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| 03:24.05 | *** join/#asterisk denbeiren (~quassel@ptr-178-51-219-117.dyn.mobistar.be) |
| 03:24.15 | denbeiren | Hi all |
| 03:26.46 | denbeiren | i have a mobotix cam (doorstation) and a grandstream phone,.. both are configured so i can look at the cam images trough the phone. I was thinking of setting up a asterisk server, but as i don't know anything about asterisk, i was wondering how it could help me in doing some new and exciting stuff |
| 03:34.10 | *** join/#asterisk saint_ (~saint_@unaffiliated/saint-/x-0540772) |
| 03:34.25 | [TK]D-Fender | Depends what you find "exciting" |
| 03:36.29 | [TK]D-Fender | So far I see no standard telephony protocol support on the Mobotix |
| 03:36.58 | [TK]D-Fender | So unless you can bridge that gap I'm wondering how * will play a role in rbing them together into the mix |
| 03:37.28 | denbeiren | it would come in handy when someone rings the doorbell and i'm not home it would call me on the cell |
| 03:38.00 | [TK]D-Fender | Do you see VoIP support in that unit's manuals somewhere? |
| 03:38.07 | [TK]D-Fender | I haven't found reference to it yet.... |
| 03:38.18 | denbeiren | i believe there is,.. just a sec |
| 03:38.53 | *** join/#asterisk ipn (4a6c86f1@gateway/web/freenode/ip.74.108.134.241) |
| 03:40.38 | ipn | can anyone help me out with a dialplan issue - I am trying to execute "exten => s,n(erase), MYSQL(Query resultid ${connid} DELETE FROM myTable WHERE extension = 101 ) but I am getting the following error in the CLI " app_mysql.c:496 aMYSQL_fetch: aMYSQL_fetch: missing some arguments" and I cannot figure out what arguments it needs. I cannot find any documentation via gogle |
| 03:41.02 | denbeiren | https://www.mobotix.com/other/content/view/full/225223 |
| 03:41.39 | denbeiren | the dutch version of the site does speak of "ip telephony" |
| 03:42.46 | *** part/#asterisk ipn (4a6c86f1@gateway/web/freenode/ip.74.108.134.241) |
| 03:43.20 | [TK]D-Fender | Nothing I see on that page... |
| 03:43.47 | denbeiren | http://prntscr.com/9uwoni i have this in the settings |
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| 03:44.32 | ipn | can anyone help me out with a dialplan issue - I am trying to execute "exten => s,n(erase), MYSQL(Query resultid ${connid} DELETE FROM myTable WHERE extension = 101 ) but I am getting the following error in the CLI " app_mysql.c:496 aMYSQL_fetch: aMYSQL_fetch: missing some arguments" and I cannot figure out what arguments it needs. I cannot find any documentation via gogle |
| 03:46.13 | [TK]D-Fender | You mean in a config screen on the unit? |
| 03:46.22 | denbeiren | yes |
| 03:46.25 | [TK]D-Fender | Because I don't see those items on the product page you linked |
| 03:46.43 | denbeiren | it's not that clear on the product page, you're right |
| 03:46.45 | [TK]D-Fender | If that's on the admin page, then SIP does give you some options probably |
| 03:46.52 | [TK]D-Fender | Go find out what codecs it supports |
| 03:48.11 | denbeiren | http://prntscr.com/9uwptf |
| 03:49.13 | [TK]D-Fender | Looks pretty standard so far. So ... it's a phone... and can do video. |
| 03:49.46 | denbeiren | and motion detection, and it has a "bellbutton" |
| 03:49.51 | denbeiren | yep |
| 03:50.36 | denbeiren | so i can probably get it configured to call a cellphonenr when the doorbell is pushed |
| 03:51.24 | denbeiren | then the question remains,.. what will a asterisk install give me "more" than what this mobotix does already? |
| 03:59.34 | *** join/#asterisk captain118 (~Adium@50-81-28-72.client.mchsi.com) |
| 04:20.17 | [TK]D-Fender | Don't know what it offers. |
| 04:20.24 | [TK]D-Fender | basically you need to realize what * is |
| 04:20.32 | [TK]D-Fender | Which is a telephony engine. |
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| 04:21.00 | [TK]D-Fender | Tis means video greetings, IVR's, voicemail, call hunting, anything you can put your mind to, including bridging to your phones |
| 04:22.50 | denbeiren | i have a * running as a vm,.. and pbx as its gui,.. |
| 04:23.16 | denbeiren | do you have advice on where to start reading or for a n00b tutorial? |
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| 04:40.09 | jimi_ | I have a dumb idea I think. I want to install asterisk on a pc in a closet and use it as an internal only sip server. I have two small kids 6 and 10, and a finance living in the house. I have wall mounted iPads in each room that we use for home automation, recipes, etc.... I would like to install an IOS sip client on them as well and use them for paging/intercomming. Thoughts? |
| 04:43.48 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 04:47.30 | [TK]D-Fender | ~book |
| 04:47.30 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 04:47.34 | [TK]D-Fender | denbeiren, ^ |
| 04:47.45 | [TK]D-Fender | And the official WIKI |
| 04:48.06 | denbeiren | ok,.. i'll have a look at that one |
| 04:48.15 | [TK]D-Fender | jimi_, sounds liek it'll all work so far... |
| 04:49.07 | jimi_ | [TK]D-Fender, Thanks. just wanted to make sure I wasn't heading down the wrong path. |
