IRC log for #asterisk on 20130217

00:00.07Neptu_laptop[TK]D-Fender, that will not be an issue i think is 8GB on the datacenter
00:00.28[TK]D-FenderFind out for sure.
00:00.37[TK]D-FenderNeptu_laptopectospasm, well i need to be able to reach 200 cellphones same time so i need a gateway someway, just guessing <-- do they really have to be "at the same time"?
00:00.49[TK]D-FenderNeptu_laptop: Precisely what do you need to be doing with them?
00:01.41Neptu_laptop[TK]D-Fender, well we have around 3000 contestants and the playing is 1 min or so we want all to give them a chance
00:02.14[TK]D-FenderAs long as you're not dreaming of conferencing them...
00:02.24Neptu_laptop[TK]D-Fender, is just a PR thingie kind of a quest... need to notify the paricipants to do some stuff with a recording that is all the problem is the numbers
00:02.29Neptu_laptopis not massive calls
00:02.41Neptu_laptopis just the number and at same time... there is the challenge
00:07.05Neptu_laptopanyway need to find a cheap one the ones i called were around 5 dolars per "line"
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00:11.08Ice_StrikeNeptu_laptop Have you seen voip.ms ? I am sure there are unlimited lines for free
00:11.19Neptu_laptopmmmmmmmmmmm
00:11.20Ice_Strikeyou just pay the top-up credit
00:12.59Neptu_laptopmmmmmmm checking but if ts unlimited lines that will be great
00:13.04Neptu_laptopthe pricing seems ok aswekk
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00:28.02Neptu_laptopIce_Strike, just sent a mail to support lets get an answer sometime soon. Thanks for the tip!!
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00:57.43Ice_StrikeNeptu_laptop your welcome!
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03:04.26allfrohello
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03:56.09ectospasm~ask
03:56.09infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
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06:11.51artyxMake menuconfig with asterisk 11.3.0-rc1 .. channel motif driver shows XXX .. what am i doing wrong?
06:14.38artyxI am reasonably certain i have added the dependencies, is there a way to force the location of these ?
06:19.04artyxtrying an uninstall of the compield version and switching to distribution default
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09:15.11blergh-Hello, anyone using Elastix or similar? Or using mydivert?
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10:52.59bpietrohi, have you some link to guide how configure Zoiper on Android tablet for local Asterisk PBX? Googled for 2 days w/o results. tnx
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10:53.36bpietroZoiper on Win and Linux configured w/o problems
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11:30.37_zoom_can i assign specific caller id to my pri line?
11:36.29[sr]only the one's your telco allows
11:37.30[sr]if your number range is 234377500-559, and you define 234377999, it'll go out with the line default
11:37.44[sr]this, for your specific line
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11:39.34WIMPyHi [sr]
11:41.14_zoom_[sr], if i have a range and i would like to do in rotate?
11:41.27[sr]WIMPy: how r u?
11:42.02WIMPyBetter. Cought a very nasty version of the flu.
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11:42.20[sr]WIMPy: i'm with a similar problem :| damn
11:43.09[sr]_zoom_: if its on your range you can do it, if not working can be a telco missconfiguration, i had a similar problem once
11:43.58WIMPy[sr]: How's your dahdi?
11:45.19[sr]WIMPy: i think i got the problem.. but i have 99% sure it's freepbx dahdiconfig that is making this happen, anyway the need to make "dahdi restart" make it work is strange
11:45.41[sr]uninstalled dahdiconfig and doesn't happen anymore (on core reload)
11:46.27WIMPyAlways these interesting features.
11:46.39[sr]:)
11:47.01WIMPyAnd just a nother reason to keep things seperate if possible.
11:47.16[sr]just dont understant the "dahdi restart" part(asterisk console)
11:47.56[sr]thats why i was thinking that was a chan_dahdi problem, but no dead locks when that is in that state
11:48.30WIMPyWell, it has to be chan_dahdi, doesn't it?
11:48.56WIMPyIt's obviousely not dahdo itself.
11:49.01WIMPydahdi
11:49.14[sr]well yes, when i say dahdi i mean chan_dahdi sorry
11:52.02WIMPyalways finds it hard to tell apart which part does what.
