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| 03:13.09 | volga629 | Hello Everyone, I am looking for testing video phone with ability to have mini HDMI port any suggestion ? |
| 03:13.42 | volga629 | Grandstream GXV-3175 Multimedia Video Phone found this one, but never used before |
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| 06:32.13 | Naikrovek | so, Raspberry Pis are awesome... |
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| 06:32.25 | Naikrovek | speaks to you now via X-Chat on an rpi |
| 06:33.56 | Naikrovek | I am wondering how Asterisk would run on this. |
| 06:38.13 | Naikrovek | powers down... |
| 06:45.16 | *** join/#asterisk hariom (~hariom@117.230.209.230) |
| 06:46.23 | hariom | I am facing a strange problem. I am using agi for ivr and recording my voice and then playing it back. I see that all prompts play well but the recorded file doesn't play back as there no sound in it. |
| 06:50.37 | hariom | The agi return code is always 0 for recorded file meaning the file not found I think. But I can get to the file with same filepath so filepath is not a problem |
| 06:52.26 | hariom | Recording of any length is creating just a 44bytes of file. |
| 06:53.42 | hariom | If recording format is gsm, file size is 0 bytes |
| 06:54.28 | hariom | I think there is some bug in *. I am using * 1.8.19.1 |
| 07:01.14 | hariom | kaldemar:^ |
| 07:08.12 | hariom | anybody? |
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| 07:36.12 | ectospasm | hariom: probably a bug in your AGI, though I don't understand why you need AGI to do basic call recording... |
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| 08:05.14 | [TK]D-Fender | hariom: You haven't shown us anything... |
| 08:17.38 | ChannelZ | Yeah. Show us your boobs. |
| 08:19.40 | carrar | boob pics? |
| 08:19.54 | [TK]D-Fender | Boobytrap is party boob backwards.... |
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| 08:20.37 | WIMPy | s/backwards/from behind/ |
| 08:20.54 | ChannelZ | Everything's a party from behind |
| 08:33.01 | *** join/#asterisk Itsuki626 (7bd38f69@gateway/web/freenode/ip.123.211.143.105) |
| 08:33.38 | Itsuki626 | Hi. Is it possible to add a custom country to Asterisk? I wish to use a custom indications.conf, but everytime I reload my configs, the country gets reverted to the default indications.conf. |
| 08:35.25 | Wiretap | Itsuki626: stop using freepbx then |
| 08:35.40 | Wiretap | reloading will not change the files |
| 08:35.49 | Wiretap | it'll just barf on them if they're wrong |
| 08:36.46 | ChannelZ | Scared another one off! |
| 08:37.03 | Wiretap | /care |
| 08:46.07 | hariom | [TK]D-Fender: This is the record command i am using. The same command works fine (i.e. stored file gets sound) in my another virtual machine. But not in new VirtualMachine I have. I can play prompts in this new VM |
| 08:46.28 | hariom | "RECORD FILE %s %s %s %d %d %s\n" %(filepath, format, escape_digits, timeout, offset, beep) |
| 08:46.48 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 08:46.51 | [TK]D-Fender | Maybe your networking is messed up and audio isn't making it through at all to record. |
| 08:47.06 | [TK]D-Fender | You should looking at the call to actually prove what's happening... |
| 08:47.28 | hariom | [TK]D-Fender: The new VM also has asterisk installed locally. So I am using the new VM for everything |
| 08:47.59 | Wiretap | maybe you've forgotten to install the record module |
| 08:48.09 | [TK]D-Fender | no |
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| 08:58.40 | hariom | [TK]D-Fender: https://gist.github.com/406aa1ae9b6d214c9869 |
| 09:00.41 | [TK]D-Fender | hariom: I just told you of the high likelihood of it being due to NETWORKING problems and you are not looking at SIP DEBUG or RTP there. |
| 09:00.58 | [TK]D-Fender | hariom: I don't trust that packets are even arriving. |
| 09:01.29 | hariom | [TK]D-Fender: I see. Will try debugging from that angle. |
| 09:01.37 | [TK]D-Fender | hariom: And .... times up. I'm off to bed. Hopefully you'll get back with something more complete that we can examine for the cause for your issue. |
| 09:03.06 | hariom | [TK]D-Fender: Thanks. Sure. Have good sleep. |
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| 09:25.45 | gavimobile | my provider is telling me that if I want the callerid to be more accurate than I need to pay more money. this is what he said "At our current price point, none of our vendors can guarantee CLI to Israel non-Mobile. We can set you up on route that guarantees CLI presentation, but it will be approximately 50% more in cost." |
