20:13.42 | p3nguin | For a company who can afford to buy a hundreds of dollars in licensed tracks, it might be a good deal... but I know what you're going to do with the music. |
| 20:13.48 | francisvgarcia | JustinCampbell I checked out the link and apeear to be expired |
| 20:13.55 | francisvgarcia | can you plz post it again |
| 20:13.56 | r0m|u | p3nguin, Ill look around. Its for me for personal use so I dont think is a huge deal |
| 20:14.02 | JustinCampbell | francisvgarcia: https://mvpbx.s3.amazonaws.com/tcpdump1.pcap |
| 20:14.13 | p3nguin | qwell shared a reasonable site for free tunes. |
| 20:14.41 | r0m|u | agreed |
| 20:14.44 | r0m|u | Thanks guys! |
| 20:15.34 | r0m|u | man is storming down here :/ |
| 20:15.37 | [TK]D-Fender | r0m|u, p3nguin, Thanks. Thats what I was afraid of. I ask dijib last night how he can get away from it but he mention that in Canada there is no such law. So I was curious if streaming was different. <- there IS |
| 20:16.22 | p3nguin | There is a law in Canada? |
| 20:16.29 | [TK]D-Fender | yes |
| 20:16.33 | [TK]D-Fender | Same sort of crap |
| 20:16.42 | r0m|u | o shit. didnt know. [TK]D-Fender |
| 20:16.51 | p3nguin | He was retransmitting a radio station. |
| 20:17.00 | p3nguin | They provide the stream online. |
| 20:17.09 | [TK]D-Fender | Dijib means well ... which is usually the point where things start going very very wrong ... |
| 20:17.12 | p3nguin | Not sure how that falls into the rules. |
| 20:17.24 | [TK]D-Fender | retransmit = license required |
| 20:18.02 | r0m|u | He changed it last night to just regular * moh. so he probably saw something. |
| 20:18.12 | r0m|u | shit it was not my intention to put him on the spot :( |
| 20:18.17 | eppigy | hello |
| 20:18.18 | eppigy | i am dave |
| 20:18.30 | [TK]D-Fender | that is factual |
| 20:19.05 | r0m|u | [TK]D-Fender, thats to big of a word for me. |
| 20:19.09 | r0m|u | lol |
| 20:20.12 | r0m|u | well now is clear to me |
| 20:20.22 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 20:20.51 | r0m|u | just use the plain * moh :) |
| 20:21.43 | [TK]D-Fender | IIRC didn't Digium's license for the MoH it used to come with expire or something? |
| 20:21.49 | [TK]D-Fender | in this past year... |
| 20:22.01 | r0m|u | I am a big fan of somafm and it looks like they wont be online for too long. damn royalty crap. |
| 20:22.13 | r0m|u | [TK]D-Fender, so it expires? |
| 20:23.01 | p3nguin | Something happened during the past couple of years because the included moh changed. |
| 20:25.08 | wcselby | hmmmm.....web designers and their fickle feelings |
| 20:26.44 | wcselby | working with a web design firm to redesign the company website, and they're not taking the criticism too well....lol |
| 20:27.11 | p3nguin | Typical. |
| 20:27.25 | p3nguin | You can't tell anyone they're wrong. |
| 20:27.29 | p3nguin | It's just simply not allowed. |
| 20:29.44 | wcselby | heh |
| 20:29.50 | wcselby | the best part of it all is |
| 20:29.56 | wcselby | they're giving us a wordpress template |
| 20:30.11 | wcselby | a "custom" website |
| 20:30.34 | wcselby | but they forgot to change the favicon away from the wordpress W |
| 20:31.15 | p3nguin | hahaha |
| 20:31.18 | p3nguin | Brilliant! |
| 20:32.40 | r0m|u | ROFL! |
| 20:32.57 | r0m|u | I had a "web developer" do that for me before |
| 20:33.14 | r0m|u | He got mad when I call him out on it |
| 20:34.57 | wcselby | so i'm not really involved in the call, but they ask me for my opinion every now and then |
| 20:35.14 | wcselby | but I just overheard the designer saying something about being offensive to his sensibilities or something along that line |
| 20:35.22 | wcselby | a particular color they were talking about I think |
| 20:35.41 | wcselby | shakes his head |
| 20:35.55 | wcselby | i used to work for a web design company, and I gotta say, web designers are some of the most arrogant people in IT |
| 20:37.08 | r0m|u | TRUE! I cant even start talking about our internal web designer. He think He is God. |
| 20:38.06 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 20:38.40 | p3nguin | You mean... |
| 20:38.44 | p3nguin | He isn't? |
| 20:38.54 | p3nguin | I was under the impression he was. |
