| 00:14.58 | *** join/#asterisk X-Rob (sid14615@gateway/web/irccloud.com/x-fvcreiwjhnleovgm) |
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| 01:03.31 | rexwin | will dialplan reload cut off any live ongoing call connections? |
| 01:04.52 | WIMPy | no. |
| 01:05.02 | WIMPy | But they will continue in the new dialplan. |
| 01:10.09 | rexwin | I want to change the dialplan like [ Context 'ivr-218' created by 'pbx_config' ] |
| 01:10.19 | rexwin | how would we do that? |
| 01:11.40 | rexwin | from cli of course as I donot have shell access |
| 01:13.14 | WIMPy | Well, there's dialpan add|remove. But you might be the first to try them in ages. So tell is if that works. |
| 01:20.02 | rexwin | No such command 'dialplan remove local' (type 'core show help dialplan remove local' for other possible commands) |
| 01:21.57 | rexwin | it worked now |
| 01:22.04 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 01:26.11 | rexwin | In my test server it worked but in my production server it didnot |
| 01:26.26 | rexwin | No such command 'dialplan remove context ivr-224' (type 'core show help dialplan remove context ivr-224' for other possible commands) |
| 01:37.03 | [TK]D-Fender | Why are you trying to do do this in the first place? |
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| 01:58.25 | rexwin | I want to do that becase the ivr is not working for this context. |
| 02:01.00 | rexwin | I remove IVR entry for Inbound route and removed Inbound route also but still it shows up in dialplan show output |
| 02:01.26 | ChannelZ | is this elastix/freepbx/some other GUI? |
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| 02:04.45 | [TK]D-Fender | Show us the actual failure |
| 02:05.05 | [TK]D-Fender | And an IVR does "fail" for a context. A context IWS an IVR |
| 02:05.23 | [TK]D-Fender | And by killing a context like that all you're going to do is break a Goto that targets it |
| 02:05.38 | [TK]D-Fender | The entire premise behind this fix is a mistake. |
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| 04:42.31 | rexwin | https://www.dropbox.com/s/jyffz77mao6o7c2/Screenshot%202015-02-09%2021.40.20.png?dl=0 |
| 04:43.15 | rexwin | this is the diff between two ivr contexts. I want to add the extra lines to right side section. How do I do that? |
| 04:43.52 | [TK]D-Fender | ... |
| 04:44.14 | [TK]D-Fender | You ar using FreePBX.... |
| 04:44.22 | [TK]D-Fender | You should be using the GUI to do all of this |
| 04:44.30 | [TK]D-Fender | Messing around manually is a waste of time |
| 04:44.46 | [TK]D-Fender | The moment you commit any change you will blow away anything you did by hand in there |
| 04:44.57 | [TK]D-Fender | You are going about this the completely wrong way |
| 04:45.17 | [TK]D-Fender | Clearly you didn't create menu options for your IVR 224 |
| 04:45.24 | [TK]D-Fender | SO GO MAKE THEM |
| 04:46.27 | [TK]D-Fender | What actually tells you you SHOULD have something there in the first place? |
| 04:46.53 | [TK]D-Fender | You are looking at what was created as evidence instead of showing what CREATES it in the first place. |
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| 04:55.44 | garymaurizi | I have a copy of chan_sip.c and I need to find out what version of asterisk it belongs to, how would you go about doing this if you had to? |
| 04:55.55 | garymaurizi | there's no change history in the header, just commits.. and no versoning |
| 04:59.47 | rexwin | I changed all the IVR Options and IVR Entries to reflect the working IVR settings but nothing gets applied when changed from GUI. I did dialplan reload after every change |
| 05:01.45 | [TK]D-Fender | rexwin, You should not be touching the dialplan manually for this period |
| 05:02.07 | [TK]D-Fender | rexwin, Show us what you're actually doing. full proper screenshots of the ivr's, etc. |
