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| 01:19.05 | *** 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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| 05:20.58 | drmessano | So I got this sponsored post on Facebook for Asterisk World, stating "See some of your favourite Astricon speakers and David Duffett at Asterisk World 2018" |
| 05:21.04 | drmessano | I have several issues with this |
| 05:22.19 | drmessano | First off, this sponsored post was on my American Facebook page and clearly the English/UK version of "Favourite" as used. I almost didn't understand the post due to this error |
| 05:23.12 | drmessano | Second, I read the speaker list, and none of them are favorites or "favourites". I now have zero reason to attend. |
| 05:24.34 | drmessano | Third, "favourite Astricon speakers and David Duffett" makes Mr Duffett sound like a consolation prize, not being grouped in with the "favourites". |
| 05:24.59 | drmessano | Whoever submitted this post from the marketing department should be shot, then hung. |
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| 05:26.15 | klow | Hello, I'm trying to figure out why 'pjsip set logger on' is telling me logging is enabled, but I am not getting any log messages to asterisk console or to my log file, and I'm certain i have some pjsip traffic going |
| 05:26.32 | klow | i compiled asterisk myself (14.6) .. wondering if i am missing a module load or something |
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| 05:40.31 | klow | also, is there any way to get better info on what would cause this? [Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id 'corona' from configuration file 'pjsip.conf' |
| 05:40.44 | klow | corona in this case is an endpoint which is supposed to set up my trunk to Twilio. |
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| 07:55.04 | klow | If anyone is able to take a look, would certainly appreciate it. posted to asterisk-users as well |
| 07:55.05 | klow | https://pastebin.com/4SzU5gxy |
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| 08:55.05 | ChannelZ | klow: Well for one I think your aors=twilio-aors needs to be up in the endpoint config rather than in that [twilio] group by its self which is ambiguous and doesn't do anything |
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| 09:00.11 | ChannelZ | also your aors=1001 under [1001](endpoint-basic) doesn't do anything because you have no type=aor entry named 1001 |
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| 12:42.51 | Samot | klow: [twilio](!) <-- Why are you using a template when you've never called on it? Templates are ignored. |
| 12:45.14 | file | there's also options which don't exist in PJSIP under the given name which would prevent creation, and that is logged on the console |
| 12:45.48 | Samot | canreinvite=no <-- like that |
| 12:46.01 | Samot | Not only is not a PJSIP setting, it's a deprecated Chan_SIP setting. |
| 12:46.06 | Samot | For almost a decade. |
| 12:50.01 | Samot | This PJSIP config is filled with issues. |
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| 15:22.15 | razzex | Hello, i need to customize Asterisk in a way that it could analyze SIP message and print info to CLI - for example the field Content Length, if the value is negative, asterisk would find out and inform the user. Is it possible? |
| 15:22.33 | razzex | More specifically, which conf file would I need to edit? |
| 15:23.16 | file | that's not really something that can be configured... |
| 15:23.25 | file | chan_sip will parse and throw some warnings/errors here and there |
| 15:23.33 | file | chan_pjsip is similar but reports it differently |
| 15:24.24 | Samot | If you need to manipulate, read, take apart, parse, etc the SIP requests/replies you need a SIP proxy. |
| 15:24.34 | razzex | I have tried sending modified messages with bad values and some PBXs noticed the bad values, some not. So i want to edit them so they analyze the messages automatically |
| 15:24.45 | Samot | Or a SBC or something that is designed to specifically do that. |
| 15:24.51 | razzex | i see |
| 15:24.54 | Samot | You need a SIP Proxy |
| 15:25.04 | Samot | Asterisk is not a proxy or a softswitch. |
| 15:25.14 | Samot | It doesn't repair messages. |
| 15:25.38 | razzex | Well i dont intent to repair them, just decide if its valid or not and either accept it or discard it |
| 15:25.58 | razzex | do i need proxy to that aswell? |
| 15:26.09 | file | SIP parsers already do that, it just depends on how lenient the parser is on the outcome |
| 15:28.31 | razzex | okay, so all i need is just to modify the parser and make it more thorough :) |
| 15:32.20 | Samot | Modifying the parser is modifying Asterisk. |
