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11:37.36 | grummund | Is there a way to control the DND state of a phone (Cisco SPA303) from asterisk? |
11:38.33 | grummund | say to automatically set DND for outside office hours. |
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12:49.18 | grummund | There appear to be some dialplan hints registered without any config for this... |
12:49.49 | grummund | any ideas where this is setup? - http://paste.debian.net/1180028/ |
12:58.39 | mrtnt | Let's say that one observes large quantities of UDP traffic destioned to various addresses across the Internet to UDP port 5060. What kind of attack this might be? SIP brute force? |
13:04.49 | sibiria | if your trunks are wide-open to the Internet, yes, likely the usual drive-by attempts of registering/inviting |
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14:20.25 | laerus | hello, in the ARI Dial event there is 'caller' field that is optional, when is it set, or how can i make this be set? |
14:21.08 | laerus | i have set 'caller'field when i hit the endpoint /channels/../dial |
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15:31.45 | mrtnt | sibiria: ok, thanks! And the idea of such registering attacks is to find a PBX which does not require authentication, register the phone and then call in to that number? |
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15:45.26 | edgars | hi! |
15:46.24 | seanbright | edgars: good morning |
15:47.19 | edgars | 17:47 here ;D |
15:49.02 | edgars | i know, its not right place, but i will ask. Is it possible tu upgrade asterisk now with asterisk 11 to freepbx distro with asterisk 16 |
15:51.15 | seanbright | unsure - no one responding in #freepbx? |
15:51.36 | sibiria | mrtnt: it's almost always to call up pay-per-call numbers to steal funds |
15:52.22 | sibiria | it's quite the lucrative crime model |
16:00.21 | edgars | seanbright: yeah, nobody responding |
16:01.05 | edgars | seanbright: i tried to import mysql dump, well... half interface is not working |
16:03.00 | edgars | ohh well, now its not working at all |
16:03.02 | edgars | :/ |
16:08.54 | laerus | hey, any idea why the ARI Dial event is missing the 'caller' field even though the caller name appears on the call and is included in the channel objects of 'peer'? i am setting the caller field on /channels/id/dial |
16:10.57 | seanbright | edgars: i'm not familiar with freepbx, so it might be best to just start from scratch |
16:12.28 | Samot | 11:12:04 AM <Samot> You do a backup of Asterisk Now and then do a restore in FreePBX 15 |
16:12.33 | Samot | From the other channel. |
16:20.01 | seanbright | laerus: what version of asterisk? |
16:20.14 | laerus | seanbright, 16 |
16:20.44 | seanbright | can you pastebin the event you are talking about? |
16:20.45 | seanbright | ~pb |
16:20.45 | infobot | pastebin is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: https://pastebin.com, https://paste.ubuntu.com, http://paste.debian.net; or install pastebinit with yum or aptitude. |
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16:23.32 | laerus | seanbright, http://paste.debian.net/1180050/ |
16:24.13 | jkroon | edgars, db structures probably changed, so fresh install, then dump the database with --no-extended-insert, then you execute the INSERT INTO rows ... you may need to modify it a bit in order to get the INSERTs working. |
16:24.20 | jkroon | then make some other change via the UI. |
16:24.33 | jkroon | not a freepbx user ... but that would be my strategy. |
16:24.44 | jkroon | disclaimer: it may well not work either. |
16:26.33 | edgars | jkroon: yeah, thats what im trying to do now |
16:29.27 | seanbright | laerus: the docs say that 'caller' on the 'Dial' event is 'The calling channel.' |
16:29.33 | seanbright | laerus: i don't think there is a calling channel in this case |
16:29.53 | seanbright | https://github.com/asterisk/asterisk/blob/16/rest-api/api-docs/events.json#L831-L835 |
16:32.15 | laerus | seanbright, i'm setting the 'caller' field on the /channels/id/dial request though, also tried with channel originate and setting the 'originator' field |
16:32.17 | seanbright | infobot: no, pastebin is <action> is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: https://dpaste.org, https://pastebin.com, https://paste.ubuntu.com, http://paste.debian.net; or install pastebinit with yum or aptitude. |
16:32.17 | infobot | okay, seanbright |
16:33.23 | seanbright | laerus: what are you setting the caller field too? |
16:33.25 | seanbright | to* |
16:36.52 | seanbright | laerus: ARI tries to find a channel with the name provided by the 'caller' field |
16:37.19 | seanbright | if it doesn't find one, it won't use it |
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16:37.50 | seanbright | oh wait. it looks by channel name and then by channel UID |
16:38.04 | laerus | seanbright, yea i am setting it to be the channel id of the caller, |
16:38.18 | laerus | the name of the caller appears on the callee |
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16:38.38 | seanbright | interesting |
16:38.44 | laerus | do make this out from the source code? |
16:39.07 | laerus | i have delved into asteirsk source code atm to find out what's going on, it's the first time so i'm a bit lost still |
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16:43.41 | seanbright | i am scanning through now |
16:50.01 | seanbright | laerus: would you be able to test a patch that adds some debug logging? what specific version of asterisk 16? |
16:51.29 | laerus | seanbright, yes, Asterisk 16.2.1~dfsg-1+deb10u2 on debian buster |
16:52.15 | seanbright | is there no more up-to-date version packaged already? |
16:53.02 | seanbright | the issue may already be fixed in newer versions |
16:53.34 | laerus | hmm maybe if there is a backport, on buster main it's the latest version |
