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| 06:32.19 | AjmiNes | Hey guys |
| 06:32.28 | AjmiNes | i have a problem with my Asterisk server |
| 06:33.01 | AjmiNes | and they told me that here i will found the skilliest people on asterisk |
| 06:33.46 | AjmiNes | Is there any body here to explain to me how it work here ? |
| 07:04.51 | raspberrypifan | what u need boy |
| 07:20.03 | AjmiNes | There is a lot of calldrop in asterusj |
| 07:20.10 | AjmiNes | asterisk* |
| 07:20.34 | AjmiNes | and i wanna found from where is the problem |
| 07:36.02 | TazzNZ | AjmiNes: we need more information |
| 07:38.57 | AjmiNes | what kind of information you need |
| 07:39.35 | TazzNZ | when does the cals drop? |
| 07:39.46 | TazzNZ | what calls are dropping? |
| 07:39.55 | TazzNZ | what does the netowrk look like? |
| 07:42.29 | AjmiNes | SIP <> INTERNET <ASTERISK> <SIP> <GSM> |
| 07:43.12 | AjmiNes | I record all call and i have noticed that the incoming voice is always lost |
| 07:43.29 | AjmiNes | the call drop from 5 to 30s |
| 07:44.07 | AjmiNes | when the called numer [GSM] don't hear the caller after few second the call drops |
| 07:44.53 | TazzNZ | sounds like an rtp issue |
| 07:45.52 | TazzNZ | is there anything on the cli saying no response from our critical packet |
| 07:46.05 | AjmiNes | how can i check if it's rtp issue |
| 07:48.12 | AjmiNes | the unique warning in console log is WARNING[29006] app_dial.c: Unable to write frametype: 2 |
| 07:48.46 | TazzNZ | no - not that one - what does the firwalling lok like? |
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| 07:59.00 | AjmiNes | There is no firewall i disabled it |
| 07:59.03 | AjmiNes | by the way |
| 07:59.22 | AjmiNes | yesterday i havent that problem |
| 07:59.44 | AjmiNes | i had sometimes like 6 simultanious call |
| 07:59.48 | AjmiNes | with any problem |
| 08:00.02 | AjmiNes | but when my provider hitted a litle bit the traffic |
| 08:00.11 | AjmiNes | i got this problem |
| 08:00.17 | AjmiNes | even with 2 calls now |
| 08:00.37 | TazzNZ | perhaps the problem is with them? have you asked them to look |
| 08:00.39 | AjmiNes | there is a voice lost or call drop |
| 08:00.50 | AjmiNes | It's sunday |
| 08:00.53 | AjmiNes | so they don't work |
| 08:00.56 | TazzNZ | is it all gsm calls? |
| 08:01.14 | AjmiNes | That i'm handling |
| 08:01.22 | AjmiNes | or call that i receive ? |
| 08:01.50 | TazzNZ | the one that is dropping? |
| 08:04.42 | AjmiNes | no call that i receive from my provider |
| 08:05.02 | AjmiNes | some of them from gsm |
| 08:05.11 | AjmiNes | and others from fixed line |
| 08:05.25 | TazzNZ | so are all calls having this issue? |
| 08:05.39 | AjmiNes | some no |
| 08:05.55 | AjmiNes | there is call with more than 7 minutes |
| 08:06.04 | AjmiNes | but the majorty of call drop |
| 08:06.08 | AjmiNes | with less then 40s |
| 08:06.49 | AjmiNes | is there a way to see if the problem is from my side |
| 08:06.55 | AjmiNes | or from the operator side |
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| 08:07.43 | TazzNZ | the best advise i can say is that you should have verbose up to 9 or so and see if the cli shows anything |
| 08:08.28 | TazzNZ | you could also try rtp debug but that is very busy etc |
| 08:11.45 | AjmiNes | Ok |
| 08:11.51 | AjmiNes | Verbose is 9 now |
| 08:13.21 | AjmiNes | so what i should look for ? |
| 08:13.37 | TazzNZ | any errors ? |
| 08:14.20 | AjmiNes | no |
| 08:21.36 | AjmiNes | it make me crazy :/ |
| 08:21.45 | AjmiNes | i can't found from where is the problem |
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| 08:22.50 | AjmiNes | How can i drop rtp |
| 08:23.11 | AjmiNes | to see if the rtp is droped from my side |
| 08:23.19 | AjmiNes | or from the provider side |
| 08:23.30 | AjmiNes | wich rtp is lost first |
| 08:23.48 | AjmiNes | which* |
| 08:32.35 | AjmiNes | Could someone tell me how to see who hangup the call |
| 08:32.48 | AjmiNes | using sip |
| 08:33.20 | AjmiNes | how i can see if the caller hangup or the called number hangup |
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| 09:38.28 | AjmiNes | could somebody help me |
| 09:38.38 | AjmiNes | sorry i had a connexion problem |
| 09:38.53 | AjmiNes | so i dont know if someone answered to my question |
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| 09:52.37 | ChannelZ | Well if you look at sip debug you'll see if you received a BYE or sent one |
