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| 05:08.16 | drmessano | Yawn |
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| 05:34.37 | adeel | is there an easy way to track the maximum number of simultaneous channels utilized by a peer, that is under their max limit? As an example, say user A is set to a limit of 30 concurrent channels, and B is limited to 15; A & B are interested to know how many of those respective channels have they actually used |
| 05:34.47 | adeel | i can't think of an approach that isn't polling based |
| 05:35.48 | WIMPy | Scan the logs an do some maths. |
| 05:36.12 | adeel | well, that's probably the least optimal way of doing it |
| 05:37.14 | WIMPy | But it will be exact. |
| 05:37.54 | adeel | so are many other approaches |
| 05:38.14 | adeel | and that assumes many other things which must hold true....1 of which, asterisk doesn't crash |
| 05:38.17 | WIMPy | Polling wouldn't |
| 05:38.38 | adeel | i know polling won't, which is why i'm not trying to use polling |
| 05:38.45 | WIMPy | You find that information in the log as well. |
| 05:41.04 | adeel | hmm....i think i have a better approach |
| 05:41.43 | WIMPy | Off course the standard answer applies here as well: Listen on AMI. |
| 05:41.53 | WIMPy | I should put that on a function key. |
| 05:43.26 | adeel | listening to AMI doesn't remotely solve the problem, but sure |
| 05:43.43 | WIMPy | You still have to count, yes. |
| 05:44.22 | adeel | you actually have to keep multiple counts |
| 05:45.28 | WIMPy | One counter and one high water mark per peer. |
| 05:46.18 | Akari | adeel: You tried GROUP() and GROUP_COUNT()? |
| 05:46.38 | adeel | Akari: at the SQL level? or in asterisk? |
| 05:46.52 | Akari | Asterisk |
| 05:47.26 | adeel | Akari: i've toyed with that approach |
| 05:47.55 | adeel | Akari: that just gets me another way to count |
| 05:49.21 | Akari | Compare to a value in astdb and if GROUP_COUNT() > astdb, replace astdb with GROUP_COUNT() |
| 05:50.00 | adeel | Akari: yeah, that's the rough approach per peer i'm going to do via SQL |
| 05:50.44 | Akari | Good luck |
| 05:51.45 | adeel | only downside with group_count()/group() is that its the instantaneous value at the time group_count() was issued....at any decent calls per second, that approach becomes decently taxing |
| 05:53.34 | WIMPy | You have to add it to all extensions and to h. |
| 05:53.48 | Akari | What about SNMP |
| 05:53.58 | WIMPy | Buth then it would also work really well. |
| 05:54.08 | WIMPy | Polling is never exact. |
| 05:55.21 | Akari | Maybe some kind of CDR reader |
| 05:56.03 | WIMPy | That's scanning logs isn't it? |
| 05:56.05 | Akari | I feel like it would be rather taxing to create a bespoke solution for querying MySQL for that information because you just get the start and end times. |
| 05:56.35 | adeel | i've done the CDR/logging approach before it...it's brutal, and gets overly complex when you've got calls that span multiple days |
| 05:57.01 | WIMPy | Whooot? |
| 05:57.28 | Akari | Like something that's prepackaged I mean. |
| 05:57.47 | adeel | imho, you need a SQL stored procedure that is called on every call setup/teardown and within that procedure, it's keeping a running count of current calls, as well as the max value it's seen over the desired interval |
| 06:00.23 | Akari | I don't think the call record is added to MySQL until the call is complete. |
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| 06:00.45 | adeel | Akari: i'm not planning on using the CDR for this count |
| 06:00.48 | Akari | And even then if someone transfers a call or conferences or something it may mess with your results. |
| 06:01.16 | Akari | It might work with CEL |
| 06:04.57 | Akari | WIMPy: Was that what you were thinking with using AMI |
| 06:05.15 | WIMPy | What? CEL? |
| 06:06.02 | Akari | Yeah enabling cel_manager |
| 06:06.18 | WIMPy | AMI tells you whenever a channel is created or destroyed. So you just have to count those events. |
| 06:06.27 | WIMPy | So no need for CEL. |
| 06:06.31 | Akari | Ah |
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| 06:28.38 | adeel | when issuing an 'core stop gracefully' ...what does asterisk respond to new INVITES with? a 500 error? |
| 06:29.06 | adeel | and what about for in-dialog calls? will it process any re-INVITES, etc? |