11:52.50[sr]WIMPy: i have an ATA for fax on local lan, works ok, but i get this, and everything's set to alaw/g.711a, what does it really mean? dont find any much info on the web, res_rtp_asterisk.c:2358 ast_rtp_read: Unknown RTP codec 100 received
11:55.58WIMPyDoesn't seem to fit. It says unknown, but it works anyway?
11:56.23[sr]ya
11:56.50[sr]is there a list to know the corresponding codec of this "100" ?
11:56.54[sr]i have no idea :)
11:57.51WIMPyYou should be able to see it if you look at sip debug.
11:58.15[sr]on the ATA for "100" i only see NSE dynamic payload=100
11:58.56WIMPyWhatever NSE is.
12:00.04[sr]i really have no idea, it's under a group called SDP Payload Types
12:00.27WIMPyAha. Cisco NSE events.
12:01.35[sr]ya, its a cisto ATA
12:01.52[sr]in fact, i wasn't able to find anything else on the market :|
12:02.39WIMPyWhy do you pass faxes through Asterisk?
12:03.06[sr]people want to use the multifunctional machines, those big one's
12:03.41[sr]wasn't my advice, but they insist
12:04.01WIMPyBut you're using BRIs, don't you? Why don't you connect them in parallel to Asterisk?
12:04.17[sr]they're analog
12:04.23WIMPyOr if you have ptp on an extra port.
12:04.54WIMPyThat should give you a bit clear connection.
12:05.22[sr]that would give me less headache, true
12:05.32eirirshangover
12:07.24[sr]WIMPy: for example i have a specific place, that has 2x BRI, and the number is attached to one of the BRI's, and i'm routing it via VPN to a distant place, this number could be detached from the BRI to an analog line, but it's an extra cost monthly
12:08.01WIMPyErr
12:08.25WIMPyI was talking about connection an analog adapter to the BRI.
12:08.40WIMPyI.E. BRI<>analog instead of SIP<>analog.
12:09.15[sr]ah, that would work for most cases, not the one's that are far, far away
12:09.47WIMPyHow far away is far away?
12:10.07[sr]well, the BRI<->analog would be better, i'm going to see if i can get that on the market
12:10.20[sr]well, about 10km's
12:10.38[sr]it's a branch
12:11.06WIMPyOk, on that one you need VOIP/FOIP.
12:11.44[sr]but you gave me a good idea to not have headaches, the BRI<->analog
12:11.53[sr]havent think on that
12:17.30[sr]well, going to lunch now, have a nice day WIMPy :)
12:18.42WIMPyFood. Good idea.
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12:24.36ayrusI'm getting this error on making a outgoing call, "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0" pastebin http://pastebin.com/YZdA3ZxH Kindly help me. I'm using freepbx distro
12:39.14WIMPy1. As you can see yourself, that message doesn;t really say anything and 2. we don;t support FreePBX here. Try #freepbx.
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12:47.21din3shcan agents as in agents.conf be dynamic? (mysql)?
12:49.24_zoom_[sr]: have you tried to rotate numbers, assign next no.
12:50.16WIMPyIt's your dialplan. You can do whatever you want.
12:52.36din3sh?
12:53.04WIMPyThat was for _zoom_
12:57.19din3sh:)
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17:41.13davlefouhi, i have that message : PostgreSQL RealTime: Failed to query database. Check debug for more info.
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18:30.58coreyf1513davlefou: you will get more information on cli if you run 'core set debug <level>' with level of atleast 3 (you may need higher i'm not sure).  then attempt to reload a module that uses realtime
18:32.31coreyf1513davlefou: if asterisk fails to start you will have to enable debug in logger.conf.  turn if off when you fixed your problem otherwise your logs will get huge
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19:26.57davlefoucoreyf1513: merci
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20:03.45ybeddyjhello
20:04.21ybeddyjim having a problem with a sip trunk from asterisk to a samsung pbx
20:05.22ybeddyjwhenever a call is placed from the samsung side it shows up on the asterisk side as unknown/anonymous
20:06.06[TK]D-Fenderybeddyj: As what do you in the actual SIP debug from that call?
