IRC log for #asterisk on 20180124

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00:32.01hitech95Hi guys I have a question about the new pjsip, i have seen a strange behavior. Looks like that if I specify a outbound proxy on the REGISTER request the Authorization parameter is not sent. Is this correct or what?
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01:18.26*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.19.0 (2018/01/11), Standard: 15.2.0 (2018/01/11); DAHDI: DAHDI-linux 2.11.1 (2016/03/01), DAHDI-tools 2.11.1 (2016/03/01); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
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06:15.42bllackjackThe 'stasis-core-control' task processor queue reached 500 scheduled tasks again.
06:15.47bllackjackCan anyone help me fix this error?
06:15.52bllackjackThe reason why it appears
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09:19.28bittisrandom question, is there some way asterisk can request from a remote end point to limit the maximum size of udp packets received?
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09:20.15bittisto lets say a max of 1500 bytes? (1472 to be specific)
09:20.48bittisi am having issues with fragmented udp packets and not sure how to best deal with it
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10:08.52SamotWhat are the issues exactly
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10:10.57SamotUDP has a max size of 65K roughly
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10:11.37Samot65,500 bytes roughly
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10:14.06SamotAnd Asterisk cannot tell another device to change it's UDP packets.
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11:48.38bittisThanks Samot
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12:55.16hitech95Hi guys could someone explain to me if this is the correct SIP protocol response or if is my ISP SIP server that work in a odd way? http://pastebin.freepbx.org/view/a3dcd59d#L96
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14:18.14mneth1Any devs around that could help with a MixMonitor and recording sync/channel alignment issue?
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14:35.34StucKmani have a question anbout automixmon, automatic recording ans stop/starting it
14:36.31StucKmanI read here: http://forums.asterisk.org/viewtopic.php?f=1&t=87976&p=191755&hilit=Automixmon&sid=96b0d8fbe477c3b5d0c4cf585761fc39#p191755 that, if you call are recorded, and automixmon activated, you can use a feature (classically *3) to stop and start recording
14:37.25StucKmanso call /¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯*3\__________*3/¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯\end of call, am I right?
14:38.36StucKmanin particular what happens to the contens of the file when *3 is pressed the second time and recording(re)starts? does the file start from 0 or does it continue ?
14:39.22StucKmanwe're interested in such a feature so we can skip recording customer's sensitive info
14:40.46SamotThat's covered in the wiki
14:40.59StucKmanhttps://wiki.asterisk.org/wiki/display/AST/One-Touch+Features ?
14:40.59SamotYou're looking at a dynamic feature code
14:46.29StucKmanhmm I think I didn't ask the questions properly
14:47.27StucKmana( is is true what the forum posts says, that you can stop and restart recording with *3 when the recording is automatic for the call?
14:48.14StucKmanb) if true, if I restart recording, does it overwrite what has already been recorded, it continues where it left, or does it create another file?
14:49.06SamotMixMonitor will override the file, if it exists, unless told to append to it.
14:50.24mneth1StucKman: it does work, we use it. When you trigger the recording resume MixMonitor for sure needs the append (I think it's 'a') option.
14:50.57SamotYes.
14:53.57StucKmanoks
14:56.15StucKmanso Monitor("<call>", ",<path>,ma") ?
14:56.26StucKmansorry, I'm quite new to A*
15:00.23SamotNo
15:00.35SamotSince that would be Monitor and not MixMonitor
15:01.25Samothttps://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_MixMonitor
15:01.47StucKmanoks
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16:08.14StucKmando PauseMonitor()/UnpauseMonitor() work only with Monitor()?
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16:18.15mandlbaumwhen dialing a sip URI, is it normal that some softphones might try to dial the IP address or the xxxx@ prefix as a PSTN phone number
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16:20.23mandlbaumI always have a heck of a time figuring out how to dial one properly depending on the phone or softphone
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18:26.42qakhanis there any way we can get queue incoming call connect status in Dialplan?
18:41.25[TK]D-Fenderyour description is too vague
18:42.27qakhani want to know when call was answered in a queue. can i get that info thorough dialplan
18:42.45[TK]D-FenderWHAT call?
18:42.51[TK]D-Fenderfrom when?
18:43.08[TK]D-FenderWhy are you in the dialplan at that point?
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18:43.44qakhancustomer incoming call in a queue.
18:44.25[TK]D-Fenderthat is not a clear and complete description
18:44.32[TK]D-Fenderthat's like a third of an idea.