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| 10:35.31 | riess82 | hi all. i have an spa232d gateway connected to asterisknow. in the gateway i have set up dialplan S0(:112) which correctly rings extension 112. is there a way to dial a certain extension through the gateway? (i am new to VOIP in general and have no idea about dialplans) |
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| 11:38.12 | wasanzy | Hi |
| 11:39.15 | wasanzy | am getting this error: SIP/+233271520077@outgoing (ringinuse disabled) (dynamic) (Invalid) has taken no calls yet |
| 11:39.15 | wasanzy | <PROTECTED> |
| 11:39.39 | wasanzy | when I use the queue application to add members to the queue |
| 11:39.46 | wasanzy | what is the invalid for? |
| 11:41.54 | ixyd | wasanzy it looks like you added a "dialstring" as queuemember, you can use a sip peer like SIP/1234 or a localchannel like Local/+12345@context as queuemember |
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| 11:43.42 | wasanzy | ixyd: from the cli, this is how I use to add it and it works: âqueue add member SIP/+233......@outgoing to queuename |
| 11:44.19 | ixyd | hm ok... |
| 11:45.21 | wasanzy | in diaplan I have this: same => n,AddQueueMember(bima,SIP/${CALLERID(num)}@outgoing ) |
| 11:45.58 | ixyd | and outgoing is your sipgw? |
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| 11:53.01 | wasanzy | sorry my laptop shutdown |
| 11:53.04 | wasanzy | any help? |
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| 12:51.00 | wasanzy | hi |
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| 13:22.34 | wasanzy | hello |
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| 13:24.28 | wasanzy | same => n,AddQueueMember(bima,SIP/${CALLERID(num)}@outgoing ) |
| 13:24.52 | wasanzy | is giving me SIP/+233271520077@outgoing (ringinuse disabled) (dynamic) (Invalid) has taken no calls yet Callers: |
| 13:25.00 | wasanzy | when I run queue show |
| 13:26.03 | [TK]D-Fender | Using "@" is not the best format for that, and I don't see anything showing the peer as valid or reachable |
| 13:26.51 | wasanzy | [TK]D-Fender: I don't see anything showing the peer as valid or reachable ( I don't understand) |
| 13:27.27 | [TK]D-Fender | WHERE'S THE SIP PEER? |
| 13:27.38 | [TK]D-Fender | Who says that is VALID? |
| 13:28.46 | wasanzy | I uses this "queue add member SIP/+233271520077@outgoing to bima" in the cli and it works |
| 13:29.30 | wasanzy | so I was wondering why AddQueueMember() doesn't work |
| 13:29.46 | [TK]D-Fender | I'm not seeing the full dump of the way that works to compare yet |
| 13:30.00 | [TK]D-Fender | that's already 2 things we should have now |
| 13:30.22 | wasanzy | ok let me get paste |
| 13:40.45 | wasanzy | https://paste.linux.community/view/524c063d |
| 13:41.12 | wasanzy | I have taken out @outgoing |
| 13:42.31 | [TK]D-Fender | Which makes no sense |
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| 13:43.17 | WIMPy | What is "outgoing"? |
| 13:43.35 | wasanzy | is a context |
| 13:43.53 | [TK]D-Fender | <[TK]D-Fender> WHERE'S THE SIP PEER? |
| 13:43.58 | [TK]D-Fender | I asked for backup. |
| 13:44.03 | [TK]D-Fender | sttill don't have it |
| 13:44.04 | WIMPy | So that's definitely wrng. |
| 13:44.26 | WIMPy | You don't add extensions to queues. |
| 13:44.27 | [TK]D-Fender | I don't see dumps from BOTH, or the rest of the config |
| 13:44.33 | WIMPy | You add channels. |
| 13:44.46 | [TK]D-Fender | or "devices" more specifically |
| 13:45.04 | [TK]D-Fender | which follow the valid spec for Dial() |
| 13:45.09 | WIMPy | For some definition of "device". |
| 13:46.16 | [TK]D-Fender | that is the actual * term for a section in sip.conf, DAHDI channel definition, etc |
| 13:46.26 | [TK]D-Fender | Channel is a live thing. |
| 13:46.32 | wasanzy | https://paste.linux.community/view/54e8311f |
| 13:47.07 | WIMPy | I don't think I have seen "device" with dahdi. |
| 13:47.41 | [TK]D-Fender | WIMPy, Still the proper "generic" term for the spec you'll use in the dialplan |
| 13:48.02 | WIMPy | Dialstring? |
| 13:48.22 | [TK]D-Fender | WIMPy, you dial a device (full spec), and that creates a channel (with a unique identifier tacked on the end). |
| 13:48.43 | [TK]D-Fender | Yes, dialstring would ahve been a sensible name for it, but "device" is whatt the offical docs used |
| 13:49.27 | [TK]D-Fender | wasanzy, Stop using "@" in those. SIP/peer/numbertodial <-------------------- |
| 13:49.47 | [TK]D-Fender | @ should only be used in actual free-form URI's |
| 13:52.17 | wasanzy | SIP/outgoing/number you mean? |