| 09:26.01 | gavimobile | is this bs? |
| 09:29.37 | ectospasm | sounds like it |
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| 09:38.21 | gavimobile | ectospasm: I hope not |
| 09:39.06 | ectospasm | wait, are you trying to set your CID, and what your customers/whatever see on the far end is not what you're setting? |
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| 11:40.39 | gavimobile | ectospasm: that is correct |
| 11:40.46 | gavimobile | it seems to be intermittant |
| 11:41.02 | ectospasm | that's shady |
| 11:41.06 | gavimobile | land line clid is less stable. they suggested I created a static clid which I did and that's not working either |
| 11:41.10 | gavimobile | this is rapidvox btw |
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| 15:14.15 | jmls | right |
| 15:14.46 | jmls | I know I'm really going to look stupid, but would rather look stupid for 10 minutes than struggle with this for another hour ... |
| 15:14.51 | jmls | I can't park calls. |
| 15:15.25 | jmls | features.conf is set up with parking extension 700, parked call extensions 701-720 |
| 15:15.34 | jmls | Park call is #72 |
| 15:15.50 | jmls | dial_external context has include => parkedcalls |
| 15:16.25 | jmls | "dialplan show 700@dial_external" gives me |
| 15:16.28 | jmls | [ Included context 'parkedcalls' created by 'features' ] |
| 15:16.31 | jmls | '700' => 1. Park() [features] |
| 15:16.49 | jmls | however, when I try to park, I get the error message |
| 15:17.10 | jmls | WARNING[1932][C-00000001]: features.c:1483 park_space_reserve: Parking lot not available to park |
| 15:18.00 | jmls | anyone care to point out where I am being daft ? |
| 15:18.04 | jmls | ta ;) |
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| 15:37.50 | BeeBuu | hello,all. |
| 15:38.01 | BeeBuu | is there anyone try sipml5 in asterisk 11? |
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| 15:38.55 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:45.30 | jmls | err. ok. figured it out. stupid dialplan was setting a parkinglot value that's not in features.conf |
| 15:45.39 | jmls | blames the dialplan |
| 15:45.44 | jmls | so, I can park. |
| 15:46.00 | jmls | can't pick up the parked call |
| 15:46.28 | jmls | so, I'm told 701 when parking. Dial 701, get an ooo. |
| 15:46.42 | jmls | must be the polycom dialplan, right ? ... |
| 15:47.29 | *** join/#asterisk appleboy (~appleboy@about/cooking/nakedchef/apple/tarts) |
| 15:57.06 | jmls | nope. asterisk dialplan |
| 15:57.11 | jmls | bad dialplan. bad! |
| 15:57.16 | jmls | ok, next. |
| 15:58.28 | *** join/#asterisk b1rkh0ff (~b1rkh0ff@178.77.10.139) |
| 15:58.53 | jmls | OTR. if a dialplan has an option in the asterisk db to start recording a call, how do you turn OTR off ? Or is it not a 1-0 thing ? From what I can see, *1 is a toggle. does that mean of recording is already happening , would a *1 turn it off (the opposite of what the user would expect) ? |
| 16:00.02 | ghost75 | is it not possible to show if a phone is busy? |
| 16:06.09 | ghost75 | i get only idle or unavailable but not busy state |
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| 16:41.33 | ghost75 | gnaaa the hints are showing only idle or unavailbe, why this was changed in newer versions? |
| 16:41.55 | jmls | busy _should_ work, surely ? |
| 16:42.07 | ghost75 | no |
| 16:43.51 | ghost75 | i cant use device_state because i dont which phone uses which channels |
| 16:45.33 | ghost75 | or is it because i did hint like this: exten => _1X,hint,SIP/${EXTEN} |
| 16:48.34 | ghost75 | nope, isnt |
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| 17:53.51 | kaldemar | ghost75: newer than what? |
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| 17:54.55 | kaldemar | ghost75: and why can't you use DEVICE_STATE really? |
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| 18:12.58 | ghost75 | kaldemar: newer than no clue :) |
| 18:13.56 | kaldemar | ghost75: you should know your own version. |
| 18:14.03 | ghost75 | 1.8 |
| 18:14.26 | kaldemar | that's a branch, not a version. anyway, hints do provide more than idle/unavailable. |
| 18:15.07 | ghost75 | 1.8.13.1~dfsg-1~bpo60+1 |
| 18:15.46 | kaldemar | and your issue with DEVICE_STATE is what? |
| 18:17.11 | ghost75 | there is no channel when the phone is not used |
| 18:17.41 | ghost75 | want to display state of all phones via web |