| 15:32.33 | Samot | And then recompiling Asterisk. |
| 15:33.43 | Samot | Also making Asterisk hard to support when you have issues because it's been modified. |
| 15:34.26 | razzex | yeah i understand, this is more of a research kind of thing |
| 15:35.17 | Samot | Also, when you say SIP message do you mean an actual SIP request and/or reply message/packet |
| 15:35.29 | Samot | Or do you mean SIP MESSAGE as in sending a body of text over SIP? |
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| 15:43.12 | razzex | i mean request/reply, or like an INVITE message |
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| 15:52.47 | Samot | Well.. |
| 15:52.55 | Samot | Asterisk is going to reject malformed INVITEs. |
| 15:53.14 | Samot | But how are you planning on informing the "user"? |
| 15:53.16 | Samot | Which user? |
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| 16:01.55 | razzex | well at first i would like asterisk to detect it and then maybe inform about malformed invite in asterisk CLI. It could then be forwarded somehow to the source of the message |
| 16:10.19 | Samot | It already will warn you. |
| 16:10.37 | Samot | And it will send back a reply to the source per SIP RFC's. |
| 16:10.47 | [TK]D-Fender | What is the nature of your reason for doing this? Do you have problems with a specific user, or group of users? |
| 16:10.53 | Samot | This is standard SIP transactions. |
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| 16:12.54 | razzex | I have tried sending malformed messages to various versions of ASterisk. Some detected the bad messages, some didnt. By editing the parser i just wanna make sure all version will discard these bad messages. |
| 16:13.27 | Samot | All versions? |
| 16:13.35 | Samot | There are two supported version 13 and 15 |
| 16:13.43 | Samot | Using anything else is completely unsupported. |
| 16:16.12 | Samot | I said this earlier, if you want to do anything specific with a SIP message including how you reply/handle that message you either A) Get something that is designed for that or B) Rewrite core parts of Asterisk and recompile, making your compiled version unsupported because the source has been modified. |
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| 22:10.52 | josefig | Hi team, how can I identify the call files when it says "attempt_thread: Queued call" ? I added the option Archive: yes but I don't find the expired file in outgoing_done ? |
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| 23:02.56 | |ance|ott | hi guys, I am facing an issue when I try to perform from dialplan using the System app: |
| 23:02.59 | |ance|ott | 'channel request hangup Local/3005520002@from-albertotest-3005520002-0000007a;1' |
| 23:03.17 | |ance|ott | it doesn't work because the ";" at the end of the channel'sname |
| 23:03.31 | |ance|ott | is there a way to workaround this issue? |
| 23:03.34 | Samot | That's incorrect. |
| 23:03.45 | Samot | Local channels always have a ;1 or a ;2 |
| 23:03.55 | Samot | Since it's a two piece channel. |
| 23:04.16 | |ance|ott | i know that, so how can I kill alocal channel from dialplan? |
| 23:04.32 | |ance|ott | i've tried from dialplan and AGI and getting same issue |
| 23:04.43 | Samot | Well ;1 goes away when ;2 happens |
| 23:04.48 | Samot | So are you killing the right one? |
| 23:05.13 | |ance|ott | if I ssh into the server an perform: asterisk -rx "channel request hangup Local/3005520002@from-albertotest-3005520002-0000007a;1" |
| 23:05.17 | |ance|ott | it does properly |
| 23:05.27 | |ance|ott | it doesn't work from dialplan or AGI |
| 23:05.31 | Samot | OK so you may have to escape the ; |
| 23:05.45 | |ance|ott | yep, that's what I was thinking |
| 23:05.50 | Samot | Because in dialplan ; = comment line so it has to be escaped |
| 23:05.54 | |ance|ott | so it might be like: |
| 23:06.02 | Samot | And ; is a special character in most code. |
| 23:06.03 | |ance|ott | asterisk -rx "channel request hangup Local/3005520002@from-albertotest-3005520002-0000007a\;1" |
| 23:06.11 | |ance|ott | that's right |
| 23:06.15 | Samot | Well from the script, yes. |
| 23:06.18 | Samot | From the cli no. |
| 23:06.39 | |ance|ott | let me try from script |
| 23:08.54 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
| 23:16.23 | *** part/#asterisk kharwell (kharwell@nat/digium/x-xghnaebsszjcpbzq) |
| 23:19.38 | |ance|ott | that worked ;) thanks @Samot |
| 23:20.33 | *** join/#asterisk |ance|ott (~|ance|ott@181.135.155.249) |
| 23:42.54 | *** join/#asterisk paulgrmn_ (~paulgrmn@184.75.212.68) |
| 23:50.12 | *** join/#asterisk Samael28 (~Samael28@176.104.56.91) |
| 23:56.48 | *** join/#asterisk zapata (~zapata@2a02:b18:581:10:25b8:97e7:4e17:9288) |