| 13:52.42 | [TK]D-Fender | yes |
| 13:52.55 | wasanzy | ok |
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| 14:48.33 | catphish | is there an off-the-shelf command line tool i can use to test that asterisk is responding to a simple OPTIONS request or similar, just to make sure neither asterisk nor chan_sip have crashed |
| 14:51.46 | catphish | looks like this script will probably do the job http://gekk.info/sipping/ |
| 14:52.01 | igcewieling | catphish: not that I'm aware of, but there are several tools you could use. sipp is one I think. |
| 14:52.24 | catphish | i looked at sipp first, but it seemed a little overkill, perhaps with the right options it would do the job |
| 14:53.01 | catphish | thanks |
| 14:54.44 | catphish | trying to use that tool, asterisk is ignoring the packets because "OPTIONS request has no from tag, dropping" |
| 14:54.50 | catphish | i'm sure i can fix this though |
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| 15:30.10 | sl4ck | hi guys |
| 15:30.30 | sl4ck | i am wondering if pbx asterisk can be used with docker container |
| 15:30.39 | sl4ck | thats is possible? |
| 15:30.46 | refeaime | Yes |
| 15:31.50 | [TK]D-Fender | https://www.google.ca/#q=running+asterisk+in+docker |
| 15:32.31 | [TK]D-Fender | I never understood asking questions in IRC that Google can answer instantly with 1/3 the words. |
| 15:33.19 | refeaime | [TK]D-Fender: well, maybe i am here noob, but i do other stuff in linux community. |
| 15:33.25 | igcewieling | lazyness or thinking their question is unique |
| 15:33.29 | refeaime | Sometimes talk to peoples is better. |
| 15:33.57 | refeaime | Sometimes you do not know which google sentence will be correct |
| 15:34.29 | [TK]D-Fender | <[TK]D-Fender> https://www.google.ca/#q=running+asterisk+in+docker <- if you can't come up with this you shouldn't be an IT admin in an capacity |
| 15:34.40 | [TK]D-Fender | any* |
| 15:34.46 | refeaime | Lol) |
| 15:34.58 | refeaime | Maybe u right. |
| 15:36.15 | refeaime | i, for example, never touched PBX in my life. Docs are cool, but i do some questions that need to be answered now. |
| 15:36.17 | refeaime | Anyway. |
| 15:36.20 | sl4ck | yeah . i know google can give you a lot of answers. But i thought the best answer is to get it from the original source. Its true I am a beginner. And I think thats whay exist this forum. Or its just for experts in here? |
| 15:36.33 | [TK]D-Fender | This isn't a complex idea. This was a "does it work?". The kind of question that will get a list of guides & results immediately. |
| 15:37.05 | [TK]D-Fender | This one was just the kind ofo thing you shove in Google, because that's the level of complexity of it |
| 15:37.16 | refeaime | Why do english community always walks in different direction then i imagine?... |
| 15:37.38 | [TK]D-Fender | "Is there an Apple iTunes version for Windows?". The OBVIOUS kind of stuff. |
| 15:37.57 | refeaime | [TK]D-Fender: does it works for FreeBSD?).. |
| 15:37.58 | igcewieling | I've just received reports from 3 different clients saying when they call out on a POTS line, they get an all circuits are busy message. Anyone here experiencing the same |
| 15:37.59 | [TK]D-Fender | refeaime, Your imagination is too little to be let out alone :) |
| 15:38.11 | refeaime | [TK]D-Fender: LOL. |
| 15:38.27 | [TK]D-Fender | igcewieling, HOW I CAN SIP?!?! |
| 15:38.45 | igcewieling | [TK]D-Fender: Step 1: obtain a straw |
| 15:39.19 | refeaime | [TK]D-Fender: yea, i never before come to english IRS or IM chats. Only ended and closed forums. That was really nice, comparing to IM where i always was... |
| 15:39.33 | [TK]D-Fender | igcewieling, I already got a full bail of it! The horse put up a fight, but I got it. IT NO WORK! |
| 15:40.27 | igcewieling | Step 1: Obtain an from a rooster |
| 15:40.33 | igcewieling | Step 1: Obtain an egg from a rooster |
| 15:43.26 | wasanzy | [TK]D-Fender: SIP/outgoing/+233271520077 (ringinuse disabled) (dynamic) (Invalid) has taken no calls yet |
| 15:44.13 | [TK]D-Fender | wasanzy, I asked for complete before & after w/ dumps. NOt 1 line at a time wirth an hour between them |
| 15:45.00 | wasanzy | sip peer dump? |
| 15:45.44 | [TK]D-Fender | AND the queue |
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| 15:46.17 | [TK]D-Fender | And that was for BOTH methods being compared |
| 15:49.05 | wasanzy | ok |
| 15:49.22 | wasanzy | to ask, is @ a special charater in Asterisk? |
