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| 07:27.51 | NirS | Hi All |
| 07:27.53 | NirS | anybody home ? |
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| 07:38.25 | Zogot | ahoy NirS |
| 07:38.41 | Zogot | fix the issue? |
| 07:40.37 | NirS | nope |
| 07:40.44 | NirS | but at least I got the lock location |
| 07:41.00 | NirS | appears to be something that pops up on the dev list once in a while |
| 07:41.04 | NirS | it's a lock in can_sip |
| 07:41.20 | NirS | but I think I'll just move to chan_pjsip for this system and be done with it |
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| 07:50.26 | Zogot | NirS: did you try the later versions of 13? |
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| 08:27.42 | bounceman | I have defined a variable under globals context, once asterisk reads it from the dialplan. How is it stored? In RAM? |
| 08:28.04 | bounceman | Because the variable is set to 1 (myvar=1) and during calls we change its value, however on a reload it returns to the value 1 |
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| 08:48.39 | ChannelZ | If you want 'persistent' data, use the AstDB |
| 08:50.23 | ChannelZ | see the DB dialplan function |
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| 09:12.26 | bounceman | ChannelZ: Ok, we already use asterisk db |
| 09:12.27 | bounceman | :-) |
| 09:13.18 | bounceman | So I am suppose to add the global variable in the extensions table? |
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| 09:23.01 | bounceman | Anyone? I wish to store a global variable in the database so that I can update its value without having it reset on reload |
| 09:52.05 | jamesc | does the ODBC driver for asterisk support multistatement BEGIN TRAN t1 INSERT ... update COMMIT TRAN t1 ? |
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| 10:01.57 | jamesc | bounceman: I believe there is a setting clearglobals? is this relevant |
| 10:02.57 | bounceman | Never heard of such setting, not present in sip.conf |
| 10:07.42 | jamesc | bounceman: clearglobalvars=no |
| 10:10.11 | bounceman | hm, in general section of sip.conf? |
| 10:10.26 | jamesc | no extensiosns |
| 10:10.54 | bounceman | under a context names general? Specifically? |
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| 10:11.04 | bounceman | or can it be placed under the globals context? |
| 10:11.30 | jamesc | Its under the globals context |
| 10:14.57 | bounceman | yeah, did not work. Resetted on reload |
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| 10:16.04 | jamesc | hmm http://www.voip-info.org/wiki/view/Asterisk+Dialplan+General |
| 10:16.20 | bounceman | Also clearglobalvars=no is present under the general context |
| 10:16.24 | bounceman | as a default |
| 10:16.50 | jamesc | sorry i mean general |
| 10:18.19 | bounceman | yeah, it was already present but I am afraid that file is not loaded. I tried another one now |
| 10:19.22 | bounceman | gah still |
| 10:19.30 | jamesc | does the ODBC driver for asterisk support multistatement BEGIN TRAN t1 INSERT ... update COMMIT TRAN t1 ? |
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| 10:39.32 | robmal | Anyone from Bulgaria? |
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| 14:45.53 | angryuser | Good day, when asterisk receives 404 or 480 from provider he sends me an 503, is there any way to make asterisk relay the exact cause codes besides custom scripting ? |
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| 14:53.40 | *** join/#asterisk Chubby (kmilasch@gateway/shell/xzibition.com/x-pljdkshbhizdzrxg) |
| 14:53.48 | Chubby | Hey guys, I have a channel thats being held open by an unknown IP/Source, SIP INfo: http://pastebin.com/qJgSyga5, How can I close this channel? |
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| 15:01.33 | voip | 503 means it's being blocked usually i think |
| 15:01.37 | voip | i could be wrong |