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| 04:56.35 | Lope | I'm going to finish my asterisk setup today. All the basics are done. The last thing I need to do is setup some variables and gotos |
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| 05:00.17 | Lope | Basically I want to check if the extension is in use. If it's busy and did-you-know-1 has not been played, then play did-you-know-1. then go back to checking if the extension is busy. Then after that fall through to did-you-know-2 and so on. |
| 05:09.02 | [TK]D-Fender | So far sounds like you want a queue.... |
| 05:09.25 | [TK]D-Fender | Which eliminates all of that work |
| 05:13.22 | Lope | oh cool |
| 05:13.30 | jimi_ | kicks himself for trying to use prebuilt packages and just installs from source |
| 05:13.33 | Lope | Thanks, will look that up. |
| 05:14.02 | Lope | Is there an if statement I can use, instead of GotoIf? |
| 05:14.39 | Lope | like normal C/Basic style. If (condition) {do stuff}; |
| 05:15.12 | Lope | (eh, that's not valid C or basic, but you get the idea :)) |
| 05:15.46 | Kunsi | jimi_: keep in mind iOS terminates non-foreground apps after some time (i think 10 minutes) |
| 05:16.31 | jimi_ | Kunsi, Ah, you are correct. Didn't think about that. I might have to whip up something in Xcode and use diawi to distribute it ... unless maybe there is an existing Sip client that can handle that |
| 05:17.55 | jimi_ | Kunsi, or if that fails, just grab some cheap GrandSucks and set them to auto answer in the kids rooms :P |
| 05:18.40 | [TK]D-Fender | GotoIf is it for the straight conditionals. There is als While() |
| 05:19.12 | Lope | Oh great, while sounds perfect. |
| 05:19.21 | Lope | There's also ExecIf |
| 05:20.33 | Lope | Has anyone got a sublime syntax highlighting file for extensions.conf? |
| 05:20.43 | [TK]D-Fender | Good if you are making a single thing conditional |
| 05:20.58 | [TK]D-Fender | I'm sure Google would ahve that answer faster than us |
| 05:21.19 | Lope | Just thought you guys would have some good solution since you do this all the time. |
| 05:21.22 | Kunsi | vim does a good job, Lope |
| 05:23.21 | Lope | Nice. Found something :) https://github.com/pnlarsson/SublimeAsteriskConfig |
| 05:23.30 | Lope | Kunsi: thanks, but I've not learned VIM yet. |
| 05:23.52 | Lope | Ah, it's so much easier to read with highlighting. |
| 05:25.35 | jimi_ | ha, it took me longer to install asterisk via apt-get than to build from source right now \o/ |
| 05:26.30 | Lope | jimi lucky you. It took me half an afternoon to build. But that's cos it was quite a hassle figuring out all the components I needed to have installed and which build scripts to run etc before building to get the features I wanted. |
| 05:26.54 | Lope | The actual build was fairly quick though. Even on a single core cheapest digitalocean instance. |
| 05:28.17 | jimi_ | all i did was literally wget <url>, tar zxfv asterisk-something.tar.gz, cd asterisk*, ./configure apt-get install libxml-dev (curses-dev...etc) make, make install |
| 05:28.38 | Lope | yeah I also installed pjsip and loads of other stuff. |
| 05:29.00 | Lope | My asterisk is pimped to the max. It's got the 20" rims, hydraulics and a mp3 sound system. |
| 05:30.15 | jimi_ | oh god |
| 05:30.32 | jimi_ | [TK]D-Fender, Kunsi thanks for the help/feedback |
| 05:30.59 | jimi_ | Im gonna head to bed (just got out of hospital w/ pneumonia yesterday) and no patience to read this other bullshit |
| 05:31.04 | *** part/#asterisk jimi_ (~jimi@unaffiliated/tuxguy) |
| 05:33.29 | Lope | #grumpy |
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| 08:03.18 | snadge | happy australia day :D |
| 08:13.19 | Lope | Happy day bro |
| 08:13.44 | Lope | Does anyone have an example of an Asterisk queue? I found this but parts are outdated. http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue (example 1) |
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| 08:16.34 | Lope | It sounds like Application_BackGround will play my choice of sound files instead of just making the ringing sound, while the extension is ringing |
| 08:20.00 | Lope | How is Example 3? decent or outdated? |
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| 08:28.39 | Lope | What do I do with my priorities with relation to While loops? |
| 08:31.58 | Lope | Here's my use case. I'm using ; instead of newline so I don't flood the channel. exten => 12345,1,NoOp(); same => n,Answer(); same => n,While(); same => n,Blah(); same => n,Blah2(); same => n,EndWhile; same => n,SomethingAfterTheLoop() |
| 08:32.42 | Lope | As you can see I've got n outside the loop flowing to n inside the loop, flowing to n outside the loop again. It seems to me like it should be fine. |
| 08:32.58 | Lope | Because n is just next, regardless of what has executed before. |
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| 08:46.48 | wdoekes | Lope: those 'n's are translated to numbers when parsing the dialplan, so the above is as if you had written 2,Answer....6,EndWhile... |
| 08:47.11 | Lope | Oh I see, then that means there will be missing numbers? |
| 08:47.48 | Lope | What should I use instead of n? |