| 15:50.46 | igcewieling | [TK]D-Fender: does he always do that? |
| 15:52.31 | [TK]D-Fender | Waste enormous amounts of time in providing backup requiring it to be repeated and never learning from that mistake from past incidents? |
| 15:52.36 | [TK]D-Fender | Absolutely |
| 15:54.21 | igcewieling | ah, so an idiot, not someone having a bad day tech wise. |
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| 16:57.54 | kfife | Any hint on how I can get the filename that features.conf (automixmon) will use to record the call? |
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| 16:58.27 | kfife | Format appears to be UniqueIDXXX-CLID-CPDN |
| 16:59.11 | kfife | I can get all but the XXX by stitching channel variables. |
| 16:59.30 | kfife | e.g. auto-1479142204-3123977003-13123653566 |
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| 17:04.10 | kfife | IOW, the uniqueID offered by ${UNIQUEID} appears to be different than the one used by the monitor file |
| 17:04.20 | kfife | usually by a few digits. |
| 17:08.58 | kfife | Appears to be a sequential time stamp, but the monitor uniqueID is not the same as the Channel uniqueID |
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| 17:20.35 | catphish | is there a command that will completely clean a source tree of any build files and config? |
| 17:21.32 | [TK]D-Fender | rm -rf |
| 17:22.00 | catphish | not sure that's the command i had in mind |
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| 17:43.23 | drmessano | rm -rf needs a response |
| 17:43.30 | drmessano | like "What are.. THOOOOOOOOOSE" |
| 17:50.04 | igcewieling | undelete with "rm -fr" |
| 17:50.42 | kfife | rm -jk |
| 17:53.54 | [TK]D-Fender | rm -oops |
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| 18:22.50 | igcewieling | Heh UK national health service accidentally sent an e-mail test to about 1%of the population. To make it worse, people are doing a Reply-All when replying. |
| 18:25.39 | [TK]D-Fender | #ccamplification |
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| 18:38.51 | drmessano | We had that happen at work.. 50,000 or so accounts |
| 18:39.02 | drmessano | Everyone kept doing a reply-alll |
| 18:39.16 | drmessano | "Stop replying ALL.. im trying to work" |
| 18:39.24 | drmessano | Replied... to.. alll |
| 18:41.39 | drmessano | We were on Exchange 2000 at the time... and we had servers at key locations across the country. So not only was it 1.5 million messages at one point, but they were all being sent server <> server across low-bandwidth links |
| 18:42.14 | drmessano | They ended up blowing out the whole mail queue and had to wait 2.5 hours for it to populate across all the clusters |
| 18:43.22 | drmessano | I knew the guy over our Corporate WAN group at the time. He said "You know what it's like to flip the kill switch on 1.5 million emails.. pretty awesome" |
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| 23:35.24 | UncleKiwi | hola |
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| 23:39.38 | UncleKiwi | i am thinking about getting a GSM FXS device and use that to forward calls too. However when I get a second call im wondering if I can somehow accomplish callwating functionality for the recieving caller where asterisk plays a beep to let him know a second call is waiting and maybe he can push # or some key to toggle between calls ? |
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| 23:42.31 | UncleKiwi | know what im asking ? |
| 23:51.00 | [TK]D-Fender | No, because FXS is for phones, not lines |
| 23:51.10 | [TK]D-Fender | and your usage description is vague, and no samples |
| 23:54.02 | UncleKiwi | sorry |
| 23:56.11 | UncleKiwi | what im trying to accomplish is .. when calls come in to the pbx the business owners mobile also rings via a GSM gateway attached to asterisk |
| 23:56.45 | UncleKiwi | reason is calls within that mobile network are free mobile to mobile |
| 23:57.53 | UncleKiwi | and yes its FXO |
| 23:58.08 | [TK]D-Fender | anf, no it isn't |
| 23:58.08 | UncleKiwi | and FXO gsm gateway |
| 23:58.14 | [TK]D-Fender | fxo = ANALOG |
| 23:58.31 | UncleKiwi | sorry |
| 23:59.11 | UncleKiwi | ahaha am new to this device i have not owned one before |
| 23:59.17 | UncleKiwi | but im thinking about getting one |
| 23:59.59 | UncleKiwi | http://www.ebay.com/itm/GoIP-VOIP-Gateway-GSM-Converter-SIP-IP-Phone-Adapter-GoIP-1-1-Channel-GSM-H-323-/280717507725?hash=item415c113c8d:g:rSsAAOSw44BYEFe9 |