| 18:17.57 | kaldemar | it uses a DEVICE, not a channel. |
| 18:18.50 | ghost75 | i think in ami a channel has to be provided |
| 18:19.10 | kaldemar | there is no such thing as ami channel. show what you're trying to do. |
| 18:20.54 | [TK]D-Fender | ghost75want to display state of all phones via web <--- |
| 18:21.13 | ghost75 | i think it goes over getvar |
| 18:21.16 | [TK]D-Fender | Hints WILL show who's on a call when you set them up |
| 18:21.16 | kaldemar | which also has little to do with hints. |
| 18:21.31 | [TK]D-Fender | ghost75: And you don't need DEVICE_STATE for this |
| 18:21.51 | [TK]D-Fender | Hints are a perfectly valid way of getting that info |
| 18:21.55 | ghost75 | no hints are only showing idle and not available, nothing else |
| 18:22.06 | [TK]D-Fender | DEVICE_STATE as a dialplan function would not make sense. |
| 18:22.20 | [TK]D-Fender | [13:21]ghost75no hints are only showing idle and not available, nothing else <- Hints also show ringing, hold, etc |
| 18:22.29 | ghost75 | not for me |
| 18:22.42 | [TK]D-Fender | ghost75: Show us |
| 18:23.20 | ghost75 | exten => _1X,hint,SIP/${EXTEN} |
| 18:23.31 | [TK]D-Fender | ghost75: .... |
| 18:23.47 | [TK]D-Fender | ghost75: Show us the DUMP and prrof of what's actually happening that does NOT agree with the actual state |
| 18:24.17 | ghost75 | how i should put state in dump? |
| 18:24.37 | [TK]D-Fender | If you can't show me their value then how can you tell me you can only see 2 values? |
| 18:24.46 | [TK]D-Fender | Show us the state of all of them |
| 18:25.01 | *** join/#asterisk gg608f (~Adium@187.207.6.250) |
| 18:25.07 | ghost75 | i look realtime at cli during call |
| 18:25.33 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 18:25.56 | [TK]D-Fender | ghost75: Show us <- |
| 18:27.57 | ghost75 | https://www.sugarsync.com/piv/D234635_65941903_648551 <- never changes |
| 18:28.36 | [TK]D-Fender | ghost75: Show us. |
| 18:28.43 | ghost75 | what ????? |
| 18:28.54 | [TK]D-Fender | ghost75: Show us that it never changes.... |
| 18:29.02 | ghost75 | how ??????? |
| 18:29.09 | [TK]D-Fender | ghost75: This is a simple request and it doesn't seem to be registering with you... |
| 18:29.25 | ghost75 | why you dont believe me? |
| 18:29.29 | [TK]D-Fender | [TK]D-Fenderghost75: Show us the DUMP and prrof of what's actually happening that does NOT agree with the actual state |
| 18:29.39 | [TK]D-Fender | Because I don't see the backup |
| 18:29.58 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1005) |
| 18:30.07 | dpilon | get on the phone..dial something...get another screenshot |
| 18:30.08 | kaldemar | ghost75: if people answering questions always believed what they were told, not many issues would get resolved. |
| 18:30.26 | ghost75 | dpilon: he doesnt take screenshot as proof |
| 18:30.40 | dpilon | well..there was only one |
| 18:30.52 | dpilon | were you on a call at the time of the screeenshot |
| 18:31.15 | kaldemar | a single screenshot is a proof that their states are what they seem at a discrete time. a single screenshot proofs nothing at all about changing. |
| 18:32.33 | kaldemar | if you showed another CLI output that shows a call to a hinted extension and that same output again, it would be something more. |
| 18:33.34 | ghost75 | https://www.sugarsync.com/piv/D234635_65941903_648551 |
| 18:33.36 | [TK]D-Fender | "Prrof that it does not agree". |
| 18:33.46 | [TK]D-Fender | I don't see a CALL happen that should cause it to change. |
| 18:33.56 | [TK]D-Fender | Someone does not seem to understand what constitutes prrof. |
| 18:34.15 | [TK]D-Fender | proof* |
| 18:34.16 | [TK]D-Fender | gah |
| 18:34.38 | kaldemar | so, intern vs. throttleon. |
| 18:35.47 | [TK]D-Fender | ghost75: pastebin your SIP config |
| 18:36.39 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1005) |
| 18:36.56 | ghost75 | so different context is then problem |
| 18:37.09 | [TK]D-Fender | No |
| 18:37.12 | [TK]D-Fender | ghost75: pastebin your SIP config |
| 18:37.16 | *** join/#asterisk b1rkh0ff (~b1rkh0ff@178.77.6.46) |
| 18:37.18 | ghost75 | see what kaldemar wrote |
| 18:37.25 | [TK]D-Fender | I did. I disagree |
| 18:37.50 | ghost75 | u mean my dialplan? |
| 18:38.04 | [TK]D-Fender | The hint should register the state regardless of watchers (assuming it's static) and has nothing to do with a call being processed in a specific context |
| 18:38.10 | [TK]D-Fender | ghost75: pastebin your SIP config <-- |
| 18:38.12 | [TK]D-Fender | SIP CONFIG. |