| 05:06.26 | [TK]D-Fender | You also shouldn't be doing reload's from CLI. The GUI "Apply" button does it all for you. |
| 05:10.12 | rexwin | yeah, I apply through gui console |
| 05:11.37 | [TK]D-Fender | So youo should be doing "dialplan reload". |
| 05:11.42 | [TK]D-Fender | Show us the actual IVR's |
| 05:11.49 | [TK]D-Fender | shouldn't* |
| 05:11.57 | [TK]D-Fender | gah... typing going ot the window tonight... |
| 05:43.44 | phix | sup? |
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| 06:01.19 | WIMPy | 's down |
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| 08:26.15 | Rico | hello |
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| 08:26.45 | Rico | question about hints and devstate : One of my phones is going to the "Busy" state when used |
| 08:27.10 | Rico | when another call comes to this phone, it goes to state 'InUse&Ringing' |
| 08:27.40 | Rico | I've set busylevel=1 on the sip acount |
| 08:28.23 | Rico | what can I do to only allow one incoming call on this phone ? |
| 08:29.01 | JCONeill | MWI? |
| 08:30.07 | Rico | ScotzMan: ? I don't want the call to be sent to the phone if it is already used |
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| 08:33.11 | Rico | oops, I hit ctrl+L and it cleared the window |
| 08:33.31 | Rico | any new message since ScotzMan response ? |
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| 09:27.53 | Rico | anobody ? |
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| 10:00.25 | Gugge | Rico: just check if the phone is in use before you Dial() it? |
| 10:00.54 | Rico | Gugge: isn't asterisk supposed to do it automaticaly ? |
| 10:01.02 | Rico | when setting busylevel ? |
| 10:01.51 | Gugge | no |
| 10:02.56 | Gugge | call-limit sets a limit, busylevel sets how many calls is needed to be marked as busy (a busy phone can make calls) |
| 10:03.03 | Gugge | as far as i can tell |
| 10:07.36 | Rico | Gugge: yes that's it |
| 10:08.03 | Rico | but my phone is marked as busy and asterisk continues to send calls to it |
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| 10:33.05 | Gugge | Rico: as i wrote, a busy phone can still receive and make calls. |
| 10:33.13 | Gugge | busy is just a state |
| 10:34.11 | Rico | but isn't asterisk supposed to sped send him calls if busylevel=1 ? |
| 10:34.18 | Rico | s/sped/send/ |
| 10:34.58 | Gugge | busylevel just sets when the state is set to busy |
| 10:35.01 | Gugge | it has nothing to do with limits |
| 10:38.51 | phix | Gugge: yeah like being old is just a state too |
| 10:39.27 | Gugge | something like that :) |
| 10:43.39 | Rico | ok |
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| 13:39.18 | g-maurizi | I need help with a simple question, how can I temporarily change SIP/103&Custom:DND10 State:Unavailable to another state like NOT_INUSE or UNKNOWN? |
| 13:52.28 | *** join/#asterisk areski (~areski@80.174.128.98.dyn.user.ono.com) |
| 13:59.01 | WIMPy | 'devstate change ...' |
| 13:59.27 | WIMPy | That only works for the custom ones, off course. |
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| 14:06.41 | g-maurizi | so I'm confused about some things.. im trying to understand dialplans hints and device states... when I set a blf key to BLF and put in only the extension 1 i see my phone as a subscriber for SIP/104&Custom:DND10 does freepbx use do not disturb state as the state for presence watching:? |
| 14:07.27 | WIMPy | You need to ask in #freepbx |
| 14:08.17 | g-maurizi | I am trying to change the state from Unavailable to "NOT_INUSE" when the extension isn't in use.. my hardphones don't respond to statestring="terminated" in chan_sip but they do respond to IN_USE/BUSY etc.. |