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| 10:25.08 | yottanami | I want to connect my gateway and Astersik machine but I have authentication error http://dpaste.com/01W463A/ |
| 10:25.08 | yottanami | I am sure about passwords that sets in gateway panel and Astersik |
| 10:42.39 | ChannelZ | You probably have the username set to newrock |
| 10:44.40 | ChannelZ | but you're trying to call as 8000 |
| 10:50.51 | ChannelZ | That is, you're using a different authentication username. |
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| 11:04.17 | yottanami | ChannelZ: I changed /etc/asterisk/sip.conf like this http://dpaste.com/0XDAZMC/ but I have this error : http://dpaste.com/2BVPQTX/ |
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| 11:25.26 | yottanami | There was a match_auth_username=yes option is Elastix where can I set it in Plain Asterisk ? |
| 11:50.26 | Gugge | in sip.conf |
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| 15:36.30 | dorphalsig | Hello |
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| 18:02.17 | *** join/#asterisk TodWulff (~TodWulff@unaffiliated/todwulff) |
| 18:08.00 | TodWulff | Good day. Just installed Asterisk 11.8.1 on an imbedded device. No packages exist for it for a GUI'd up configurator or a UI head like FreePBX. I am not a dev and porting one of those packages is not gonna happen. :) Given that there are a couple of different mechanisims that give support for remote administration of Asterisk (AMI and JAMI? come to mind, based on my reading), I find myself asking if there are stand alone tools that can touc |
| 18:08.29 | TodWulff | And push it to a server. Does anyone have any ideas if such tools exist? |
| 18:09.01 | TodWulff | lols at imbedded vs embedded |
| 18:10.14 | cusco | hi folks |
| 18:11.18 | cusco | TodWulff: I only know about configuring files manually |
| 18:11.41 | cusco | I guess, as long as it is configuration, you could a gui elsewhere such as freepbx, and send configured files to your device .. ? |
| 18:12.10 | cusco | so... I testing cdr with odbc (in a ms sql server) |
| 18:12.27 | cusco | err not cdr, I mean queue_log via extconfig |
| 18:12.36 | cusco | and... column time is type datetime |
| 18:12.54 | cusco | however.. Conversion failed when converting date and/or time from character string. (94) |
| 18:12.58 | cusco | any hints? |
| 18:13.15 | cusco | looked at the debug, the field goes like: 2014-06-01 19:07:01.924733 |
| 18:15.03 | TodWulff | hi cusco. yeah, I am struggling through that learning curve. /me isn't linux saavy so that is painful. interesting concept you propose. will have to consider it more. I did look at doing so with freePBX on a centos box, but the task is daunting as FreePBX is so heavily integrated with it's asterisk framework that it presents impossible for one not benefitting from messing with the package for a long while. |
| 18:15.06 | cusco | nvm |
| 18:15.36 | cusco | well I've never used freepbx |
| 18:15.48 | cusco | asterisk as a built in web gui |
| 18:16.02 | cusco | may not be as friendly as freepbx |
| 18:16.02 | TodWulff | i read that it was deprecated and not supported any longer. |
| 18:16.18 | cusco | hmm that might be.. not sure.. I've tested it not long ago |
| 18:16.22 | cusco | but could be old |
| 18:17.03 | TodWulff | https://wiki.asterisk.org/wiki/display/AST/Asterisk+GUI <-- is that what you were talking about? |
| 18:18.36 | TodWulff | that package does exist and I've installed it but it is failing to function (likely lacking a proper config). The http daemon is running and I can touch the server, but the config pages are not being offered - again as I am ignorant and haven't the knowledge to know what I need to do to get it to work. screeny forthcoming. |
| 18:20.01 | TodWulff | http://i.imgur.com/NISoCtS.png |
| 18:21.57 | TodWulff | the host is an asus router running merlin's asuswrt (not that it matters here, just an fyi). |
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| 18:33.08 | gusto | i just discovered a problem that my asterisk ran for a few days at 100% cpu load because of dnsmgr |
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| 18:57.11 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 18:59.48 | dan_j | Hi. I'm trying to workaround recordings and call transfers. Am I correct in saying that if a call comes in and answered (lets call that channel A), then it Dials a local channel (lets say B). Then the call is transferred to channel C. Am I correct that Channel A will still be connected, while Channel B will no longer exist? |