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| 09:00.52 | wdoekes | huh? no.. afaict, what you're doing is fine. I don't know how the while/endwhile works, but your priorities should be (and are): 1 n n n n ... |
| 09:01.06 | wdoekes | (which is translated to: 1 2 3 4 5 ...) |
| 09:01.16 | wdoekes | see 'dialplan show YOUR_CONTEXT_NAME' |
| 09:02.25 | Lope | okay thanks |
| 09:02.37 | Lope | I'm trying to setup a queue with agents. |
| 09:02.47 | Lope | How do agents translate to my chan_sip members? |
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| 09:05.33 | Lope | before, when I didn't have a queue. I just used Dial(SIP/bob) |
| 09:06.34 | Lope | Now the examples I've seen. In queues.conf you have [myqueuename]; member => Agent/1234 |
| 09:07.13 | Lope | I'm unfamiliar with this Agent/#### syntax. Could I use Agent/bob ? |
| 09:07.37 | Lope | or Agent/SIP/bob? Doesn't look right. |
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| 09:24.36 | Lope | Then in agents.conf I've got [agents]; ackcall=no; agent => 1234,0000,Agent1_Name; |
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| 09:42.39 | snadge | im trying to convert from using exten => h to a hangup handler.. and im having a bad time with it |
| 09:42.46 | wdoekes | Lope: do you need agents? you can just add member=>SIP/bob |
| 09:43.20 | snadge | my extensions.conf has heaps of sections in it.. and i just want to make a global hangup handler.. but im seeing an error: |
| 09:43.23 | snadge | [Jan 26 19:23:20] ERROR[28962][C-00000004]: app_stack.c:594 gosub_exec: Attempt to reach a non-existent destination for Gosub: (Context:handler, Extension:s, Priority:1) |
| 09:44.19 | wdoekes | snadge: how is that message not self-explanatory? |
| 09:44.33 | snadge | but the destination exists |
| 09:44.39 | snadge | im just trying to follow the example in the documentation |
| 09:44.56 | wdoekes | 'dialplan show handler' |
| 09:45.39 | snadge | [ Context 'handler' created by 'pbx_config' ] |
| 09:45.44 | snadge | in 1 context |
| 09:46.20 | wdoekes | do you see an s => 1 ? |
| 09:47.28 | snadge | well.. i have a same => s,1,NoOp() as the first line in that handler |
| 09:47.35 | snadge | which is probably wrong.. but im just following what im seeing here: |
| 09:47.42 | snadge | https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause |
| 09:49.12 | wdoekes | same => s,1,... is wrong |
| 09:49.17 | wdoekes | exten => s,1,Bla() |
| 09:49.20 | wdoekes | <PROTECTED> |
| 09:49.38 | wdoekes | you should see an error/warning if you 'dialplan reload' |
| 09:49.49 | wdoekes | if you don't, you should fix your logger.conf |
| 09:50.04 | Lope | wdoekes: thanks! |
| 09:50.47 | snadge | yeah i've implied the meaning of same .. our current dialplan doesn't use it.. i see the error now.. its just scrolled off the screen |
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| 09:54.08 | wdoekes | in the future: try to read all errors/notice/warnings. only ignore any if you're 100% sure you can |
| 09:54.11 | wdoekes | wiki is fixed |
| 09:57.38 | snadge | now im just trying to figure out the best place to put this hangup handler |
| 09:57.51 | snadge | we have lots of contexts.. default throws an error.. global makes it think its a global variable |
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| 10:23.36 | snadge | tempted to just pay someone to do it |
| 10:25.21 | snadge | im seeing the same behaviour from the hangup handler as i am from exten => h anyway.. incorrect disposition, incorrect destination number etc |
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| 10:36.37 | snadge | its like beating my head against a brick wall.. except more painful, and less satisfying |
| 10:39.16 | snadge | thats my opinion of asterisk 13.. if i paid for this software, i'd be pissed off ;) |
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| 10:53.51 | wdoekes | snadge: cdrs are unfortunately very unsatisfying |
| 10:54.08 | wdoekes | did you try the endbeforehexten option? |
| 11:09.15 | snadge | i did.. and fortunately im starting to find some success.. i think i still have my hangup context in the wrong place, but im seeing much more encouraging results now |
| 11:09.52 | snadge | i've just billed a test call correctly.. |
| 11:20.59 | snadge | oh wow.. setting endbeforehexten to no explicitly (the old default) .. creates 2 cdrs.. both with different missing information |
| 11:21.53 | snadge | i basically just wanted to a) switch to a more modern maintained base to extend its shelf life.. b) get a threaded chan_sip driver that was introduced in v12 |
| 11:22.21 | snadge | but the cdr stuff has been completely fucked over.. with no option to change it back the way it was.. (ie.. not fundamentally broken) |
| 11:22.59 | snadge | awesome ;) |
| 11:23.59 | wdoekes | sarcasm is generally served best in small portions |
| 11:25.04 | snadge | theres always a bright side to everything i guess.. if i leave things alone.. (ie, what we use in production).. then nothing is broken and life goes on |
| 11:25.27 | snadge | im just having a sook, because something i thought that should be relatively easy.. isn't |
| 11:26.07 | snadge | i can only suggest perhaps delaying the EOL of asterisk 11.. until some of these CDR "issues" have been sorted out |
