| 00:01.22 | *** join/#asterisk sinaowolabi (~Sina@169.159.64.192) |
| 00:26.59 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 01:22.56 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 01:32.47 | *** join/#asterisk hfb (~hfb@47.139.16.87) |
| 01:44.40 | *** join/#asterisk EmleyMoor (42b789682f@firthpark.tinsleyviaduct.com) |
| 01:46.00 | *** join/#asterisk EmleyMoor (42b789682f@firthpark.tinsleyviaduct.com) |
| 01:50.12 | *** join/#asterisk EmleyMoor (42b789682f@firthpark.tinsleyviaduct.com) |
| 02:41.40 | *** join/#asterisk tsal_ (~tsal@i59F52EDE.versanet.de) |
| 03:09.21 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 03:42.44 | *** join/#asterisk electronic_eel (~quassel@213.240.182.76) |
| 03:43.25 | *** join/#asterisk FH_thecat (~FH_thecat@75.11.25.212.ftth.as8758.net) |
| 03:49.54 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 04:29.21 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 04:29.40 | *** join/#asterisk FH_thecat (~FH_thecat@75.11.25.212.ftth.as8758.net) |
| 04:50.43 | *** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net) |
| 05:24.28 | *** join/#asterisk jkroon (~jkroon@165.16.203.106) |
| 06:11.47 | *** join/#asterisk netman (~netman@185.94.249.222) |
| 06:29.49 | *** join/#asterisk jkroon (~jkroon@165.16.203.106) |
| 06:33.33 | *** join/#asterisk Jesterboxboy (~Thunderbi@84-115-150-8.cable.dynamic.surfer.at) |
| 06:42.07 | *** join/#asterisk dgt (~dgt@mysterio.trickey.uk) |
| 06:42.20 | *** join/#asterisk dgt (~dgt@mysterio.trickey.uk) |
| 06:42.32 | *** join/#asterisk dgt (~dgt@mysterio.trickey.uk) |
| 06:43.13 | *** join/#asterisk dgt (~dgt@mysterio.trickey.uk) |
| 06:50.05 | *** join/#asterisk TriJetScud (~TriJetScu@S01060418d6314a02.vc.shawcable.net) |
| 06:57.56 | *** join/#asterisk Dovid (~dovid@38.105.7.34) |
| 07:05.18 | *** join/#asterisk Dovid (~dovid@38.105.7.34) |
| 07:08.29 | *** join/#asterisk akp55_ (~akp55@c-73-148-15-158.hsd1.va.comcast.net) |
| 07:15.09 | *** join/#asterisk opal (~wowaname@volatile/founder/wowaname) |
| 07:34.42 | *** join/#asterisk maximCH (~maximCH@mail.swill.org) |
| 09:14.49 | *** join/#asterisk TriJetScud (~TriJetScu@S01060418d6314a02.vc.shawcable.net) |
| 09:21.16 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 10:03.01 | *** join/#asterisk Dovid (~dovid@38.105.7.34) |
| 10:08.17 | *** join/#asterisk Dovid (~dovid@129.250.201.4) |
| 10:57.53 | *** join/#asterisk spatel (~spatel@pool-96-237-230-175.bstnma.fios.verizon.net) |
| 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. |
| 12:34.04 | *** join/#asterisk mrtnt (~martint@martint.data.ee) |
| 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 |
| 13:18.07 | *** join/#asterisk Dovid (~dovid@ool-4356e81f.dyn.optonline.net) |
| 13:25.48 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 13:44.16 | *** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net) |
| 14:19.06 | *** join/#asterisk laerus (~kon@dslb-094-220-142-040.094.220.pools.vodafone-ip.de) |
| 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 |
| 14:52.30 | *** join/#asterisk dgt (~dgt@mysterio.trickey.uk) |
| 14:53.08 | *** join/#asterisk dgt (~dgt@mysterio.trickey.uk) |
| 14:54.03 | *** join/#asterisk kharwell (uid358942@gateway/web/irccloud.com/x-zxrxyhtxqkjemnzu) |
| 14:54.03 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 15:07.01 | *** join/#asterisk laerus (~kon@dslb-094-220-142-040.094.220.pools.vodafone-ip.de) |