| 15:03.00 | Chubby | are you talking to me voip |
| 15:04.43 | voip | no angryuser |
| 15:05.16 | Chubby | oh alright |
| 15:05.27 | Chubby | Man I cannot work this out without force closing all channels |
| 15:05.28 | voip | soft hangup [tab] |
| 15:05.31 | Chubby | and pissing off everyone |
| 15:05.37 | Chubby | CLI tells me that command doesnt exist |
| 15:05.37 | voip | yea don't do that |
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| 15:05.47 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 15:05.54 | angryuser | voip: can i specify the cause code for Hangup like Hangup(1) or i need to do like hangup(AST_CAUSE_UNALLOCATED) ? |
| 15:06.08 | voip | channel request hangup |
| 15:06.27 | voip | channel request hangup [tab] |
| 15:06.42 | voip | channel request hangup $channel_you_want_closed |
| 15:06.46 | voip | thanks |
| 15:07.26 | Chubby | how do I match those channels against what I get from the tab complete from channel reuqest |
| 15:07.31 | voip | angryuser: asterisk sends the 503 when you get a 404 or 480 |
| 15:07.37 | Chubby | they look different |
| 15:10.53 | Chubby | so in the pastebin that I gave for sip show channel 712495-2e83 |
| 15:11.11 | Chubby | I don't see anything in that output that correlates with channel request hangup output |
| 15:13.46 | voip | sterisk2*CLI> channel request hangup SIP/4003-00000a2f |
| 15:13.55 | voip | :\ |
| 15:14.10 | voip | channel request hangup SIP/712495-2e83 |
| 15:14.14 | voip | try that |
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| 15:15.55 | Chubby | doesn't exist |
| 15:16.10 | Chubby | now a known channel |
| 15:19.03 | Chubby | http://pastebin.com/aVwqypJY here's the output voip |
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| 15:25.21 | voip | i don't have access to pastebin, at work |
| 15:25.24 | voip | they block it lol |
| 15:27.25 | Chubby | bummer |
| 15:27.36 | Chubby | 31.223.182.249 712405-2e83 0000000000 0000000000 ( 0.00%) 0.0000 0000000000 0000000000 ( 0.00%) 0.0000 |
| 15:27.36 | Chubby | 31.223.182.249 712405-2e82 0000000000 0000000000 ( 0.00%) 0.0000 0000000000 0000000000 ( 0.00%) 0.0000 |
| 15:27.37 | Chubby | 31.223.182.249 7123e6-2e7d 0000000000 0000000000 ( 0.00%) 0.0000 0000000000 0000000000 ( 0.00%) 0.0000 |
| 15:27.37 | Chubby | 31.223.182.249 7123f6-2e7f 0000000000 0000000000 ( 0.00%) 0.0000 0000000000 0000000000 ( 0.00%) 0.0000 |
| 15:27.43 | Chubby | Bridge/0x7f25209f09d8-input Bridge/0x7f25209f09d8-output Bridge/0x7f252165a468-output |
| 15:27.43 | Chubby | Bridge/0x7f264032f8b8-input Bridge/0x7f264032f8b8-output ConfBridgeRecorder/conf-3003-uid-1790814069 |
| 15:27.43 | Chubby | ConfBridgeRecorder/conf-3007-uid-349381862 IAX2/boston-2944 SIP/4000004265211-004329fa |
| 15:27.44 | Chubby | SIP/400000428510-004329f6 SIP/Mediant-004329f7 SIP/Mediant-004329f9 |
| 15:27.48 | Chubby | Not too floody, but sorry |
| 15:30.14 | Chubby | alright, watched it like a hawk til no active calls |
| 15:30.25 | Chubby | channel request hangup all |
| 15:30.36 | Chubby | It *DID NOT* close the 31.223 channels |
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| 15:52.44 | Chubby | awesome |
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| 16:11.22 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 16:17.50 | Chubby | Hey [TK]D-Fender |
| 16:17.57 | Chubby | Maybe you can help me out...I have a really tricky problem |
| 16:18.35 | Chubby | <PROTECTED> |
| 16:18.49 | Chubby | Now, I tried using sip show channel stats |
| 16:19.13 | Chubby | It gives this output http://pastebin.com/bLkC99xs and then trying to do Channel request hangup (tab), the bottom is that output |
| 16:19.19 | Chubby | I did try a force close all channels |
| 16:19.27 | Chubby | but it did not close the channels held open by the unknown source |
| 16:19.30 | Chubby | any ideas on what to do |
| 16:27.45 | [TK]D-Fender | core show channels concise |