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| 11:58.05 | snadge | hmm.. i have some more data.. it looks like it is writing the wrong cdr |
| 11:59.28 | snadge | 0x7f0190000b28 - Dispatching CDR for Party A SIP/someprovider_nn2-00000003, Party B <none> |
| 12:00.24 | snadge | in the database i can see a cdr with dstchannel set to SIP/someprovider_nn2-00000003 |
| 12:00.43 | snadge | except my hangup code does some additional processing.. then i see in the logs |
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| 12:01.50 | snadge | 0x7f0190007f98 - Dispatching CDR for Party A SIP/myuserid-00000002, Party B SIP/someprovider_nn2-00000003 |
| 12:04.51 | snadge | so i guess i need to find a way to set the hangup handler for the call that ends up creating a cdr.. ie.. the last leg, and not the one prior to that |
| 12:08.05 | Gugge | snadge: are you looking for the endbeforehexten option in cdr.conf ? |
| 12:12.21 | snadge | well.. it doesn't seem to help me.. if i change it from the default in v13 of yes.. to no.. then i get 2 cdrs for one call |
| 12:13.48 | snadge | the docs for hangup handlers say "Hangup handlers can also be attached to any call leg because of pre-dial routines." .. but no example of that |
| 12:20.24 | snadge | https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers .. google pulls up this though.. so i'll try setting the hangup handler from a pre dial routine |
| 12:21.09 | snadge | seems like an awful lot of faffing about though, just to do some work after a call has finished |
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| 12:35.50 | file | for your comment about CDRs there's an entire specification now for them detailing how they work - if you are seeing behavior that doesn't match that, then it's a bug and you should check the issue tracker for an existing one and if it does not exist then create one with the exact details to reproduce |
| 12:35.54 | file | the wiki page is at https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification |
| 12:35.56 | ipn | can anyone help me out with a dialplan issue - I am trying to execute "exten => s,n(erase), MYSQL(Query resultid ${connid} DELETE FROM myTable WHERE extension = 101 ) but I am getting the following error in the CLI " app_mysql.c:496 aMYSQL_fetch: aMYSQL_fetch: missing some arguments" and I cannot figure out what arguments it needs. I cannot find any documentation via google |
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| 12:42.13 | snadge | thanks file.. im reviewing that document now.. i added the hangup handler to a pre-dial routine and got lulworthy results.. my script was trying to add upstream peer details to the cdr (and failing of course because its already finalised at that point apparently) |
| 12:43.02 | file | mjordan recently added the ability to modify CDRs some as part of a blog post series for fixing an issue, it has not yet hit a release |
| 12:43.36 | file | ah sorry, that was for getting billsec data |
| 12:45.58 | snadge | we're just adding some custom fields to them.. and writing to writeable fields.. eg userfield |
| 12:46.14 | snadge | then it goes into a cdrdb.. mysql |
| 12:46.35 | snadge | previously you could just do that in the hangup context .. or hangup handler.. they appear to be roughly equivalent |
| 12:47.45 | snadge | but obviously in asterisk 12.. the cdrs have been split because of the new bridging behaviour..so, in a nutshell.. the cdrs are now missing the custom fields.. and sometimes userfield is not being updated, presumably because it happens after the cdr is inserted.. or to the "wrong" cdr |
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| 12:53.06 | wdoekes | snadge: did you try to add the upstream peer details into the userfield from a post-answer Dial handler? |
| 12:53.38 | wdoekes | Dial option U() iirc |
| 12:54.27 | snadge | the userfield just has the cost of the call in it.. or is supposed to anyway, if its a billable call.. or NOANSWER or a few other things |
| 12:54.39 | snadge | but its just ending up with the default value |
| 12:59.16 | snadge | strange.. its dispatching the cdr after setting the custom variables.. but they're missing from the cdr |
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| 13:16.39 | ipn | so I am assuming my error was simply because I was doing a fetch on a DELETE command. I was hoping for a validation method but the documentation sucks |
| 13:19.17 | snadge | this cdr spec is making my head explode.. and i dont understand why i need to know about it.. if theres just one a party and one b party |
| 13:19.38 | snadge | if thats the case.. then it should behave the way it always has.. create one cdr, with all the relevant information in it :| |
| 13:19.54 | snadge | its not fucking rocket science |
| 13:21.55 | snadge | apparently it cant even get the simplest case right.. how am i supposed to have any faith in anything even remotely complicated ;) |
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| 14:19.44 | Lope | I want to say something like If you want to leave a message, press 9 |