20:06.17ybeddyjhowever when you do sip show channel on the call it sshows the calling id in the ur
20:06.31ybeddyjURI
20:06.43[TK]D-FenderPerhaps you could show us...
20:06.45[TK]D-Fender~pb
20:06.45infobotA "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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude.
20:06.47[TK]D-Fender^^^^
20:09.50ybeddyjhttp://pastebin.com/tj71e2UQ
20:10.37[TK]D-FenderNo... in the actual CALL debug
20:13.29ybeddyjlike a tcpdump?
20:14.15[TK]D-FenderAsterisk SIP DEBUG from CLI
20:14.26[TK]D-Fender"sip set debug on" <-----
20:16.20ybeddyjok hold on let me do that
20:19.12ybeddyjoh boy, the person has left the remote location
20:20.26ybeddyj[TK]D-Fender, will you be online about the next 18hrs?
20:21.33[TK]D-Fenderin 18 hours?
20:22.19ybeddyjyes the person had  to pickup his kids so he left work so i wont be able to complete testing until tomorrow
20:22.49ybeddyjthats in about 18 hrs for me
20:22.52[TK]D-FenderProbably.  If not, others will be.
20:23.09[TK]D-FenderProvide proper debug and someone should be able to help point out what;'s what in there for you
20:23.14ybeddyjok thx ..i hoped that the ifno earlier could help
20:24.11ybeddyjmy last hope was to creat a rscript that could parse the uri and place the extension in the calernum) variable
20:24.28[TK]D-Fenderybeddyj: Where do you see the # in there?
20:24.45ybeddyjmy keyboard misses keys often sorry
20:25.31ybeddyjthis line has the number Original uri:           sip:2953@192.168.0.2:5060
20:25.50[TK]D-FenderThat loks to be the number they dialed in (exten), not the caller ID
20:25.52ybeddyjthis line has the number -------- "Original uri:           sip:2953@192.168.0.2:5060 "
20:26.05[TK]D-FenderYou should see that in your logs & CDDR
20:27.08ybeddyjthat happen to be the extension the person was calling from .... there was also this line  ----------------"Route:                  sip:2953@192.168.0.2:5060"
20:27.37[TK]D-FenderWe'd have to see the full call.
20:27.47[TK]D-FenderBring it up when your guy comes back
20:28.06[TK]D-FenderCan you should us logs from it that you have now?
20:28.10[TK]D-Fenderbase CLI at least?
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20:32.35ybeddyjcdr logs?
20:32.59[TK]D-Fender*'s log.
20:33.05[TK]D-Fendernot CDR for this part
20:34.48ybeddyjok letme get the clearance brb
20:35.10ybeddyjany specific prt of the logs?
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20:36.19*** mode/#asterisk [+o sruffell] by ChanServ
20:36.33[TK]D-Fenderhow about the part with your call in it?
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20:50.38resist0rjaja
20:50.43resist0ror haha
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21:20.55jnemethhello...  anybody home here?  asterisk-dev is a ghost town...
21:21.55jnemethI was wondering if there was a way to access the SVN repository on the web?  Something like http://cvsweb.netbsd.org/
21:22.21[TK]D-FenderSVN = web
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21:22.45[TK]D-Fenderhttp://svnview.digium.com/svn/asterisk/
21:23.13jnemethnot exactly...  pointing a browser at http://svn.asterisk.org/ doesn't work too well...
21:23.25[TK]D-FenderSee above.
21:24.05jnemethah, that works better...  thanks...  poking around www.asterisk.org didn't produce that answer...
21:28.42[TK]D-Fenderfirst page of Google search fo "asterisk SDVN" yields INSTANT results.
21:28.49[TK]D-FenderSVN*
21:28.51[TK]D-Fendergah...
21:30.48jnemethone would hope that one could find answers from the "official source" without having to consult Google...
21:32.44[TK]D-Fenderhttp://www.asterisk.org/downloads/source-code
21:33.14[TK]D-FenderIt was right therehttp://svn.asterisk.org/svn/asterisk
21:33.22[TK]D-FenderAlternate right off their page
21:34.06jnemethI didn't want to do a checkout, I wanted to look at it on the web...
