00:25.27 | FuriousGeorge | Samot: ill word on them. so far none want sip clients |
00:25.36 | FuriousGeorge | s/word/work |
00:27.22 | Samot | OK, I'll bite. Why don't they want SIP clients? |
00:30.09 | FuriousGeorge | Samot: 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) |
00:32.51 | Samot | OK well using SIP clients (and decent ones) would resolve most of what you need to do. |
00:32.52 | *** join/#asterisk FuriousGeorge5 (47bb0404@pool-71-187-4-4.nwrknj.fios.verizon.net) |
00:32.57 | Samot | OK well using SIP clients (and decent ones) would resolve most of what you need to do. |
00:33.20 | FuriousGeorge5 | i know you are right. ill use one, and ill work on them |
00:33.35 | Samot | No need for them to dial in, enter a super secret voicemail box and hope no one finds it... |
00:33.43 | Samot | They just hit the voicemail button on the client. |
00:34.19 | FuriousGeorge5 | though ill probably primarily use a desk phone and forward to my mobile |
00:34.33 | Samot | OK. |
00:34.42 | FuriousGeorge5 | ive used bria et al, and i still don't like having mobile clients |
00:35.04 | FuriousGeorge5 | when they get sick of running everything thru their mobile numbers, they'll come around |
00:35.15 | Samot | No, instead it sounds like a more complicated setup and still have gaps. |
00:35.24 | FuriousGeorge5 | right now it's like explaining color to the blind |
00:35.50 | FuriousGeorge5 | ceretaily more complicated |
00:35.58 | FuriousGeorge5 | working on gaps |
00:36.06 | Samot | You're actually introducing problems you then need to solve. |
00:36.25 | FuriousGeorge5 | it's like 150 lines or something |
00:37.14 | FuriousGeorge5 | 1/3 of the people wouldnt know a sip phone from a marlin anyway |
00:37.18 | FuriousGeorge5 | maybe 40% |
00:37.53 | FuriousGeorge5 | im kinda glad i dont have to teach them attended xfer |
00:37.58 | Samot | OK so do they pay for usage? |
00:38.19 | FuriousGeorge5 | prorated and we get few calls, so that's negligble |
00:38.30 | Samot | Do they pay per minute on inbound and outbound? |
00:38.34 | FuriousGeorge5 | yeah |
00:38.44 | Samot | OK so now every call is 2X the cost. |
00:39.08 | Samot | You've also created them a DISA based PBX system. That has other concerns to go with it. |
00:39.16 | FuriousGeorge5 | yeah. i can't chain them to a chair clovkwork orange style and make them agree with me |
00:39.18 | Samot | You're exposing their internal voicemail to external IVR systems |
00:39.26 | FuriousGeorge5 | i had to fight for individual voicemail |
00:39.29 | Samot | I never get that |
00:39.44 | FuriousGeorge5 | i'll get a separate did for that |
00:39.44 | Samot | You are hired to provide your expertise and do a job... |
00:39.54 | Samot | Yet no one wants to listen to you? |
00:39.57 | FuriousGeorge5 | no, i'm not, asterisk is a fraction of what i do |
00:40.02 | Samot | OK. |
00:40.17 | FuriousGeorge5 | i don't want to spend my capital on this |
00:40.25 | FuriousGeorge5 | i always have "i told you so" |
00:40.29 | FuriousGeorge5 | and they will see the light |
00:40.35 | Samot | What capital? |
00:40.42 | Samot | They are the client, they aer paying for this. |
00:40.59 | FuriousGeorge5 | political capital. it's an esoteric expression. didn't mean to be vague |
00:41.13 | FuriousGeorge5 | no they are partners, and yes they are paying |
00:41.28 | FuriousGeorge5 | im not here because they needed a small pbx |
00:41.30 | Samot | Well good luck. |
00:41.34 | FuriousGeorge5 | ty |
00:45.52 | FuriousGeorge5 | i will need it. lol. |
00:47.25 | FuriousGeorge5 | one of them sends me texts messages of websites and expects me to punch in the url |
00:48.24 | FuriousGeorge5 | from 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? |
00:49.30 | Samot | Yes. |
00:50.07 | Samot | It's not a setting available. It's set in the [general] section of the voicemail.conf file. |
00:52.57 | FuriousGeorge | Samot: thanks! |
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01:03.11 | drmessano | Why does it seem like this PBX is gonna end up on a tech news site at some point? |
01:04.52 | FuriousGeorge | i 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 |
01:05.17 | FuriousGeorge | i forgot to mention, they'd have to guess the DID before they do all that |
01:05.36 | FuriousGeorge | because the did is not the same as the one that's advertised |
01:07.35 | FuriousGeorge | Samot: 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 |
01:09.54 | Samot | ; Max number of failed login attempts |
01:09.54 | Samot | maxlogins=3 |
01:12.53 | FuriousGeorge | im looking for a way to hang up on the person if the exceed maxlogins |
01:13.28 | FuriousGeorge | not 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 |
01:13.40 | Samot | ${VMSTATUS} * voicmail() |
01:13.40 | Samot | ${VMBOXEXISTSSTATUS} * vmboxexists() |
01:13.54 | Samot | You realize people have been making PBXes that do this for years now, right? |
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01:14.33 | Samot | I mean Asterisk based PBX systems, specifically. |
01:16.17 | FuriousGeorge | do they end up on tech sites? |
01:16.32 | FuriousGeorge | couldn't help it :P |
01:37.18 | FuriousGeorge | Samot: 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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02:06.38 | igcewieling | https://pastebin.com/dKPc6Y8G in case it is useful to anyone, this is my FreePBx/Asterisk voicemail check script. |
