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00:32.01 | hitech95 | Hi 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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06:15.42 | bllackjack | The 'stasis-core-control' task processor queue reached 500 scheduled tasks again. |
06:15.47 | bllackjack | Can anyone help me fix this error? |
06:15.52 | bllackjack | The reason why it appears |
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09:19.28 | bittis | random 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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09:20.15 | bittis | to lets say a max of 1500 bytes? (1472 to be specific) |
09:20.48 | bittis | i am having issues with fragmented udp packets and not sure how to best deal with it |
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10:08.52 | Samot | What are the issues exactly |
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10:10.57 | Samot | UDP has a max size of 65K roughly |
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10:11.37 | Samot | 65,500 bytes roughly |
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10:14.06 | Samot | And Asterisk cannot tell another device to change it's UDP packets. |
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11:48.38 | bittis | Thanks Samot |
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12:55.16 | hitech95 | Hi 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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14:18.14 | mneth1 | Any devs around that could help with a MixMonitor and recording sync/channel alignment issue? |
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14:35.34 | StucKman | i have a question anbout automixmon, automatic recording ans stop/starting it |
14:36.31 | StucKman | I 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 |
14:37.25 | StucKman | so call /¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯*3\__________*3/¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯\end of call, am I right? |
14:38.36 | StucKman | in 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 ? |
14:39.22 | StucKman | we're interested in such a feature so we can skip recording customer's sensitive info |
14:40.46 | Samot | That's covered in the wiki |
14:40.59 | StucKman | https://wiki.asterisk.org/wiki/display/AST/One-Touch+Features ? |
14:40.59 | Samot | You're looking at a dynamic feature code |
14:46.29 | StucKman | hmm I think I didn't ask the questions properly |
14:47.27 | StucKman | a( 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? |
14:48.14 | StucKman | b) 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? |
14:49.06 | Samot | MixMonitor will override the file, if it exists, unless told to append to it. |
14:50.24 | mneth1 | StucKman: it does work, we use it. When you trigger the recording resume MixMonitor for sure needs the append (I think it's 'a') option. |
14:50.57 | Samot | Yes. |
14:53.57 | StucKman | oks |
14:56.15 | StucKman | so Monitor("<call>", ",<path>,ma") ? |
14:56.26 | StucKman | sorry, I'm quite new to A* |
15:00.23 | Samot | No |
15:00.35 | Samot | Since that would be Monitor and not MixMonitor |
15:01.25 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_MixMonitor |
15:01.47 | StucKman | oks |
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16:08.14 | StucKman | do PauseMonitor()/UnpauseMonitor() work only with Monitor()? |
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16:18.15 | mandlbaum | when 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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16:20.23 | mandlbaum | I always have a heck of a time figuring out how to dial one properly depending on the phone or softphone |
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18:26.42 | qakhan | is there any way we can get queue incoming call connect status in Dialplan? |
18:41.25 | [TK]D-Fender | your description is too vague |
18:42.27 | qakhan | i want to know when call was answered in a queue. can i get that info thorough dialplan |
18:42.45 | [TK]D-Fender | WHAT call? |
18:42.51 | [TK]D-Fender | from when? |
18:43.08 | [TK]D-Fender | Why are you in the dialplan at that point? |