| 18:38.13 | [TK]D-Fender | SIP.CONF |
| 18:39.10 | ghost75 | to see context in sip.conf? |
| 18:39.25 | [TK]D-Fender | clearly. |
| 18:39.42 | [TK]D-Fender | and that'd be "device entry", not "context". |
| 18:40.16 | ghost75 | [10] |
| 18:40.20 | ghost75 | context=phones998780 |
| 18:40.54 | [TK]D-Fender | ghost75: PASTEBIN |
| 18:40.59 | [TK]D-Fender | the entire damn peer |
| 18:41.11 | [TK]D-Fender | and you showed screenshots from CLI you could have pastebinned as plain te3xt as well |
| 18:41.17 | [TK]D-Fender | I did NOT say "context" |
| 18:41.27 | [TK]D-Fender | pastebin the entire entry for that device |
| 18:42.08 | dpilon | pastebin....not here :) |
| 18:43.38 | ghost75 | http://pastebin.com/139Ei74w |
| 18:44.17 | [TK]D-Fender | callcounter=yes <- |
| 18:44.20 | [TK]D-Fender | add it. |
| 18:44.22 | [TK]D-Fender | reload. |
| 18:44.26 | [TK]D-Fender | retest |
| 18:44.34 | ghost75 | oh -.- |
| 18:44.55 | ghost75 | didnt know that one |
| 18:46.20 | dpilon | i like this from wiki: IMPORTANT: In Asterisk 1.6.2.X in sip.conf you need callcounter=yes for presence to work |
| 18:47.24 | [TK]D-Fender | Yup |
| 18:49.12 | ghost75 | thx, works |
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| 18:50.20 | ghost75 | except that exten => _1X,hint,SIP/${EXTEN} was not register anymore but exten => 10,hint,SIP/10 was working |
| 18:50.59 | [TK]D-Fender | you created a static hint.... and your pattern overlaps anyway |
| 18:51.17 | [TK]D-Fender | the pattern generates static hints once something SUBSCRIBES to one. |
| 18:51.25 | [TK]D-Fender | Which of course you don't even have any of |
| 18:51.45 | ghost75 | i added it after it didnt work |
| 18:51.57 | [TK]D-Fender | You showed no subscriptions of any kind |
| 18:52.06 | [TK]D-Fender | it does nothing until something subscribes to one anyway |
| 18:52.19 | [TK]D-Fender | You should not expect to see anything for it |
| 18:53.48 | ghost75 | k when i poll it over ami then it registers |
| 18:54.07 | [TK]D-Fender | Polling over AMI won't impact that pattern |
| 18:54.09 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 18:54.15 | [TK]D-Fender | your fixed ones should do their jobs though |
| 18:54.44 | ghost75 | i have only that dynamic one now |
| 18:54.48 | ghost75 | works |
| 18:55.42 | ghost75 | == Manager 'manager' logged on from 127.0.0.1 |
| 18:55.42 | ghost75 | <PROTECTED> |
| 18:55.42 | ghost75 | and so on .... |
| 18:56.05 | [TK]D-Fender | And what are you doing in there? |
| 18:56.57 | ghost75 | getting sip_peer and exten_state |
| 18:57.21 | *** join/#asterisk acedia (~rage@unaffiliated/ffs) |
| 18:59.40 | [TK]D-Fender | exten_state should have nothing to do with device_state |
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| 19:22.00 | ghost75 | i bought a used 7941g for 25$ (no psu) |
| 19:22.43 | ghost75 | in the next auction its was even only 13$ |
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| 19:50.57 | doctorray | Can a Conference (either new or old style) be created in which the admin user can hear everyone, but the non-admin users can't hear each other? |
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| 19:51.08 | doctorray | or something like that |
| 19:55.13 | doctorray | selectively routing audio based off the user's choice |
| 19:56.17 | doctorray | think a security application, a central monitoring station to listen to multiple areas and talk to some/all, but not enabling the areas to talk to each other. |
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| 20:08.18 | [TK]D-Fender | doctorray: MeetMe and ConfBridge both have these options. Read the apps's instructions and config files. it's all in there. |
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| 20:55.46 | volga629 | Hello Everyone, trying troubleshoot [2013-01-13 15:44:02] WARNING[31598]: res_rtp_asterisk.c:2143 ast_rtp_read: RTP Read too short |
| 20:55.48 | volga629 | [2013-01-13 15:44:59] WARNING[31598]: res_rtp_asterisk.c:2143 ast_rtp_read: RTP Read too short |
| 20:56.12 | volga629 | looking for tips and suggestion |
| 20:56.27 | volga629 | How to troubleshoot it right |
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| 23:18.51 | doctorray | [TK]D-Fender: Thanks. I did some preliminary reading in the wiki.asterisk.org documentation before I asked but it wasn't 100% apparent. I'll set up a test environment and try a few things out before I ask more about it here. |
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