IRC log for #asterisk on 20200427

00:25.27FuriousGeorgeSamot:  ill word on them.  so far none want sip clients
00:25.36FuriousGeorges/word/work
00:27.22SamotOK, I'll bite. Why don't they want SIP clients?
00:30.09FuriousGeorgeSamot:  hard for me to speak for them, but i assume it's largely a combination of frame of reference (they have never used a mobile sip client)
00:32.51SamotOK well using SIP clients (and decent ones) would resolve most of what you need to do.
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00:32.57SamotOK well using SIP clients (and decent ones) would resolve most of what you need to do.
00:33.20FuriousGeorge5i know you are right.  ill use one, and ill work on them
00:33.35SamotNo need for them to dial in, enter a super secret voicemail box and hope no one finds it...
00:33.43SamotThey just hit the voicemail button on the client.
00:34.19FuriousGeorge5though ill probably primarily use a desk phone and forward to my mobile
00:34.33SamotOK.
00:34.42FuriousGeorge5ive used bria et al, and i still don't like having mobile clients
00:35.04FuriousGeorge5when they get sick of running everything thru their mobile numbers, they'll come around
00:35.15SamotNo, instead it sounds like a more complicated setup and still have gaps.
00:35.24FuriousGeorge5right now it's like explaining color to the blind
00:35.50FuriousGeorge5ceretaily more complicated
00:35.58FuriousGeorge5working on gaps
00:36.06SamotYou're actually introducing problems you then need to solve.
00:36.25FuriousGeorge5it's like 150 lines or something
00:37.14FuriousGeorge51/3 of the people wouldnt know a sip phone from a marlin anyway
00:37.18FuriousGeorge5maybe 40%
00:37.53FuriousGeorge5im kinda glad i dont have to teach them attended xfer
00:37.58SamotOK so do they pay for usage?
00:38.19FuriousGeorge5prorated and we get few calls, so that's negligble
00:38.30SamotDo they pay per minute on inbound and outbound?
00:38.34FuriousGeorge5yeah
00:38.44SamotOK so now every call is 2X the cost.
00:39.08SamotYou've also created them a DISA based PBX system. That has other concerns to go with it.
00:39.16FuriousGeorge5yeah. i can't chain them to a chair clovkwork orange style and make them agree with me
00:39.18SamotYou're exposing their internal voicemail to external IVR systems
00:39.26FuriousGeorge5i had to fight for individual voicemail
00:39.29SamotI never get that
00:39.44FuriousGeorge5i'll get a separate did for that
00:39.44SamotYou are hired to provide your expertise and do a job...
00:39.54SamotYet no one wants to listen to you?
00:39.57FuriousGeorge5no, i'm not, asterisk is a fraction of what i do
00:40.02SamotOK.
00:40.17FuriousGeorge5i don't want to spend my capital on this
00:40.25FuriousGeorge5i always have "i told you so"
00:40.29FuriousGeorge5and they will see the light
00:40.35SamotWhat capital?
00:40.42SamotThey are the client, they aer paying for this.
00:40.59FuriousGeorge5political capital.  it's an esoteric expression.  didn't mean to be vague
00:41.13FuriousGeorge5no they are partners, and yes they are paying
00:41.28FuriousGeorge5im not here because they needed a small pbx
00:41.30SamotWell good luck.
00:41.34FuriousGeorge5ty
00:45.52FuriousGeorge5i will need it.  lol.
00:47.25FuriousGeorge5one of them sends me texts messages of websites and expects me to punch in the url
00:48.24FuriousGeorge5from what i'm reading, looks like there's no way to make voicemailmain hangup if the person enters the wrong password too many times.  i'm not missing anything there, am i?
00:49.30SamotYes.
00:50.07SamotIt's not a setting available. It's set in the [general] section of the voicemail.conf file.
00:52.57FuriousGeorgeSamot:  thanks!
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01:03.11drmessanoWhy does it seem like this PBX is gonna end up on a tech news site at some point?
01:04.52FuriousGeorgei suppose it's cuz you want to believe that.  to get to voicemail they'd have to know the extension, know the CID of an authorized user, spoof that CID, and then guess their password
01:05.17FuriousGeorgei forgot to mention, they'd have to guess the DID before they do all that
01:05.36FuriousGeorgebecause the did is not the same as the one that's advertised
01:07.35FuriousGeorgeSamot:  it looks like this ability is dependent on writing an external script to handle vm passwords.  it doesn't look like there's a way to natively have asterisk hangup after attempts are exhausted
01:09.54Samot; Max number of failed login attempts
01:09.54Samotmaxlogins=3
01:12.53FuriousGeorgeim looking for a way to hang up on the person if the exceed maxlogins
01:13.28FuriousGeorgenot to change the max.  the dialplan continues and there seems to be no way to branch if the user fails or succeeds to log in
01:13.40Samot${VMSTATUS} * voicmail()
01:13.40Samot${VMBOXEXISTSSTATUS} * vmboxexists()
01:13.54SamotYou realize people have been making PBXes that do this for years now, right?
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01:14.33SamotI mean Asterisk based PBX systems, specifically.
01:16.17FuriousGeorgedo they end up on tech sites?
01:16.32FuriousGeorgecouldn't help it :P
01:37.18FuriousGeorgeSamot:  doesn't seem like it's possible to do with VoiceMailMain.  i just did a chandump after purposefully failing to log in and also noop'ed vmstatus.  nothing to indicate that authentication failed
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02:06.38igcewielinghttps://pastebin.com/dKPc6Y8G in case it is useful to anyone, this is my FreePBx/Asterisk voicemail check script.