| 14:09.10 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:09.14 | g-maurizi | in chan_sip.c AST_EXTENSION_UNAVAILABLE uses the statestring "terminated" it used to be "confirmed" which made these blf keys work. patching chan_sip breaks MWI .. so im just trying to come up with another solution here |
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| 14:25.23 | yun1989 | hello all |
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| 14:25.40 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 14:26.16 | yun1989 | eu tento fazer um registro de chamada a trabalhar no meu sistema, mas eu não tenho sucesso |
| 14:26.30 | yun1989 | i try to make a call record working in my system but i don't have sucess |
| 14:26.36 | yun1989 | in english |
| 14:26.39 | yun1989 | sorry |
| 14:27.28 | yun1989 | i use this exten => _66XX,n,Record(/tmp/prompt${EXTEN:2}:wav) |
| 14:27.41 | yun1989 | but the call is not recording |
| 14:30.02 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 14:35.28 | [TK]D-Fender | yun1989: Show us the complete call attempt from CLI |
| 14:35.30 | [TK]D-Fender | ~pb |
| 14:35.30 | infobot | 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.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 14:35.32 | [TK]D-Fender | ^^^ |
| 14:37.46 | yun1989 | http://pastebin.com/dUEQaYXv |
| 14:38.25 | [TK]D-Fender | Go end the call and show us the folder |
| 14:39.58 | yun1989 | the folder is empty |
| 14:40.13 | *** join/#asterisk ayrjola (~ayrjola@80.248.109.192) |
| 14:42.24 | [TK]D-Fender | Does it actually exist? Is it owned by Asterisk? |
| 14:42.27 | yun1989 | this is my dialplan http://pastebin.com/NP0ZGhm2 |
| 14:42.58 | *** join/#asterisk zpotoloom (~zpotoloom@173.88.50.84.sta.estpak.ee) |
| 14:43.25 | [TK]D-Fender | "ls -la /var/spool/asterisk/records" |
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| 14:44.39 | yun1989 | http://pastebin.com/9yj4LU0H |
| 14:44.54 | yun1989 | the result ls -la /var/spool/asterisk/ |
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| 14:48.11 | [TK]D-Fender | drwxr-xr-x 2 root root 4096 Feb 10 14:41 records |
| 14:48.25 | [TK]D-Fender | I hope you're not running * as root.... |
| 14:48.37 | [TK]D-Fender | (though you'd HAVE to be for this to work in its current state) |
| 14:50.23 | yun1989 | the result is the same when is root or not root |
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| 14:56.00 | [TK]D-Fender | Try putting it somewhere else |
| 14:56.17 | [TK]D-Fender | And You didn't actually show me what I requested |
| 15:01.41 | yun1989 | I change the location for /var/spool/asterisk/monitor |
| 15:01.51 | yun1989 | and asterisk creat the file |
| 15:01.53 | yun1989 | -rw-r--r-- 1 root root 98348 Feb 10 15:01 audi.wav |
| 15:02.07 | yun1989 | but after some seconds the file desapear |
| 15:02.10 | [TK]D-Fender | folder permissions issue then |
| 15:02.34 | [TK]D-Fender | Go make some place separate for this entirely |
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| 15:03.00 | yun1989 | chmod -R 775 /var/spool/asterisk/monitor ? |
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| 15:07.25 | [TK]D-Fender | sure |
| 15:08.37 | yun1989 | i don't understand your solution |
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| 15:08.53 | yun1989 | please explain again |
| 15:10.45 | yun1989 | i change the destination for home folder and the result is the same |
| 15:11.01 | [TK]D-Fender | I'm not sure why your file is going away if that is still happening. |
| 15:11.19 | [TK]D-Fender | yun1989i change the destination for home folder and the result is the same <- And you aren't being SPECIFIC as to WHICH home folder. |
| 15:12.03 | yun1989 | @[TK]D-Fender is the /home/user folder |
| 15:12.49 | [TK]D-Fender | "core show settings" <- |
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| 15:14.14 | yun1989 | http://pastebin.com/FLcMMCVK |