16:56.09 | seanbright | we can start simple... in main/stasis_channels.c around line 356 you should see: |
16:56.20 | seanbright | if (caller) { |
16:57.02 | seanbright | just before that line add: ast_log(LOG_ERROR, "caller: %p\n", caller); |
16:57.14 | seanbright | build, install, test |
16:58.07 | grummund | Debian testing (bullseye) is Asterisk 16.15.0~dfsg-1. |
16:58.27 | grummund | not sure about 'sid'. |
17:01.45 | laerus | i will try bto uild the debian package with the log added and come back to you |
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17:03.08 | grummund | Debian unstable (sid) is also Asterisk 16.15.0~dfsg-1. |
17:03.38 | seanbright | that code hasn't really been touched, so i doubt it is fixed in newer versionf |
17:03.41 | seanbright | versions* |
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17:47.48 | seanbright | laerus: any luck? |
17:58.41 | ckb | Hi guys, I literally just found out about the CLI, but am having problems finding out how to list all available extensions. |
17:59.53 | grummund | sip show peers ? |
18:01.44 | laerus | seanbright, sorry i had to take a break and i also need to build from debian source, is there any other medium to reach out to you? i've also opened this https://community.asterisk.org/t/ari-dial-event-caller-field/87046 |
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18:53.48 | zgu | "UDP is usually the standard, but this is not an available option unless enabled via provisioning. If your SIP server uses UDP, you will have to provision this phone through a centralized provisioning server to enable UDP." well that's pretty retarded |
18:59.08 | Samot | zgu: What phone is this? |
18:59.49 | zgu | avaya j129 |
19:00.09 | zgu | looks like i have to put a settings file on an http server |
19:06.26 | seanbright | laerus: nah. i'm always around here, even when i'm not around. |
19:07.39 | laerus | cool! i'm trying to build from latest debian 16.15.0 atm |
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19:50.29 | laerus | seanbright, i got this log 3 times `ERROR[44692]: stasis_channels.c:355 ast_channel_publish_dial_internal: caller: (nil)` |
19:51.35 | laerus | i've tried with Asterisk 16.15.0~dfsg-1.5 |
19:53.33 | seanbright | ok, so what are you passing as the 'caller' argument |
19:58.46 | laerus | `"caller": "1610049456.3"` the channel id i get from the StasisStart event |
19:59.30 | laerus | the caller name appears correctly on the channel object in the the Dial 'peer' field |
20:00.48 | laerus | in the `connected` field of the `peer` channel object actually, not in the `caller` field of the channel object, which is also odd |
20:05.58 | seanbright | that does not look right |
20:06.08 | seanbright | pastebin the stasisstart event |
20:13.24 | seanbright | i guess i could set up a test here but i really don't wanna |
20:13.56 | laerus | i am creating a pastebin with some of the flow, i have logged my requests to ARI as well |
20:15.09 | laerus | http://paste.debian.net/1180095/ |
20:16.32 | laerus | i have logged everything except the ChannelSetvar events really, including my requests/responses to ARI if you want all the info |
20:16.58 | laerus | ChannelVarset* |
20:19.16 | laerus | on the last Dial event it seems that `caller` and `connected` fields of the `peer` channel objects are reversed. 7000 is the actual caller |
20:22.03 | seanbright | it would seem to be a bug to me... however i know very little overall about statis/ari/etc |
20:22.24 | seanbright | so i would need to lab it up to figure out what was going on |
20:23.12 | seanbright | but... and this is critical... i don't want to |
20:23.25 | laerus | want me to open a bug report? |
20:23.46 | seanbright | kinda. but if it gets closed don't get mad at me. |
20:23.52 | laerus | it seems like there are two bugs here then? |
20:25.06 | seanbright | if you do open an issue, please be sure to include comprehensive reproduction steps. the easier it is for someone to reproduce the more likely it will get addressed. |
20:27.28 | laerus | yes i will add all the json logs that include the requests/responses to ARI as well |
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20:38.11 | seanbright | it's going to come down to creating and then dialing rather than just originating |
20:38.14 | seanbright | that is my theory |
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23:20.20 | grummund | Anyone know if there's a way to control the DND state of a phone (Cisco SPA303) ? |
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23:43.35 | Samot | grummund: the DND button? |
23:44.39 | grummund | as if user pressed the DND button, but to do it from asterisk. |
23:45.02 | grummund | say to automatically set DND for outside office hours. |
23:45.39 | Samot | You need to program that |
23:46.11 | grummund | can it even be done? |
23:46.11 | Samot | DND sets the phone to return a busy |
23:46.17 | Samot | Of course |
23:47.27 | grummund | and the user can continue to toggle DND for themselves to override it? |
23:47.55 | grummund | and it shows up as DND on the phone? |
23:48.26 | grummund | ...or if it can't be done, then what would be the closest approximation to this? |
23:51.19 | Samot | You can make star codes for that |
23:51.31 | Samot | You can make a GUI too |
23:52.23 | Samot | You would need a BLF on the phone |
23:52.43 | Samot | The phone doesnt know what the PBX does and vice versa |
23:56.11 | grummund | Without any specific config there appear to be some dialplan hints registered already. |
23:56.43 | grummund | would these be hardcoded ? - http://paste.debian.net/1180028/ |
23:57.24 | Samot | Yes |
23:57.33 | Samot | Unless youre using realtime |
23:57.55 | file | "dialplan show" will show you the loaded dialplan including hints, and tell you from what PBX module it originates |