18:53.31Samothttps://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_Queue
18:53.52SamotCalls only return to the dialplan if they cannot enter a queue
18:53.55qakhanok, so there is a caller 3049849900. he calls and there are 10 agents avaiable to answer the call. but i want to send that call to 1st 5 agents.
18:53.55Samotor the queue doesn't exist
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18:54.12SamotThat has nothing do with what you asked
18:54.23qakhani used exten => 0,1,Set(QUEUE_MEMBER=(regencytaxi,penalty,Local/7812@regencyagent,10))
18:54.28SamotThat is completely based on ring strategy and agent penalties.
18:54.49qakhanbut queue was sending calls to all agents.
18:55.02Samot1:54:26 PM <Samot> That is completely based on ring strategy and agent penalties.
18:55.16SamotWhich are queue configurations, not dialplan.
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18:56.12qakhannow i am thinking to use PauseQueueMember for last 5 agents
18:56.34rrittgarnis there a function/application in dialplan that will allow you to trigger a sip notify to a peer that isn't connected to the channel calling the function?
18:56.57qakhanand when 3049849900 call connects with agent then UnPauseQueueMember the last 5 agents
18:57.16[TK]D-Fender<qakhan> i used exten => 0,1,Set(QUEUE_MEMBER=(regencytaxi,penalty,Local/7812@regencyagent,10)) <- what says this variable can be used in this way and has a meaningful effect?
18:58.03qakhan[TK]D-Fender i am sorry i did not get you
18:58.26[TK]D-FenderQUEUE_MEMBER <- where do you see this as being a valid channel variable that DOES something?
18:58.36[TK]D-FenderAnd that what you ar setting it to is valid syntax?
19:02.16qakhanyes, i took example from * CLI QUEUE_MEMBER([queuename],option[,interface])
19:03.22[TK]D-Fenderthat looks NOTHING like what you wrote
19:03.42[TK]D-Fenderand you have again given a usless description.  WHAT is that crap from * CLI?
19:03.46[TK]D-Fenderyou glued PARTIAL text.
19:04.21qakhancore show function QUEUE_MEMBER
19:04.26SamotUhm.
19:04.37SamotReturns the number of members currently associated with the specified queuename.
19:04.39qakhanit gives detail of function
19:04.53SamotQUEUE_MEMBER()
19:04.53SamotSynopsis
19:04.53SamotCount number of members answering a queue.
19:04.53SamotDescription
19:04.53SamotReturns the number of members currently associated with the specified queuename.
19:05.03SamotLike that?
19:05.46qakhanhere https://pastebin.com/idtJ3QL8
19:06.07SamotSo it's not something  you set
19:06.11SamotSo what are you setting it?
19:06.58qakhanexten => 0,1,Set(QUEUE_MEMBER=(regencytaxi,penalty,Local/7812@regencyagent,10))
19:07.05SamotRight
19:07.13SamotWhat variable are you SETTING?
19:07.37SamotAs well
19:08.01qakhanQUEUE_MEMBER
19:08.16SamotYou know what SET() does right?
19:08.21SamotIt Sets a variable.
19:08.32[TK]D-Fenderor a ....
19:08.49SamotSo you are trying to set a variable with the same name as a function name?
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19:09.45Kobazhmmmmm
19:09.52Kobazanyone work with audiocodes mediant?
19:10.06qakhanohh
19:10.06Kobaztrying to do a firmware update and while uploading the wizard just dies
19:10.55SamotSo you are setting the penalty for this interface?
19:11.00SamotOr trying to get the information?
19:11.18qakhani was setting the penalty
19:16.39qakhanSamot i was trying to use QUEUE_MEMBER function
19:17.12qakhanyes
19:19.03Kobazaudiocraps
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19:24.53[TK]D-Fender<qakhan> Samot i was trying to use QUEUE_MEMBER function <- you weren't even calling a function
19:25.13[TK]D-FenderSet(QUEUE_MEMBER= <----- NOT A FUNCTION.
19:25.37[TK]D-FenderDo you see brackets on the LEFT side of the equals sign?  No?  Then it's NOT a function call.
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19:38.19jpsharpI'm having a horrible brainfart.  I'm trying to build a custom res_ module, but none of the other modules can seem to see it.  They always fail to load with "undefined symbol".  How do I tell the build environemtn to export the symbols from my custom res_ module?
19:46.03qakhan[TK]D-Fender is it right syntax Set(QUEUE_MEMBER=regencytaxi,penalty,Local/7813@regencyagent,10)
19:46.18Samotjpsharp: #asterisk-dev <-- That's the place.
19:49.51[TK]D-Fender<[TK]D-Fender> Do you see brackets on the LEFT side of the equals sign?  No?  Then it's NOT a function call.