| 15:14.41 | yun1989 | the dialplan don't establish the call but it's init the record |
| 15:15.01 | yun1989 | i think this configurantion is not correct can you help me in this ? |
| 15:15.11 | yun1989 | 2[TK]D-Fender |
| 15:16.07 | [TK]D-Fender | You are using the wrong command ENTIRELY |
| 15:16.23 | [TK]D-Fender | Record is to record that channel IMMEDIATELY for you to save a file. |
| 15:16.35 | [TK]D-Fender | This is NOT for recording a call you intend to dial AFTER. |
| 15:16.43 | yun1989 | the resultp core show settings http://pastebin.com/jLMrSmbr |
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| 15:18.43 | yun1989 | @[TK]D-Fender how record the call ? |
| 15:18.55 | [TK]D-Fender | "core show application monitor" <- |
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| 15:20.49 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:22.57 | yun1989 | @[TK]D-Fender http://pastebin.com/i2EcCuJu |
| 15:23.20 | yun1989 | what i make "core show application monitor ? |
| 15:25.45 | [TK]D-Fender | go USE IT |
| 15:25.55 | yun1989 | how ? |
| 15:26.10 | [TK]D-Fender | Call it before your dial |
| 15:26.24 | yun1989 | i run this command in CLI and the result is http://pastebin.com/i2EcCuJu´ |
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| 15:26.47 | [TK]D-Fender | yes, those are the INSTUCTIONS for how to use it in your dialplan |
| 15:30.11 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 15:32.39 | yun1989 | thank for your help @[TK]D-Fender |
| 15:33.04 | *** join/#asterisk sbrath (~sbrath@unaffiliated/sbrath) |
| 15:35.08 | DivideBy0 | when I originate calls from my stasis app, I don't always get the dst in my cdr report. The call is answered. debug shows DEBUG[3279] cdr.c: Finalized CDR (with the fields that match) where else do I look? |
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| 16:01.51 | darkdrgn2k3 | mornign all |
| 16:02.01 | darkdrgn2k3 | i know its a long shot but any one here familiar with Shore Tel at all? |
| 16:07.22 | *** join/#asterisk timahvo1 (~rogue@105.163.204.105) |
| 16:08.53 | WIMPy | On shore or off shore? |
| 16:08.58 | WIMPy | scnr |
| 16:11.02 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-fmpsgfkawavguqbb) |
| 16:11.06 | darkdrgn2k3 | waa waa.. "shoretel" to company. |
| 16:11.20 | darkdrgn2k3 | i just dont knwo where you punch down the pairs on a rattail |
| 16:11.25 | darkdrgn2k3 | for an analog line |
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| 16:17.03 | WIMPy | wonders how to connect wires to software. |
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| 16:24.22 | sbrath | I just built a fresh asterisk 11 install (PIAF) and I'm having a wierd issue, when I do a "recording" or a voice mail, it's all like it's recorded in fast forward? I think this might be a bus-master issue as i'm using an old HP DC5800 workstation... Ideas? |
| 16:25.01 | WIMPy | 'timing test' |
| 16:26.08 | WIMPy | Although, if it's really like fast forward, that sounds more like a format issue. |
| 16:26.55 | sbrath | It's like it plays it back at double speed. Both for Voicemails and Recordings. |
| 16:27.30 | sbrath | Where is timing test? |
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| 16:27.46 | WIMPy | Enter that at the *CLI |
| 16:27.52 | sbrath | ahh. |
| 16:28.03 | WIMPy | But That sounds like a conversion issue. |
| 16:28.41 | sbrath | I tried the hdparm -t /dev/sda thing and it looks ok. |
| 16:29.13 | [TK]D-Fender | not STORAGE timing |
| 16:29.18 | [TK]D-Fender | CPU <- |
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| 16:30.01 | darkdrgn2k3 | what is the difference between the "ring" and the "tip" pairs? |
| 16:30.08 | WIMPy | Not CPU. Application. |
| 16:30.26 | sbrath | I guess it's worth saying that there is a Wildacrd in the computer too, and it is acting as the timing source for the PRI |