21:34.49jnemethI just wanted to know the history of one particular change...  I didn't need a full checkout for that...
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21:36.24wdoekesjnemeth: waiting less than 3 minutes on #asterisk-dev isn't really what I'd call patient
21:38.49jnemethyou're probably right :->
21:39.21jnemethis #asterisk-dev usually sparsely populated/attended?
21:41.04[TK]D-Fenderhttp://svn.asterisk.org/svn/asterisk <- on their web page
21:41.07[TK]D-Fenderweb-viewable
21:44.29jnemethweb-viewable yes, but doesn't show me history or commit messages...  This may be useful if all you want to do is see the latest revision of a file, but didn't help me with what I needed...
21:45.04[TK]D-FenderDetails you didn't specify earlier.
21:45.31jnemeth> I didn't want to do a checkout, I wanted to look at it on the web...
21:45.32jnemeth> I just wanted to know the history of one particular change...  I didn't need
21:45.32jnemeth+a full checkout for that...
21:46.06[TK]D-FenderYes, after I gave you the first full browseable link with history, and then a second
21:46.29[TK]D-FenderI already provided both becfore you gave anything restricting the range
21:47.02[TK]D-Fender[16:45]jnemeth> I didn't want to do a checkout, I wanted to look at it on the web... <- both satisfy this.
21:47.15[TK]D-Fender[16:45]jnemeth> I just wanted to know the history of one particular change... I didn't need <- THIS was news however.
21:47.31wdoekesjnemeth: activity on -dev fluctuates. most activity during US work days
21:47.44[TK]D-FenderAnyway, you've had your answer from the 2nd line I gave you
21:48.10jnemethwdoekes: in what time zone?
21:48.18wdoekesUS time zone
21:49.10jnemeththe US has four:  pacific, mountain, central, and eastern...  I'm in Canada, but my time zone is equivalent to US pacific...
21:49.42[TK]D-Fenderjnemeth: Digium folk are all over North Americ.  This isn't a raw set of business hours.
21:49.48[TK]D-FenderDon't be too anal about the answer.
21:49.57[TK]D-Fenderthere is no magical 5PM in X time-zone cut-off
21:50.54wdoekesif it fluctuates, it doesn't really help to know which one it is, does it?
21:52.02jnemethI was only trying to narrow it down...  5 PM for me, is 8 PM in New York City (i.e. well past the end of the business day for most people)...
21:52.38wdoekessure.. but if you're aiming for work hours, 5+PM seems a bit late
21:52.45[TK]D-FenderDev's are there fromall over the WORLD
21:52.46jnemethbasically what you're telling me, is that if the sun is up, there is a fairly good chance that somebody is there...
21:52.51[TK]D-FenderAnd this is the WEEKEND
21:53.08[TK]D-FenderStop restricting yourself like that
21:53.27[TK]D-Fendermany are across North Americ.  Many are not.
21:53.53[TK]D-FenderIf you already know that 5PM is pushing it for EST then ask EARLIER
21:54.07[TK]D-Fenderthat channel does not have "business hours"
21:54.32[TK]D-FenderDitch the idea there is some specific clock you can pin them to and use some common sense
21:57.31jnemeth<wdoekes> jnemeth: activity on -dev fluctuates. most activity during US work
21:57.32jnemeth+days
21:58.22jnemeththe concept of "US work days" fluctuates with location in the US...
21:58.53jnemethcommon sense tells me that I should figure out the location...
21:59.23wdoekes.. or aim for the middle
22:01.01wdoekes*gone*. good luck with your issue
22:02.25jnemethDigium appears to be located in Alabama, which is mostly in central time zone...  so, yes, aiming for the middle would work pretty good...
22:03.00jnemethI got the immediate answer to my issue some time ago, I now have something I can investigate...
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23:21.13tflgen2hi guys, i've just set up a incrediblepbx on my raspberry pi but i'm getting a message spammed in the asterisk shell when I look. chan_sip.c:3705 __sip_xmit: sip_xmit of 0x4384dc90 (len 594) to 192.168.20.85:51732 returned -2: No such file or directory
23:32.28ghost75do you have cisco phone?
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23:57.00tflgen2ghost75: no, just sip soft clients

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