| 14:19.55 | Lope | What's the function to check if they press 9? |
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| 14:25.03 | ipn | exten => s,n,Read(password,enter-your-pin,5) -> this is what I am using for a pin check. not sure if it is exactly what you are looking for but you can extrapolate. There is also the Background() function |
| 14:26.36 | [TK]D-Fender | You normally never run a Background without a WaitExten() following it |
| 14:27.34 | ipn | correct |
| 14:27.38 | ipn | https://wiki.asterisk.org/wiki/display/AST/Creating+a+Simple+IVR+Menu |
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| 14:35.03 | snadge | has the threaded chan_sip driver been backported to asterisk 11? |
| 14:35.30 | [TK]D-Fender | Generally things don't get backported |
| 14:35.43 | [TK]D-Fender | and chan_sip is in "slow death mode" I wouldn't hold my breath |
| 14:37.07 | wdoekes | threaded chan_sip driver? I don't think much has changed between 11 and 13 in chan_sip |
| 14:37.13 | wdoekes | do you mean chan_pjsip? |
| 14:37.19 | snadge | i haven't looked into it that far but i've been told threaded chan_sip was introduced in v12 |
| 14:37.25 | [TK]D-Fender | wdoekes: Clearly not |
| 14:37.48 | snadge | im presuming that the realtime db doesn't work with pjsip |
| 14:38.07 | [TK]D-Fender | snadge: because? |
| 14:38.36 | snadge | because that would make things too easy.. and that's just not possible ;) |
| 14:38.39 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime |
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| 14:38.59 | [TK]D-Fender | You mean like this link I got as the FIRST RESULT of a simple google search? |
| 14:39.31 | *** join/#asterisk onebree (d1bf0df7@gateway/web/cgi-irc/kiwiirc.com/ip.209.191.13.247) |
| 14:39.37 | onebree | Hello, all |
| 14:43.09 | snadge | the idea of switching to pjsip is entertaining and all.. but when you have a billing system, and web interface which manages an existing realtime db with chan_sip |
| 14:45.42 | onebree | Speaking of pjsip, we are moving to it with our transition to Asterisk 13. |
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| 14:46.36 | onebree | I was tasked to work on video webrtc with *13, however, that was with the chan_sip driver. Looking at the settings in pjsip.conf, I do not see things for videosupport, say. |
| 14:48.03 | [TK]D-Fender | have you tried just allowing the codecs? |
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| 14:50.08 | onebree | I have not tried yet, as we are still in the process of converting. (We are doing realtime lookups with mysql) |
| 14:50.27 | onebree | I was just wondering if there was a direct config in pjsip for sip/videosupport |
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| 14:51.30 | [TK]D-Fender | maybe you're reading too much into it |
| 14:51.38 | Lope | When you do something like this: Read(choice|1) does the pipe mean that the 2nd option is omitted? |
| 14:51.54 | [TK]D-Fender | You should not be using a pipe in your dialplan at all |
| 14:52.07 | Lope | okay, so how can I omit option 2? |
| 14:52.21 | Lope | Read(choice,,1) |
| 14:52.29 | Lope | I mean parameter 2. Hows that ^ |
| 14:52.35 | onebree | Lope: correct. |
| 14:52.42 | Lope | ok, thanks. |
| 14:52.57 | onebree | it will take the default value for param 2 (which may be null, depending on the app) |
| 14:53.34 | onebree | [TK]D-Fender: Okay, I will try enabling the codecs I need, once I insert the users into the db |
| 14:55.31 | onebree | Lope: I read somewhere that pipes COULD be used to delineate params, but it is more common, and preferred, to use commas. |
| 14:58.17 | [TK]D-Fender | That died in 1.4 almost a DECADE ago. |
| 14:58.22 | wdoekes | in the olden days, it read: Read|choice||1, now you should write Read(choice,,1) |
| 15:01.05 | onebree | That's where I read it from then -- the first asterisk guide book (only one we have in the office), v1.2 |
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| 15:01.18 | Lope | Looks like read has changed, or the voip-info.org site is outdated. |
| 15:01.41 | Lope | Oh yeah, they use the pipes, |
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| 15:02.42 | Lope | Anyway I'm using Read(choice,,1,,1,7) meaning: variable choice, no files, limit 1 digit, no options, limit 1 attempt, 7 sec timeout. |
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| 15:03.54 | onebree | Lope: I have learned the hard way -- just stop using the voip-info.org site. It became popular because Asterisk did not have as well of a wiki as they do now. |
| 15:04.01 | [TK]D-Fender | voi-info is decreptic crap ... |
| 15:04.13 | [TK]D-Fender | Always read the isntructions from CLI <- |
| 15:04.30 | [TK]D-Fender | "core show application read" |
| 15:04.38 | Lope | TKD ok |
| 15:05.52 | keithf | have you guys ever dealt with sieze failed when trying to make a 10 or 4 digit outbound call |
| 15:07.13 | Lope | Where can I get descriptions of https://wiki.asterisk.org/wiki/display/AST/Application_Queue QUEUESTATUS Basically I want to detect that something went wrong and they were unable to speak to a consultant? |
| 15:08.19 | [TK]D-Fender | keithf: No. Just show us your failed call |