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18:43.44 | qakhan | customer incoming call in a queue. |
18:44.25 | [TK]D-Fender | that is not a clear and complete description |
18:44.32 | [TK]D-Fender | that's like a third of an idea. |
18:53.31 | Samot | https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_Queue |
18:53.52 | Samot | Calls only return to the dialplan if they cannot enter a queue |
18:53.55 | qakhan | ok, 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. |
18:53.55 | Samot | or the queue doesn't exist |
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18:54.12 | Samot | That has nothing do with what you asked |
18:54.23 | qakhan | i used exten => 0,1,Set(QUEUE_MEMBER=(regencytaxi,penalty,Local/7812@regencyagent,10)) |
18:54.28 | Samot | That is completely based on ring strategy and agent penalties. |
18:54.49 | qakhan | but queue was sending calls to all agents. |
18:55.02 | Samot | 1:54:26 PM <Samot> That is completely based on ring strategy and agent penalties. |
18:55.16 | Samot | Which are queue configurations, not dialplan. |
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18:56.12 | qakhan | now i am thinking to use PauseQueueMember for last 5 agents |
18:56.34 | rrittgarn | 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? |
18:56.57 | qakhan | and 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? |
18:58.03 | qakhan | [TK]D-Fender i am sorry i did not get you |
18:58.26 | [TK]D-Fender | QUEUE_MEMBER <- where do you see this as being a valid channel variable that DOES something? |
18:58.36 | [TK]D-Fender | And that what you ar setting it to is valid syntax? |
19:02.16 | qakhan | yes, i took example from * CLI QUEUE_MEMBER([queuename],option[,interface]) |
19:03.22 | [TK]D-Fender | that looks NOTHING like what you wrote |
19:03.42 | [TK]D-Fender | and you have again given a usless description. WHAT is that crap from * CLI? |
19:03.46 | [TK]D-Fender | you glued PARTIAL text. |
19:04.21 | qakhan | core show function QUEUE_MEMBER |
19:04.26 | Samot | Uhm. |
19:04.37 | Samot | Returns the number of members currently associated with the specified queuename. |
19:04.39 | qakhan | it gives detail of function |
19:04.53 | Samot | QUEUE_MEMBER() |
19:04.53 | Samot | Synopsis |
19:04.53 | Samot | Count number of members answering a queue. |
19:04.53 | Samot | Description |
19:04.53 | Samot | Returns the number of members currently associated with the specified queuename. |
19:05.03 | Samot | Like that? |
19:05.46 | qakhan | here https://pastebin.com/idtJ3QL8 |
19:06.07 | Samot | So it's not something you set |
19:06.11 | Samot | So what are you setting it? |
19:06.58 | qakhan | exten => 0,1,Set(QUEUE_MEMBER=(regencytaxi,penalty,Local/7812@regencyagent,10)) |
19:07.05 | Samot | Right |
19:07.13 | Samot | What variable are you SETTING? |
19:07.37 | Samot | As well |
19:08.01 | qakhan | QUEUE_MEMBER |
19:08.16 | Samot | You know what SET() does right? |
19:08.21 | Samot | It Sets a variable. |
19:08.32 | [TK]D-Fender | or a .... |
19:08.49 | Samot | So you are trying to set a variable with the same name as a function name? |
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19:09.45 | Kobaz | hmmmmm |
19:09.52 | Kobaz | anyone work with audiocodes mediant? |
19:10.06 | qakhan | ohh |
19:10.06 | Kobaz | trying to do a firmware update and while uploading the wizard just dies |
19:10.55 | Samot | So you are setting the penalty for this interface? |
19:11.00 | Samot | Or trying to get the information? |
19:11.18 | qakhan | i was setting the penalty |
19:16.39 | qakhan | Samot i was trying to use QUEUE_MEMBER function |
19:17.12 | qakhan | yes |
19:19.03 | Kobaz | audiocraps |
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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-Fender | Set(QUEUE_MEMBER= <----- NOT A FUNCTION. |
19:25.37 | [TK]D-Fender | Do you see brackets on the LEFT side of the equals sign? No? Then it's NOT a function call. |
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19:38.19 | jpsharp | I'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? |
19:46.03 | qakhan | [TK]D-Fender is it right syntax Set(QUEUE_MEMBER=regencytaxi,penalty,Local/7813@regencyagent,10) |
19:46.18 | Samot | jpsharp: #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. |