| 16:31.01 | WIMPy | darkdrgn2k3: They aren't both pairs. They together become a pair. And uaually it doesn't matter which is which. |
| 16:31.30 | darkdrgn2k3 | Ahhh ok that makes sense! |
| 16:31.31 | WIMPy | sbrath: Then your timing should hopefully be good. |
| 16:31.43 | sbrath | darkdrgn2k3: The polarity of the T+R is usually only critical for Fax machines.... |
| 16:32.12 | [TK]D-Fender | Except as you're running DC you'll invert the push-pull of your speakers, etc |
| 16:32.19 | WIMPy | No. Fax machines don't care, either. |
| 16:32.22 | [TK]D-Fender | Get them right. Don't do stupid stuff :) |
| 16:32.34 | WIMPy | Some automativc switches may care. But that's all history. |
| 16:32.56 | darkdrgn2k3 | so according to this my analog trunk shoudl be on pair 1.... damit i hate non voip technology |
| 16:33.27 | WIMPy | Analog is evil. |
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| 17:28.23 | mattsl | Can anyone recommend a good tutorial on setting up some SIP testing? I have a server that seems to be having some intermittent issues with inbound calls. I would like to make a few thousand test calls over a couple days and see if I can find any patterns of problems. |
| 17:28.44 | Qwell | mattsl: sipp is a great tool for that |
| 17:29.09 | Qwell | https://wiki.asterisk.org/wiki/display/AST/Basic+Test+with+SIPp |
| 17:30.39 | stefan27 | how do i query other channels variables from dialplan? |
| 17:33.19 | stefan27 | other SIP channels i mean |
| 17:35.55 | stefan27 | if SIP/A has done Set(Boo=2) and then SIP/B wants to know A's value of Boo |
| 17:37.09 | stefan27 | the only way of accessing Boo without A executing dialplan that im aware of is the cli "dialplan show chanvar SIP/A..." but thats impractical |
| 17:40.37 | [TK]D-Fender | stefan27: How is B going to know the exact channel to look at? |
| 17:43.01 | *** join/#asterisk CeBe (~CeBe@port-92-200-45-135.dynamic.qsc.de) |
| 17:47.21 | stefan27 | it should look at any channel with interface name SIP/A |
| 17:47.27 | stefan27 | which it knows from a channel-variable |
| 17:47.46 | [TK]D-Fender | that is PART of the cahnnel name |
| 17:48.13 | stefan27 | there'll only be one SIP/A-channel in asterisk at any given time in my usercase, when there's not i accept errors |
| 17:50.28 | stefan27 | can i query asterisk for channels with names matching SIP/A and then once i have the exact channel name get a variable with some dialplan function? |
| 17:54.22 | robmal | I have a bit harder version of above: what is the best way to match CDRs between PBXes? Assuming i have sh*tload of them. |
| 17:54.36 | robmal | Any best practices? |
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| 18:30.20 | mjordan | stefan27: you can either use global variables, or the AstDB. |
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| 18:30.30 | mjordan | or |
| 18:30.40 | mjordan | always forgets the name of this function |
| 18:32.53 | mjordan | SHARED: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_SHARED |
| 18:34.42 | [TK]D-Fender | mjordan: persistent until * restart? |
| 18:39.48 | mjordan | SHARED wouldn't be. But that would let SIP/A shared the value of Boo with other channels if it so desired, and the prefix is acceptable for locating the shared variable area |
| 18:40.13 | mjordan | to persistent until restart, you'd have to store it in a global variable or in the AstDB |
| 18:40.22 | mjordan | of course, AstDB would go longer than a restart... |
| 18:41.47 | *** join/#asterisk cklimos (~Claude@207.35.85.92) |
| 18:42.31 | [TK]D-Fender | Oh, so the space is tied to a channel that sets something in it? |
| 18:43.16 | [TK]D-Fender | A comes up. Sets value in namespace. Namespace/var set. Channel ends = all values gone? |