| 16:43.19 | Chubby | Requested hangup on the two channels that came up, Peers are still active |
| 16:45.32 | [TK]D-Fender | Show it all |
| 16:45.46 | [TK]D-Fender | Then plan B : use an AMI Redirect to toss them off a cliff |
| 17:03.01 | Chubby | http://pastebin.com/efbMNd2W [TK]D-Fender, thanks |
| 17:11.15 | [TK]D-Fender | AMI time... |
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| 18:26.07 | drab | hi, apparently my google foo is weak and I cannot find some kind of web frontend to put on top of a vanilla asterisk 11 install to get a phone directory |
| 18:26.31 | drab | I don't nede anything fancy, basically just listing what's in sip.conf (extensions + description field) |
| 18:26.31 | Kunsi | a phone directory? |
| 18:26.47 | Kunsi | sip.conf doesn't define any extensions |
| 18:27.04 | drab | I could script it myself through AMI, but I thguoht there had to be something there |
| 18:27.29 | drab | eeer, right, I think "exntesion" back, sip clients definitions, I'd like to list those |
| 18:27.35 | drab | s/think/take/ |
| 18:39.01 | [TK]D-Fender | There is no concept of a directory in vanilla Asterisk |
| 18:39.08 | [TK]D-Fender | Wh is a person? |
| 18:39.20 | [TK]D-Fender | That very idea is something that only exists in your head, not Asterisk |
| 18:40.20 | [TK]D-Fender | And I doubt anyone is going to distribute a script for this. Who's to say all peers should be considered for the list? Ones you use for phones are mixed in just as fairly with those you'd use for providers, etc |
| 18:41.02 | [TK]D-Fender | Too many assuptions. Then, who would be allowed to see it? What about security? What kind of extra UI should it have? Logins? |
| 18:41.24 | [TK]D-Fender | So you can't just ask "why isn't this a thing, because it's a DIFFERENT thing to everyone who wants something like it |
| 18:44.47 | drab | [TK]D-Fender: makes sense, thank you |
| 18:46.59 | [TK]D-Fender | sense++ |
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| 20:01.19 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 20:03.18 | jeffspeff | so, it's april fools and I'm wanting to prank some of our employees. i want * to call 2 extensions and it appear to each person that they have called each other, hoping to result in each of them thoroughly confused and discussing who actually called who |
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| 20:13.19 | newtonr | jeffspeff, I used to set it up so that Asterisk would call and connect my friend's cell to the local pizza place, set the CallerID to the Pizza place so it looked like they were calling him |
| 20:14.20 | newtonr | same sort of idea. it was fun |
| 20:14.54 | jeffspeff | newtonr, so, your friends cell had the pizza place's CID and the pizza place got your friends CID? |
| 20:15.30 | newtonr | jeffspeff, nah in my case I didn't bother changing the CID on my call to the pizza place |
| 20:15.49 | jeffspeff | any suggestion how i should go about this? |
| 20:15.56 | jeffspeff | i'm sure something involving call files |
| 20:17.13 | jeffspeff | I think i just found what i was looking for http://serverfault.com/questions/219373/asterisk-trying-to-use-call-files-to-create-a-conference-call-between-two-dyna |
| 20:17.53 | newtonr | yeah, call file, originate a local channel to an extension where you set callerid and dial party A, then have the callfile connect on answer to dial Party B (you may need to also set CallerID in the callfile for that leg) |
| 20:18.22 | newtonr | that should work I think, I'm really tired today |
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| 20:51.28 | morsing | What is monitored and unmonitored for peers? |
| 20:54.11 | newtonr | whether the option qualify is used or not |
| 20:54.16 | [TK]D-Fender | Whether you use qualify to check up on them to see if they respond |
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| 20:57.02 | morsing | Ok, thanks |
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