| 15:10.03 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 15:11.59 | *** join/#asterisk giesen (~ggiesen@ego.giesen.me) |
| 15:17.03 | *** join/#asterisk gavlee (~gav@unaffiliated/gavlee) |
| 15:19.15 | *** join/#asterisk bford (uid283514@gateway/web/irccloud.com/x-fhwzlfjxupvocyiy) |
| 15:19.16 | *** mode/#asterisk [+o bford] by ChanServ |
| 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? |
| 15:33.36 | *** join/#asterisk post-factum (~post-fact@vulcan.natalenko.name) |
| 15:45.00 | *** join/#asterisk edgars (~edgars@w.dtg.lv) |
| 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. |
| 16:22.53 | *** join/#asterisk Kernkraft4Ever (~quassel@2001:16b8:c155:8700:1404:512b:68eb:aaa1) |
| 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 |
| 16:37.33 | *** join/#asterisk EmleyMoor (42b789682f@firthpark.tinsleyviaduct.com) |
| 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 |
| 16:38.19 | *** join/#asterisk ckb (~ckb@unaffiliated/ckb) |
| 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 |
| 16:41.48 | *** join/#asterisk mboehn_ (mathias@hh1.nuxis.org) |
| 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 |
| 17:02.18 | *** join/#asterisk akp55 (~akp55@c-73-148-15-158.hsd1.va.comcast.net) |
| 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* |
| 17:13.52 | *** join/#asterisk Isaac (~imcdona@2602:41:642e:a610:e081:f6d8:2fa5:b868) |
| 17:35.09 | *** join/#asterisk n0tiz (~n0tiz@82-69-15-38.dsl.in-addr.zen.co.uk) |
| 17:39.44 | *** join/#asterisk Kernkraft4Ever (~quassel@2001:16b8:c155:8700:1404:512b:68eb:aaa1) |
| 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 |
| 18:03.58 | *** join/#asterisk hfb (~hfb@47.139.21.141) |
| 18:22.39 | *** join/#asterisk clarjon1 (~clarjon1@unaffiliated/clarjon1) |
| 18:27.21 | *** join/#asterisk Kritnich (~Kritnich@kritni.ch) |
| 18:27.31 | *** join/#asterisk hfb (~hfb@47.139.16.87) |
| 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 |
| 19:11.09 | *** join/#asterisk sa02irc (~mbax@155-079-043-212.ip-addr.inexio.net) |
| 19:25.19 | *** join/#asterisk ^MillerBoss (~biffies@gives.you.more.taste.at.only.96.calories.millerboss.com) |
| 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 |
| 20:37.53 | *** join/#asterisk hfb (~hfb@47.139.22.162) |
| 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 |
| 20:56.51 | *** join/#asterisk hfb (~hfb@47.139.16.87) |
| 20:59.21 | *** join/#asterisk Kernkraft4Ever (~quassel@2001:16b8:c155:8700:1404:512b:68eb:aaa1) |
| 21:17.39 | *** join/#asterisk hfb (~hfb@47.139.22.167) |
| 21:27.03 | *** join/#asterisk hfb (~hfb@47.139.22.167) |
| 21:49.53 | *** join/#asterisk hfb (~hfb@47.139.22.167) |
| 21:57.38 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 22:20.53 | *** join/#asterisk DodgeThis (~DodgeThis@246.102.90.149.rev.vodafone.pt) |
| 22:32.08 | *** join/#asterisk hfb (~hfb@47.139.22.162) |
| 23:02.35 | *** join/#asterisk hfb (~hfb@47.139.22.167) |
| 23:11.52 | *** join/#asterisk drathir_tor (~drathir@gateway/tor-sasl/drathir) |
| 23:20.20 | grummund | Anyone know if there's a way to control the DND state of a phone (Cisco SPA303) ? |
| 23:34.56 | *** join/#asterisk hfb (~hfb@47.139.16.87) |
| 23:41.50 | *** join/#asterisk imcdona (~imcdona@65-100-46-162.dia.static.qwest.net) |
| 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 |