02:06.48igcewielingit was a quick hack, but works for me.
02:09.52igcewielingInspired by https://www.datagenetics.com/blog/september32012/
02:10.58SamotI think he's trying to get what happens in voicemailmain when trying to access the mailbox with a password.
02:11.17SamotThat I don't think has a status, it just kicks them out of the voicemail app and back into the dialplan.
02:11.32igcewielingI know.  my post is unrelated to what he was referring to.
02:15.00igcewielingThe best he can hope for is to use the "s" option and handle authentication in the dialplan, that will allow tracking the number of failed / successful login attempts.
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02:24.00SamotWell this is basically a DISA system for whatever reason.
02:24.05SamotNo endpoints.
02:24.30SamotSo just making more work than needed. It's basically because voicemail access will be open to those dialing in.
02:24.46SamotAnd I'm sure it's to track if someone is trying to "hack it"
02:25.05drmessanoSo um
02:25.10drmessanoIn 1980
02:25.15drmessanoI could get a page
02:25.24drmessanocall a number, retrieve a message
02:25.30drmessanoSo this system is 1980?
02:25.59SamotThe end users will call from the cell, it will figure out who they are (I guess), mask the callerid and let them dial a destination.
02:26.14Samotthey must also dial a DID to access the voicemail system.
02:26.27igcewielingAh!  He is setting up a system for failure!
02:26.56SamotInbound calls to the "extensions" will forward to cell phones, that was his other issue.
02:27.15SamotCan't modify CNAM with prefixes cuz the other carrier is overriding it
02:27.56SamotDoesn't know how to handle when they don't answer these calls..
02:28.31Samot9:03:12 PM <drmessano> Why does it seem like this PBX is gonna end up on a tech news site at some point?
02:28.41Samotigcewieling: ^^^ that
02:29.12igcewieling*nod*
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07:59.06qermithi, may I have question regarding monitor? I have problem with voice quality in both channel when one side starts sending voice frames in a bursts of four or more.
08:01.23qermitI'm not sure if code is broken (https://github.com/asterisk/asterisk/blob/master/main/channel.c#L4120) or have I missed something in asterisk settings (13.25)
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08:57.46FuriousGeorgehi all.  is it fair to say it is never a good idea to use _X.?  i'm looking to replace the invalid handler in one particular case, but "i" becomes the extension when you get there, so i lose what the user dialed
08:59.03FuriousGeorgenm, dumb question, found ${INVALID_EXTEN}
08:59.15sibiriawe use it as a catch-all for all our DIDs. the actual dial plans are context-based and no context will simply lead to a 404
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10:37.09vltwould use _X!
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12:49.27retentiveboyWorking on a new instance of Asterisk for my home office. I keep losing pjsip identifies. Restart and all is well then a day or so later, I notice "No matching endpoint found" in response to qualifies from a peer at another office.
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12:49.45retentiveboypjsip show identifies reports no objects found.
12:50.10retentiveboyrestart and it's back.
12:50.55retentiveboymeaning, show identities reports the peers after the restart and all is working fine.
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15:06.20DovidIs there any reason why max_pseudo_channels defaults to 512
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15:15.50igcewielingDovid: you could try #asterisk-dev.   I have a few guesses, but they are only guesses.
15:15.56Dovidt
15:15.57Dovidty*
15:26.48jkroonDovid, most likely because nobody envisioned requiring more.  try updating the code, and if possible, make it dynamic (ie, no compiled max)
15:27.26jkroonthen send in for code review
15:30.52igcewielingmaybe rethink the need for > 512 pseudo channels.
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18:28.14igcewieling*grumble* This COVID-19 thing is going to be the death of privacy.
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18:37.54drmessanoHow so?
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19:13.37SamotOh...yeah, please explain igcewieling. Something you saw/experienced?
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19:50.26wonderworldhey, there used to be that sip flow visualizing ncurses tool. sadly i forgot the name.
19:50.59filesngrep
19:52.20wonderworldyes, thats it. thank you.
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20:17.45verzohi here
20:18.21verzoI try to migrate from chan_sip to res_pjsip on ast16 platform
20:19.19verzoafter converting sip.conf to pjsip.conf by sip_to_pjsip.py script I can't get it working
20:20.22verzoin asterisk console I receive an error while loading res_pjsip
20:20.41verzoloading res_pjsip.so
20:22.12verzosorcery.c:883 __ast_sorcery_object_type_insert_wizard: Wizard 'realtime' could not be applied to object type 'endpoint' as it was not found
20:23.48verzosorcery.c:644 __ast_sorcery_open: Error attempting to apply configuration res_pjsip to sorcery.
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20:30.48verzores_pjsip.c:5269 local_module: Failed to initialize SIP 'system' configuration section. Aborting load
20:32.25verzoI haven't touch extentions.ael after migrating to pjsip.conf, just try to have res_pjsip loaded with asterisk 16.5.1
20:32.49verzoanybody faced the problem like I have?
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20:45.29wk057Samot: just a note, after yesterday’s convo I ended up settling on a setup where I have a BLF line button that has no function when pressed just for “people in queue” notification... and several parking lot buttons for calls that can wait a moment.  can park a call, which technically frees up the line, and then the queue will ring them within a few seconds of being “free”... then
20:45.29wk057can get back to the parked call.
20:49.34SamotI see.
20:50.39SamotBasically what I said yesterday but using a parking lot to hold them.
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