| 15:08.47 | [TK]D-Fender | Lope: What could go "wrong"? |
| 15:10.10 | Lope | well I don't know what JOINEMPTY but it doesn't sound good. Leave empty might be good or bad, I don't know what it means. They left the queue because it was empty of consultants, and nobody could help them. Or someone helped them and the queue is empty because there are no more customers waiting? |
| 15:10.30 | Lope | JOINUNAVAIL: no idea, but sounds bad. |
| 15:10.40 | Lope | LEAVEUNAVAIL: ^ |
| 15:10.43 | keithf | [TK]D-Fender: When greping for the extension he says hes getting seize failed but im not seeing his attempt hit the pbx at all |
| 15:10.59 | keithf | i see inbound calls to his extension |
| 15:11.07 | [TK]D-Fender | "core show application queue" <- read what it spits out |
| 15:11.17 | Lope | CONTINUE: Sounds good. |
| 15:11.18 | Lope | ok |
| 15:11.47 | [TK]D-Fender | keithf: Show us the call |
| 15:11.57 | keithf | outbound ?. |
| 15:12.11 | keithf | wish i could i dont see it when he tries i think its some kind of network issue |
| 15:12.18 | onebree | lOPE: https://github.com/asterisk/asterisk/blob/c944263e3688ab257aaf98980190f359b61d31f7/configs/samples/queues.conf.sample#L441 |
| 15:12.24 | keithf | i can see when i call him but when he dials my extension there is nothing generated |
| 15:12.25 | keithf | at all |
| 15:12.47 | ipn | when you call does his phone ring? |
| 15:12.54 | keithf | yes |
| 15:13.00 | keithf | his phone rings he can receive inbound calls |
| 15:13.07 | keithf | but soon as he tries to dial either 4 or 10 digit |
| 15:13.08 | keithf | nothing |
| 15:13.12 | keithf | just gets seize failed |
| 15:13.25 | keithf | and i dont see any attempt generated on the pbx |
| 15:13.26 | [TK]D-Fender | Are we going to get some confirmation about what is being used? |
| 15:13.40 | keithf | can you clarify ? |
| 15:13.47 | [TK]D-Fender | You also are telling us what you're looking at. |
| 15:13.52 | keithf | im still new trying to figure this all out so i apologize in advance |
| 15:13.59 | keithf | im in the asterisk cli |
| 15:14.09 | keithf | i did the following asterisk -vvvvvr |grep 4027 |
| 15:14.21 | [TK]D-Fender | Yes, and there are a base of 5 completely different kinds fo debug. |
| 15:14.26 | keithf | and i can see when he gets a queue call and i can see his direct ext calls |
| 15:14.34 | [TK]D-Fender | verbose assumes that call is getting ACCEPTED in the first place |
| 15:14.36 | keithf | ahh ok |
| 15:14.39 | [TK]D-Fender | Doesn't mean the attempt doesn't arrive |
| 15:14.52 | [TK]D-Fender | What kind of PHONE are you even talking about? |
| 15:14.53 | keithf | maybe thats why im not seeing the attempt im looking at the wrong place |
| 15:15.00 | keithf | aastra 6757i |
| 15:15.10 | [TK]D-Fender | Asterisk speaks a TON of completely different techs & protocols |
| 15:15.20 | [TK]D-Fender | What ver of *? |
| 15:15.30 | keithf | 11.19 |
| 15:15.35 | [TK]D-Fender | "sip set debug on" <- |
| 15:15.47 | [TK]D-Fender | If you aren't loking at sip debug ... you aren't really looking |
| 15:15.57 | [TK]D-Fender | Go enable it an look for actual packets coming in |
| 15:16.06 | keithf | that changes the cli |
| 15:16.09 | Lope | onebree: thanks! |
| 15:16.17 | keithf | a lot of extra stuff and this is a VERY busy PBX |
| 15:16.53 | Lope | TKD: core show application queue did not describe the QUEUESTATUS values. |
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| 15:17.20 | Lope | But onebree's link answered my questions. |
| 15:18.29 | [TK]D-Fender | LopeTKD: core show application queue did not describe the QUEUESTATUS values. <- Oh yes it does... |
| 15:18.47 | Lope | TKD: it lists them but does not describe them. |
| 15:19.25 | [TK]D-Fender | Lope: The words are pretty obvious for just about all of them |
| 15:20.16 | Lope | Just about != all :p |
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| 15:21.10 | [TK]D-Fender | TIMEOUT <- no answer and your limit came up. FULL <- Too many people already in the queue. JOINEMPTY <- No callable members. LEAVEEMPTY <- Last member left. JOINUNAVAIL <- no FREE agents. LEAVEUNAVAIL <- last Zfree one just left CONTINUE <- only one that isn't just about 100% obvious |
| 15:21.17 | [TK]D-Fender | But if you want to nit-pick... |
| 15:21.27 | [TK]D-Fender | ~pb |
| 15:21.31 | 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. |
| 15:21.32 | [TK]D-Fender | keithf: ^^^ |
| 15:22.37 | Lope | TKD thanks, will save those. I'm new to queues etc. |
| 15:25.37 | keithf | http://pastebin.com/X2RLRKqr |
| 15:26.02 | Lope | Previously, before I used a queue. I was using MixMonitor to setup the call recording, and I was using StopMixMonitor() after Dial() finished. Now that I'll be using a queue with MixMonitor enabled for the queue I guess the queue will call StopMixMonitor on it's own when the queue exits? |
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| 15:26.27 | Lope | when the caller exits the queue I mean. |
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| 15:30.11 | [TK]D-Fender | You should really go read the queue config sample. |
| 15:30.22 | [TK]D-Fender | And you shouldn't have to call for monitor to stop |