| 20:39.01 | r0m|u | LOL |
| 20:39.12 | r0m|u | rofl |
| 20:44.58 | *** join/#asterisk Netgeeks (~chris@gw1.netgeeks.net) |
| 20:45.32 | r0m|u | looks like the show "the office" use voip. lol cisco phones. |
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| 20:47.21 | p3nguin | Having that show on makes my nerves very unsettled. |
| 20:47.38 | p3nguin | Every time a phone rings, my body jumps to want to answer the phone. |
| 20:48.53 | jaytee | man, bandwidth.com support sucks |
| 20:49.45 | merlin077 | Hey Folks.. I have a question ... If I have a dial rule like that: exten => SIP-EXTERN-NUMBER,1,Dial(SIP/7424) then it will route the call from the SIP-EXTERN NUMBER to the internal Phone 7424. Is it possible to route the call to other Phones (e.g. 7425,7426,..) simultaneously? |
| 20:50.28 | merlin077 | So that all phones ring, and the first one I pick up, is being connected with the incoming call |
| 20:52.58 | wcselby | r0m|u p3nguin yeah, they've used cisco phones on that show since the early seasons (maybe since the beginning). the show NCIS Los Angeles uses polycom phones and when there's an incoming call, it will show an incoming call from XXX@192.168.150.xxx or whatever (i.e the extension @ some private IP address). that always made me laugh |
| 20:52.58 | r0m|u | p3nguin, lol. why is that? |
| 20:52.58 | navaismo | merlin077: Dial(SIP/7424&SIP/7425&SIP/7426) |
| 20:52.59 | [TK]D-Fender | <jaytee> man, bandwidth.com support sucks <- I've gotten some good business picking up their failed leftovers... |
| 20:52.59 | r0m|u | wcselby, Thats funny. I never noticed that |
| 20:53.36 | wcselby | it'll actually say sip:1004@192.168.123.123 or whatever, so I mean, someone's making an actual call to the phones for them to start ringing on the show |
| 20:53.52 | r0m|u | Hey at least they got something right |
| 20:53.55 | wcselby | you only notice it if they focus in on the phone for something dramatic (like they've been waiting for a phone call etc) |
| 20:53.57 | r0m|u | LOL |
| 20:54.04 | merlin077 | navaismo: oh man, you´re a god.. thank you :) |
| 20:54.14 | r0m|u | everything else in the show specially their IT jargon is so fucked up |
| 20:54.15 | r0m|u | lol |
| 20:54.21 | wcselby | heh |
| 20:55.46 | p3nguin | I liked the one where the hard drive was ruined, so they were pulling the platters out of it to put into another drive that wasn't ruined so they could get the data off it. |
| 20:56.03 | p3nguin | Of course I could be mistaking NCIS with CSI. |
| 20:56.23 | p3nguin | Too similar to distinguish between them at times. |
| 20:56.48 | wcselby | that happened on ncis LA p3nguin |
| 20:56.49 | wcselby | i think |
| 20:57.07 | wcselby | which is actually one of the techniques that hard drive recovery specialists will try, but it's last ditch |
| 20:57.21 | r0m|u | LOL! yes! indeed. It cracks me up! |
| 20:57.23 | *** join/#asterisk celord (~celord@201.195.243.194) |
| 20:57.24 | wcselby | ncis la tries to be a bit more realistic, but ncis (the original) just says "fuggit" |
| 20:57.58 | wcselby | ncis la uses up all their crazy in the storyline, ncis original spreads it around to the it forensics too |
| 20:58.00 | wcselby | lol |
| 20:58.15 | wcselby | BUT!!! |
| 20:58.18 | r0m|u | wcselby, how about cis and the"hacker" ladie and the guy using the same keyboard at the same time |
| 20:58.23 | p3nguin | Moving platters is the absolute last thing I would want to try, because chances are that will ruin it. |
| 20:58.42 | wcselby | p3nguin there are special ways to do it, obviously they left that stuff out for the show |
| 20:58.46 | wcselby | :) |
| 20:58.53 | [TK]D-Fender | <wcselby> that happened on ncis LA p3nguin <-- would be about the only tech-related task they've probably ever done that even vaguely resembles reality... |
| 20:59.00 | p3nguin | heh |
| 21:00.04 | p3nguin | Did they ever remove the platters and examine under an electron microscope? |
| 21:00.10 | wcselby | what I was gonna say - in the first or second season of NCIS, when they were bringing their first computer guy on board, he was back tracing ss7 signals to get a location or something....it made me laugh, because it was a blatant use of completely un-understood terminology to sound legit |