19:50.26 | rmudgett | qakhan: Set(QUEUE_MEMBER(params)=new_value) |
19:52.04 | [TK]D-Fender | Today's magic word : Left. |
19:52.14 | jpsharp | Samot: Thanks. had forgotten about that channel. |
19:52.37 | *** join/#asterisk areski (~areski@37.223.2.207) |
19:53.02 | Samot | I kept asking what he was trying to SET() while using a function name |
19:53.13 | jpsharp | Figured it out anyway. Didn't have the right flags in AST_MODULE_INFO |
19:53.54 | [TK]D-Fender | You use Set ... to set a function. |
19:54.08 | [TK]D-Fender | If you do something crazy like ... reference it proerly or something... |
19:54.11 | [TK]D-Fender | properly* |
19:58.25 | rrittgarn | Still 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? |
20:00.19 | qakhan | Thank you @rmudgett |
20:00.35 | qakhan | thank you [TK]D-Fender and Samot |
20:00.40 | qakhan | its working now |
20:04.43 | *** part/#asterisk jpsharp (~jsharp@linode.fivecats.org) |
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20:40.21 | *** join/#asterisk danielyk (~danielyk@p4FC7D6E7.dip0.t-ipconnect.de) |
20:40.33 | danielyk | It is able to send a SMS with PJSIP? |
20:41.50 | [TK]D-Fender | depends if your provider supports SIP MESSAGE as a means of transmitting |
20:42.59 | danielyk | I 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-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_MESSAGE |
20:44.15 | [TK]D-Fender | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MessageSend |
20:44.24 | [TK]D-Fender | There are samples on the WIKI |
20:44.27 | [TK]D-Fender | look around |
20:49.53 | Samot | danielyk: Who is the provider? |
20:50.05 | danielyk | @Samot: Telekom Germany |
20:50.55 | Samot | OK, so it is possible they send it via SIP MESSAGE or other standard format |
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20:58.21 | Kobaz | poor asterisk 13 |
20:58.28 | Kobaz | i've locked it up like 6 times today |
20:59.03 | Kobaz | i'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 |
20:59.55 | Samot | What version of 13? |
21:00.04 | Kobaz | 13.17.2 |
21:00.47 | Samot | Try to update. |
21:00.51 | Samot | See if that fixes ths iss. |
21:00.55 | Samot | issue |
21:01.48 | Kobaz | yeaaah |
21:01.56 | Kobaz | very production system |
21:01.59 | Kobaz | on a very emergency cutover |
21:12.01 | Kobaz | it's one of those, "we have this old system, and we need to migrate to the new one" |
21:12.15 | Kobaz | so and so is on vacation this week, we can't do it |
21:12.23 | Kobaz | we have meetings all next week, yeah lets push it |
21:12.31 | Kobaz | OUR OLD SYSTEM DIED, CUT IT OVER NOW |
21:16.55 | Kobaz | and of course i inherited this... and the original guy left in a bajillion hard coded test items |
21:19.09 | Samot | Well |
21:19.16 | Samot | You mean the system itself is old? |
21:19.19 | Samot | The hardware? |
21:19.30 | Samot | Because 13.17.2 isn't that old. |
21:19.54 | Samot | 13.19 just was release a week or so ago |
21:21.13 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
21:22.33 | Kobaz | no old system is dead dead |
21:22.44 | Kobaz | the new box is 13.17.2 |
21:22.54 | Samot | Well then updating Asterisk might not help with the deadlocks. |
21:22.59 | Kobaz | yeah |
21:23.09 | Kobaz | i have to reproduce the issue after hours, get the exact sequence |
21:23.16 | Kobaz | and then try on 13.19 |
21:23.34 | Kobaz | but, heh this is quite common :P |
21:23.40 | Kobaz | i try a new asterisk, and i break it |
21:24.18 | *** join/#asterisk jjrh (~weechat12@ppp-199-167-117-242.storm.ca) |
21:35.59 | danielyk | @[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? |
21:55.07 | n\ack | Strange. I'm getting an error about a dynamic feature |
21:55.08 | n\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/... |
21:55.23 | n\ack | testfeature isn't in features.conf, nor dialplan, |
22:01.45 | n\ack | If I want to remove a dynamic feature, i need to reload the dialplan and reload the feature module ONLY, is that right? |
22:02.11 | n\ack | (after removing references to it in extensions.conf and features.conf of course) |
22:03.04 | n\ack | Is that right? Or do I need to reload channel drivers or somethign to purge the residual state? |
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