| 15:30.27 | keithf | hope that was enough info to get some kind of idea whats going on |
| 15:30.38 | [TK]D-Fender | Typcailly you only want to record a n actual conversation... which means a BRIDGED call. |
| 15:30.41 | [TK]D-Fender | And there are options for that |
| 15:30.45 | Lope | do you mean the default queue.conf? |
| 15:31.09 | [TK]D-Fender | And Iunless you are providing special options Dial = the end of a call because if you talk, ending that call normally kills the channel |
| 15:31.15 | [TK]D-Fender | Lope: the SAMPLE. |
| 15:31.22 | [TK]D-Fender | There is no such thing as "default" |
| 15:32.26 | Lope | okay. Well as far as I can tell there's no connectivity difference between me using Dial(SIP/bob) and having a queue with a member => SIP/bob |
| 15:32.59 | Lope | Both would be bridged MixMonitor? |
| 15:33.47 | [TK]D-Fender | "core show application mixmonitor" <- |
| 15:33.58 | [TK]D-Fender | [10:30][TK]D-FenderAnd there are options for that |
| 15:34.48 | stefan27 | After setting outofcall_message_context = messages in sip.conf, then when an (unregistered) sip friend sends a SIP Message (out of dialog) asterisk creates something called "Message/ast_msg_queue" which executes dialplan in the context 'messages'. Can I from that context extract the SOURCE IP of the received sip message? |
| 15:36.14 | Lope | TKD: thanks |
| 15:36.38 | keithf | http://pastebin.com/X2RLRKqr |
| 15:37.06 | Lope | Previously I was using DEVICE_STATE to see if my extension was busy. Now I need to replace that with a check to see if my queue is busy. I googled "asterisk queue state" but didn't find anything. |
| 15:38.18 | Lope | QUEUE_MEMBER_COUNT should do it? |
| 15:39.14 | Lope | Sorry I shouldn't have said busy. |
| 15:39.40 | Lope | Basically not BUSY and not NOT_INUSE.. |
| 15:40.53 | Lope | I've not made my last question clear. NVM it should be ok. |
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| 15:51.30 | mjordan | Lope: there are specific device states for queues that you can query using the same dialplan function |
| 15:52.00 | mjordan | For queues, it should be "Queue:(name)_avail" if at least one member is available in the queue |
| 15:52.59 | mjordan | For the paused status of a queue member in a particular queue, it is "Queue:(queue_name)_pause_(member_name)" |
| 15:53.10 | Lope | mjordan: thanks. Basically I want to use QUEUE_MEMBER_COUNT to determine whether to tell the caller that nobody is online and tell them how to send an email or offer them to leave a voice mail. |
| 15:53.33 | mjordan | Lope: if the Queue is unavailable, that would be the same thing |
| 15:53.54 | mjordan | using the Queue:(name)_avail device state, that is |
| 15:54.08 | mjordan | the Queue:(name) device state just tells you if someone is waiting in the queue or not |
| 15:54.13 | Lope | same => n,GotoIf($["${QUEUE_MEMBER_COUNT(myqueue)} != 0]?enterqueue) |
| 15:54.20 | mjordan | or that :-) |
| 15:55.40 | Lope | Does Avail mean BUSY || NOT_INUSE ? |
| 15:55.43 | [TK]D-Fender | Reconsider your use of a queue |
| 15:55.58 | [TK]D-Fender | beacuse only allowing one... sounds pointless. might as well be a straight dial |
| 15:56.17 | Lope | I intend to add more members though. |
| 15:56.21 | Lope | It's a startup :p |
| 15:56.23 | [TK]D-Fender | you said you would redial if they were busy |
| 15:56.38 | [TK]D-Fender | but leaving the queue ... or deciding not to even go in is completely contradictory |
| 15:58.05 | mjordan | Lope: It's a lot trickier than that. It means that the configuration of the channel driver has told us that the device that maps to that device state (which in turn maps to a Queue member) has told us that they can have another channel. |
| 15:58.21 | Lope | I think there's been a misunderstanding. GotoIf($["${QUEUE_MEMBER_COUNT(myqueue)} != 0]?enterqueue) this allows the caller into the queue if there are any members online. BUSY or NOT_INUSE. If there are no members online, I provide the email address and offer them to leave a voicemail. |
| 15:58.22 | mjordan | Lope: Generally, that means they don't have a channel going to their device. |
| 15:58.32 | mjordan | But you can have configurations where a SIP device can have multiple channels. |
| 15:59.20 | Lope | mjordan: oh, well my SIP devices are all softphones, and can only take 1 channel. |
| 16:03.56 | [TK]D-Fender | GotoIf($["${QUEUE_MEMBER_COUNT(myqueue)} != 0]?enterqueue) <- joinempty=no |
| 16:04.06 | [TK]D-Fender | this was a single SETTING, not something you have to select in dialplan |
| 16:05.15 | Lope | TKD: Even though I'll only need to add more than 1 member to the queue later, I opted for a queue right now for a number of benefits. 1: if I'm busy, the caller can listen to music briefly and I can still take the call if they wait. 2. I can have my PC and smartphone ring at the same time. 3. I can get someone else to answer the phone I don't have to change the config when they start and stop working. If they're offline or don't answer for some reason I can s |
| 16:05.15 | Lope | till answer the call. |
| 16:06.04 | Lope | TKD Oh I see what you're saying. It would be better to rather let the queue give me an error message. |