| 21:00.19 | wcselby | but they gave up on that kind of crap shortly there after |
| 21:00.32 | r0m|u | Is't the needle calibrate it for the platters? makes it impossible to use one platter to another |
| 21:00.32 | wcselby | r0m|u yeah the two people on one keyboard thing was funny |
| 21:00.42 | wcselby | i do that shit all the time yo. my wife and i speed up tons of stuff doing that |
| 21:00.44 | wcselby | lol |
| 21:00.50 | r0m|u | LMAO |
| 21:01.42 | _Corey_ | I like whenever CNBC talks about hackers, they have some *NIX window up with a guy typing 'ls' on repeat in /var or something |
| 21:01.53 | wcselby | _Corey_ lol |
| 21:02.02 | wcselby | SO.......remember that web designer with the wordpress template |
| 21:02.20 | p3nguin | or using ping 127.0.0.1 |
| 21:02.20 | wcselby | just found this in the page source of the demo page they're using |
| 21:02.24 | wcselby | <!-- Check out other <a href="http://www.templatemonster.com/wordpress-themes.php" title="WordPress themes" target="_blank">awesome WordPress templates</a> on TemplateMonster. |
| 21:02.33 | [TK]D-Fender | p3nguin, Good work Dorothy! |
| 21:02.40 | r0m|u | _Corey_, LOL |
| 21:02.44 | *** join/#asterisk xpot-mobile (~james@155-99-213-133.uconnect.utah.edu) |
| 21:03.57 | r0m|u | http://www.gideontech.com/content/articles/326/1 |
| 21:04.49 | r0m|u | http://themostboringblogintheworld.wordpress.com/2006/09/13/worlds-worst-hacker-irc-transcript/ |
| 21:07.10 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 21:07.51 | wcselby | so |
| 21:07.57 | wcselby | 3 grand to customize a wordpress template |
| 21:08.01 | wcselby | interesting |
| 21:08.03 | r0m|u | damn |
| 21:08.04 | eppigy | lol |
| 21:08.07 | eppigy | bloggers |
| 21:08.13 | wcselby | no no no |
| 21:08.15 | wcselby | what I mean is |
| 21:08.32 | wcselby | my company is paying a web design firm somewhere in the realm of 3 grand for a "new, custom" website |
| 21:08.38 | p3nguin | I think I would have done it for you for $250. |
| 21:08.52 | wcselby | and what they're trying to give us is going to be a customized word press template |
| 21:09.10 | p3nguin | That's my rate for a templated site. |
| 21:10.44 | wcselby | lol |
| 21:10.48 | wcselby | 4 grand apparently |
| 21:10.51 | r0m|u | lol |
| 21:10.55 | wcselby | "but that includes a custom CMS" |
| 21:10.59 | wcselby | so we'll see |
| 21:11.08 | wcselby | that was what the COO just told me |
| 21:11.15 | wcselby | they don't understand what I was trying to say |
| 21:12.33 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 21:14.14 | p3nguin | Of course they don't. |
| 21:14.16 | [TK]D-Fender | wcselby, 3k$? That's nothing ... its amazing how much my company got ass-raped for their Joomla wonder going about 3.5 years... Lets just say "high-5-figures) |
| 21:14.24 | p3nguin | ALl they see is a new web site. |
| 21:15.46 | wcselby | heh |
| 21:25.03 | [TK]D-Fender | Alrighty.... checkout time, BBIAB |
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| 21:33.39 | p3nguin | Why would MixMonitor suddenly not be mixing the two channels in sync? |
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| 23:28.06 | Penggu | hi all. probably not the most appropriate way to ask: i am toying with AsteriskNOW and have installed the Asterisk-GUI version. I'm unhappy with it and want to switch to FreePBX. is there a quick way to do it? |
| 23:28.39 | Penggu | i realise that there's editing config files by hand, etc., but i want to taste these config managers |
| 23:28.54 | Penggu | (or *mis*managers, if you like..) |
| 23:29.56 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-lshkhlezwvuslohn) |
| 23:36.31 | [TK]D-Fender | Penggu: Yes. Take your AsteriskNOW CD. Instert it into the machine. Reboot. Reformat and stat fresh |
| 23:36.34 | [TK]D-Fender | start* |
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| 23:44.05 | WIMPy | Penggu: We generally do not support the GUIs in here. Both of them have their own channels. |
| 23:44.44 | p3nguin | Starting over is the best way. |
| 23:44.55 | p3nguin | Just choose the other option. |
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