| 18:43.27 | [TK]D-Fender | "A" |
| 18:44.46 | mjordan | IIRC, yes |
| 18:44.51 | mjordan | I haven't used this one much. |
| 18:45.08 | mjordan | So if I'm a channel, and I use Set(SHARED(foo)=bar), then that sets it in my namespace |
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| 18:45.29 | mjordan | Another channel can use ${SHARED(foo,SIP/A)} it pulls it out of that channel's namespace |
| 18:45.44 | [TK]D-Fender | Certainly viable, and blanking on end of channel is a sign not to bother continuing making it a possibly better idea than AstDB |
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| 19:27.48 | JerJer | meep meep |
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| 19:39.54 | TazzNZ | Hey all - anyone here know off (or run) ITSP's that can give me DDI's in the Carribean ? |
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| 21:26.55 | [TK]D-Fender | checkout time, BBIAB |
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| 21:59.41 | stkoch | WIMPy: Hi, today they have brought a new switch. Now the packet loss is less than 2 % instead of less than 50 % |
| 22:00.40 | stkoch | If the packet loss is less than 1 % the calls don't abort, some hearable short jitter let the connection alive |
| 22:01.43 | stkoch | The longest call was about 18 minutes, but the termination problem still exist |
| 22:03.14 | stkoch | I have tried to set genericplc => true in [plc] section of codecs.conf but the asterisk debug messages doesn't show that plc was enabled |
| 22:03.44 | stkoch | There is only a message that codecs.conf was parsed |
| 22:03.44 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 22:06.12 | stkoch | [TK]D-Fender: hi back. i have posted some messages 5 minutes before you logged in, see here: http://pastebin.com/sejSyxLQ |
| 22:06.51 | stkoch | but the codecs.conf was not there initially I have created it |
| 22:07.22 | stkoch | How to ensure that genericplc was enabled? |
| 22:13.04 | WIMPy | I think the interesting question is where or in what circumstances that would be used. |
| 22:13.15 | WIMPy | And I have NFI. |
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| 22:15.11 | stkoch | WIMPy: What is NFI? |
| 22:15.37 | WIMPy | No F... Idea |
| 22:15.39 | rrittgarn1 | related to NFC... but slightly different... |
| 22:15.53 | rrittgarn1 | (Clue vs Idea) |
| 22:16.58 | stkoch | https://wiki.asterisk.org/wiki/display/AST/Requirements+for+PLC+Use |
| 22:17.30 | WIMPy | Does the device you're using have multiple hardware ports? |
| 22:20.05 | stkoch | WIMPy: yes, 2xfxs, 1xS0 and one ADSL port |
| 22:20.32 | WIMPy | And I guess you can make calls between those ports without issues? |
| 22:21.03 | stkoch | WIMPy: not tested, but with packet loss < 1% voip works fine |
| 22:21.34 | WIMPy | It's probably more about jitter than PL. |
| 22:24.07 | stkoch | The best would be if there is packet loss/jitter asterisk inserts dummy silence into the alaw stream so that the lantiq soc doesn't come in touch with the jitters |
| 22:24.48 | stkoch | Or asterisk doesn't hangup and does instead reopen the channel after failure |
| 22:24.59 | WIMPy | Maybe someone who knows about the media handling can tell if it might be worth to try to add a local channel. |
| 22:25.39 | WIMPy | Or the channel driver needs to be ... less bad. |
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| 22:39.07 | stkoch | have I to change exten => USERNAME,1,Dial(TAPI/1) to exten => USERNAME,1,Dial(local/TAPI/1/nj) that local channel is used with nj for jitter buffer? |
| 22:39.23 | stkoch | it's in [sipgate_in] section in extensions.conf |
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| 23:35.04 | stkoch | WIMPy: thanks |
| 23:37.15 | file | falls over |
| 23:39.10 | j4janicej | flaps on file |
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