19:50.26rmudgettqakhan: Set(QUEUE_MEMBER(params)=new_value)
19:52.04[TK]D-FenderToday's magic word : Left.
19:52.14jpsharpSamot: Thanks.  had forgotten about that channel.
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19:53.02SamotI kept asking what he was trying to SET() while using a function name
19:53.13jpsharpFigured it out anyway. Didn't have the right flags in AST_MODULE_INFO
19:53.54[TK]D-FenderYou use Set ... to set a function.
19:54.08[TK]D-FenderIf you do something crazy like ... reference it proerly or something...
19:54.11[TK]D-Fenderproperly*
19:58.25rrittgarnStill wondering, is there a function/application in dialplan that will allow you to trigger a sip notify to a peer that isn't connected to the channel calling the function?
20:00.19qakhanThank you @rmudgett
20:00.35qakhanthank you [TK]D-Fender and Samot
20:00.40qakhanits working now
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20:40.33danielykIt is able to send a SMS with PJSIP?
20:41.50[TK]D-Fenderdepends if your provider supports SIP MESSAGE as a means of transmitting
20:42.59danielykI do not know it exactly but I would try it. What is the right command to send a SIP MESSAGE?
20:44.10[TK]D-Fenderhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE
20:44.15[TK]D-Fenderhttps://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MessageSend
20:44.24[TK]D-FenderThere are samples on the WIKI
20:44.27[TK]D-Fenderlook around
20:49.53Samotdanielyk: Who is the provider?
20:50.05danielyk@Samot: Telekom Germany
20:50.55SamotOK, so it is possible they send it via SIP MESSAGE or other standard format
20:54.38*** join/#asterisk imcdona (~imcdona@2607:f0d8:20:1001:29aa:6539:5d0e:79a)
20:58.21Kobazpoor asterisk 13
20:58.28Kobazi've locked it up like 6 times today
20:59.03Kobazi'll try and reproduce after hours... basically if you have dialplan reload going, and you're trying to do other dialplan related stuff, you get deadlocks
20:59.55SamotWhat version of 13?
21:00.04Kobaz13.17.2
21:00.47SamotTry to update.
21:00.51SamotSee if that fixes ths iss.
21:00.55Samotissue
21:01.48Kobazyeaaah
21:01.56Kobazvery production system
21:01.59Kobazon a very emergency cutover
21:12.01Kobazit's one of those, "we have this old system, and we need to migrate to the new one"
21:12.15Kobazso and so is on vacation this week, we can't do it
21:12.23Kobazwe have meetings all next week, yeah lets push it
21:12.31KobazOUR OLD SYSTEM DIED, CUT IT OVER NOW
21:16.55Kobazand of course i inherited this... and the original guy left in a bajillion hard coded test items
21:19.09SamotWell
21:19.16SamotYou mean the system itself is old?
21:19.19SamotThe hardware?
21:19.30SamotBecause 13.17.2  isn't that old.
21:19.54Samot13.19 just was release a week or so ago
21:21.13*** join/#asterisk Samael28 (~Samael28@176.104.56.91)
21:22.33Kobazno old system is dead dead
21:22.44Kobazthe new box is 13.17.2
21:22.54SamotWell then updating Asterisk might not help with the deadlocks.
21:22.59Kobazyeah
21:23.09Kobazi have to reproduce the issue after hours, get the exact sequence
21:23.16Kobazand then try on 13.19
21:23.34Kobazbut, heh this is quite common :P
21:23.40Kobazi try a new asterisk, and i break it
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21:35.59danielyk@[TK]D-Fender: I use the command MessageSend(pjsip:endpoint/sip:mobilephonenumber@tel.t-online.de,myprivatenumber@t-online.de). But the SIP MESSAGE is not sent to the actual IP address of the registration. Do you have an idea?
21:55.07n\ackStrange.  I'm getting an error about a dynamic feature
21:55.08n\ack[Jan 24 15:24:55] WARNING[16868][C-00006990]: features_config.c:1306 ast_get_chan_applicationmap: Unknown DYNAMIC_FEATURES item 'testfeature' on channel DAHDI/i1/...
21:55.23n\acktestfeature isn't in features.conf, nor dialplan,
22:01.45n\ackIf I want to remove a dynamic feature, i need to reload the dialplan and reload the feature module ONLY, is that right?
22:02.11n\ack(after removing references to it in extensions.conf and features.conf of course)
22:03.04n\ackIs that right?  Or do I need to reload channel drivers or somethign to purge the residual state?
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