| 16:06.24 | Lope | Because there might be members in the queue when I send the caller in, but of all the members leave I need to handle the caller the same way. |
| 16:07.44 | Lope | Oh... there was a good reason for checking before sending them into the queue. |
| 16:08.22 | Lope | I play a msg saying "Please wait while we transfer you to sales. All our calls are recorded" |
| 16:08.38 | Lope | I don't want to play that msg if the queue has no members and is going to fail immediately. |
| 16:10.45 | [TK]D-Fender | Not bad idea then |
| 16:13.43 | Lope | ok. Is this a good way to terminate my dialplan after the queue? same => n,ExecIf($[${CHANNEL(QUEUESTATUS)} = "CONTINUE"]?Hangup()) |
| 16:14.16 | Lope | oh NVM I want to play the bye sound. |
| 16:15.00 | Lope | GotoIf($[${CHANNEL(QUEUESTATUS)} = "CONTINUE"]?bye) |
| 16:15.43 | Lope | Thanks for all the help!!! I'm gonna test the queue tomorrow :) |
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| 16:16.50 | [TK]D-Fender | Lope: IIRC Continue is when the caller got ANSWERD and you told it to continue even after they hang up. |
| 16:17.30 | Lope | Okay, yeah I need to tweak the queue options tomorrow. |
| 16:17.38 | Lope | Night night |
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| 18:29.50 | dan_j | Hi. Any idea how I can end up with two CDR entries for one channel? I'm not using forkcdr or anything |
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| 18:41.26 | onebree | I am looking between pjsip.conf and sip.conf, looking to see which sip vars can be reused. But I cannot find the PJSIP versions of the following vars: |
| 18:43.05 | onebree | (creating list) |
| 18:46.04 | onebree | https://gist.github.com/onebree/b780d7c6ce7111079a0d |
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| 18:58.23 | [TK]D-Fender | limitonpeer=yes <- clearly not required and is in the aor |
| 18:58.24 | [TK]D-Fender | videosupport=yes <- didn't I already tell you this probably is meaningless and to just check the codecs? |
| 18:58.26 | [TK]D-Fender | notifyringing=yes <- checked what the normal notify's go out like? |
| 18:58.37 | [TK]D-Fender | allowguest=no <- there is already a sample for unauthed calls |
| 18:59.05 | [TK]D-Fender | host=dynamic <- of course this is in there |
| 18:59.24 | [TK]D-Fender | read the guide and sample configs again |
| 19:00.57 | onebree | [TK]D-Fender: thank you. All I did was copy/paste the config options to a gist. |
| 19:01.18 | onebree | Did not mean to ask about videosupport again. |
| 19:01.35 | onebree | I am reading the sample configs again, seeing where I missed things |
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| 20:14.48 | saint_ | hey all - can someone explain to me why would a 180 RINGING would have a MESSAGE HEADER in it, and another one won't ? Is there something that trigger this part of the SIP message ? |
| 20:35.54 | jameswf | does dahdi support the clone cards? |
| 20:38.05 | [TK]D-Fender | If they are proper clones |
| 20:42.44 | saint_ | never mind, i found my issue. now... i have 2 sip devices calling in. one gets ring back tone, and the other does not. the SIP server generates the ring back tone, right ? The messages look identical in wiresharkk. |
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| 22:36.13 | snadge | i believe i have enough information now to submit a bug for asterisk 12/13 CDR handling |
| 22:36.35 | snadge | i expect it to be closed as "wontfix" or "by design" .. which is why im going to have to put some effort into it, and have it reviewed etc, before submission |
| 22:37.10 | snadge | the wording of the new specification contains an admission that its broken by design |
| 22:37.59 | snadge | but my interpretation of the spec suggests that the behaviour i'm seeing, is incorrect |
| 22:38.04 | snadge | hence bug report |
| 22:39.18 | snadge | even if the problem is discovered and fixed though.. it would mean the fix goes into bleeding edge current version.. oh well |
| 22:40.41 | snadge | i will provide configuration examples, brief but accurate description of the problem.. and refer to the spec for what i believe to be the correct behaviour, and what actually happens instead |
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| 23:02.14 | klow | Hello, looking for help. I have just 2 soft phones register to asterisk 13.7 (pjsip) . They show registered and pjsip show endpoints lists them. When I try to call one to the other the call does not ring and says no answer in the log. Would someone mind taking a look at my pjsip debug log? http://pastebin.com/sXUNPjmH |
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| 23:47.14 | varesa | Hi. I'm having issues with setting up chan_dongle |
| 23:47.47 | varesa | chan_dongle.c:223 opentty: unable to open /dev/ttyUSB2: Permission denied |
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| 23:48.30 | varesa | ttyUSB2 is owned by root:dialout, asterisk is member of dialout. I can even su -c asterisk |
| 23:48.46 | varesa | su -c "screen /dev/ttyUSB2" asterisk |
| 23:48.56 | varesa | Does chan_dongle run as some other user? |
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