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| 19:27.44 | cusco | hi... so i'm testing asterisk 11 |
| 19:27.50 | cusco | and trying hints as state_interface |
| 19:28.01 | cusco | its hint:ext@context right? |
| 19:29.11 | cusco | queue show <queue> shows members: |
| 19:29.12 | cusco | <PROTECTED> |
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| 19:29.32 | dan_j | Hi. Is it possible to have a macro play different sounds to each side of the channel? |
| 19:29.33 | cusco | its invalid... |
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| 19:55.09 | TodWulff | http://i.imgur.com/69zRV6R.png <-- progress, in babysteps albiet, but still progress... |
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| 20:02.30 | ChannelZ | dislikes asterisk GUIs |
| 20:04.55 | cusco | hi ChannelZ ! any hints regarding hints ? :p |
| 20:07.59 | ChannelZ | What are you trying to do? |
| 20:08.22 | cusco | using realtime queue members, set column state_interface to hint:ext@context ? |
| 20:08.31 | cusco | so app_queue knows about hinted member' |
| 20:08.32 | cusco | ? |
| 20:08.51 | cusco | however, queue show <queue> shows |
| 20:08.55 | cusco | 150 (Local/150@agents/n from HINT:150@agents) with penalty 10 (ringinuse disabled) (realtime) (Invalid) has taken no calls yet |
| 20:10.20 | ChannelZ | I have no idea, I'm not a realtime user and no idea how hints apply to queue agents. |
| 20:10.37 | cusco | :/ |
| 20:10.48 | *** part/#asterisk smkelly (~smkelly@mykonos.smkelly.org) |
| 20:15.56 | dan_j | leifmadsen: are you around? is there any documentation around on call recording and transfer with asterisk 12? |
| 20:17.24 | gusto | isnt it possible that the problem i encountered on netbsd, that my asterisk eats up 100% cpu time, when dnsmgr is on, may have something to do with time_t? |
| 20:17.46 | gusto | because it came up to me that this netbsd-6 is already using 64bit time_t |
| 20:18.20 | gusto | so it may not be a problem with timing, because it would be a KVM VirtIO, but because of the time format |
| 20:18.54 | ChannelZ | I assume you actually need dnsmgr? |
| 20:19.09 | gusto | well, i disabled it for now, because i noticed what happened |
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| 20:19.28 | gusto | but maybe there will be problems with one device which is set not to register |
| 20:19.58 | gusto | i can fix that in a week, but it makes me angry, because asterisk just does not break! |
| 20:20.43 | ChannelZ | You really only need it if you have peers with dynamic IPs but need to hardcode a hostname for it into your config. |
| 20:21.02 | gusto | yes |
| 20:21.08 | gusto | that was what i was doing |
| 20:21.27 | gusto | however, i have several asterisk installations, on MIPS, ARM and bullshit like that and every one of them works |
| 20:22.02 | gusto | only this netbsd one makes trouble, and a good explanation would be time_t. ... one would need to test it on netbsd-5 |
| 20:22.18 | gusto | i ll have to reboot for that to kick in virtualbox and test that possibility |
| 20:25.39 | gusto | i am going to do that fasdt |
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| 20:29.47 | gusto | so |
| 20:32.13 | gusto | lets see |
| 20:43.48 | gusto | eh, same problem on netbsd-5 |
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| 21:00.17 | gusto | maybe a better solution would be to kick out dnsmgr and doing lookups when they come around, so when someone makes a call or there is an incomming call and asterisk would look who the peer is |
| 21:00.30 | gusto | and caching that according to TTL |
| 21:07.16 | leifmadsen | dan_j: there might be some documentation on the wiki. I haven't really played with Asterisk 12. The way I plan to do call recording in the future is not to let Asterisk control it at all (at least in terms of the actual recording). I'll just be triggering things via AMI that my API interface will listen for, and then start/stop recording |
| 21:07.25 | leifmadsen | this is not really well documented |
| 21:07.37 | leifmadsen | vanishes again |
| 21:08.58 | dan_j | leifmadsen: so you'll still actually record using asterisk? Just the control will be external. |
| 21:12.31 | dan_j | Is there any reason why that couldnt work with asterisk 11? |
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| 21:35.23 | dan_j | When using heartbeat, does it failover when asterisk fails, or just the server itself? |
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