02:06.48 | igcewieling | it was a quick hack, but works for me. |
02:09.52 | igcewieling | Inspired by https://www.datagenetics.com/blog/september32012/ |
02:10.58 | Samot | I think he's trying to get what happens in voicemailmain when trying to access the mailbox with a password. |
02:11.17 | Samot | That I don't think has a status, it just kicks them out of the voicemail app and back into the dialplan. |
02:11.32 | igcewieling | I know. my post is unrelated to what he was referring to. |
02:15.00 | igcewieling | The 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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02:24.00 | Samot | Well this is basically a DISA system for whatever reason. |
02:24.05 | Samot | No endpoints. |
02:24.30 | Samot | So just making more work than needed. It's basically because voicemail access will be open to those dialing in. |
02:24.46 | Samot | And I'm sure it's to track if someone is trying to "hack it" |
02:25.05 | drmessano | So um |
02:25.10 | drmessano | In 1980 |
02:25.15 | drmessano | I could get a page |
02:25.24 | drmessano | call a number, retrieve a message |
02:25.30 | drmessano | So this system is 1980? |
02:25.59 | Samot | The 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. |
02:26.14 | Samot | they must also dial a DID to access the voicemail system. |
02:26.27 | igcewieling | Ah! He is setting up a system for failure! |
02:26.56 | Samot | Inbound calls to the "extensions" will forward to cell phones, that was his other issue. |
02:27.15 | Samot | Can't modify CNAM with prefixes cuz the other carrier is overriding it |
02:27.56 | Samot | Doesn't know how to handle when they don't answer these calls.. |
02:28.31 | Samot | 9:03:12 PM <drmessano> Why does it seem like this PBX is gonna end up on a tech news site at some point? |
02:28.41 | Samot | igcewieling: ^^^ that |
02:29.12 | igcewieling | *nod* |
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07:59.06 | qermit | hi, 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. |
08:01.23 | qermit | I'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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08:57.46 | FuriousGeorge | hi 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 |
08:59.03 | FuriousGeorge | nm, dumb question, found ${INVALID_EXTEN} |
08:59.15 | sibiria | we 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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10:37.09 | vlt | would use _X! |
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12:49.27 | retentiveboy | Working 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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12:49.45 | retentiveboy | pjsip show identifies reports no objects found. |
12:50.10 | retentiveboy | restart and it's back. |
12:50.55 | retentiveboy | meaning, show identities reports the peers after the restart and all is working fine. |
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15:06.20 | Dovid | Is there any reason why max_pseudo_channels defaults to 512 |
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15:15.50 | igcewieling | Dovid: you could try #asterisk-dev. I have a few guesses, but they are only guesses. |
15:15.56 | Dovid | t |
15:15.57 | Dovid | ty* |
15:26.48 | jkroon | Dovid, most likely because nobody envisioned requiring more. try updating the code, and if possible, make it dynamic (ie, no compiled max) |
15:27.26 | jkroon | then send in for code review |
15:30.52 | igcewieling | maybe rethink the need for > 512 pseudo channels. |
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18:28.14 | igcewieling | *grumble* This COVID-19 thing is going to be the death of privacy. |
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18:37.54 | drmessano | How so? |
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19:13.37 | Samot | Oh...yeah, please explain igcewieling. Something you saw/experienced? |
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19:50.26 | wonderworld | hey, there used to be that sip flow visualizing ncurses tool. sadly i forgot the name. |
19:50.59 | file | sngrep |
19:52.20 | wonderworld | yes, thats it. thank you. |
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20:17.45 | verzo | hi here |
20:18.21 | verzo | I try to migrate from chan_sip to res_pjsip on ast16 platform |
20:19.19 | verzo | after converting sip.conf to pjsip.conf by sip_to_pjsip.py script I can't get it working |
20:20.22 | verzo | in asterisk console I receive an error while loading res_pjsip |
20:20.41 | verzo | loading res_pjsip.so |
20:22.12 | verzo | sorcery.c:883 __ast_sorcery_object_type_insert_wizard: Wizard 'realtime' could not be applied to object type 'endpoint' as it was not found |
20:23.48 | verzo | sorcery.c:644 __ast_sorcery_open: Error attempting to apply configuration res_pjsip to sorcery. |
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20:30.48 | verzo | res_pjsip.c:5269 local_module: Failed to initialize SIP 'system' configuration section. Aborting load |
20:32.25 | verzo | I haven't touch extentions.ael after migrating to pjsip.conf, just try to have res_pjsip loaded with asterisk 16.5.1 |
20:32.49 | verzo | anybody faced the problem like I have? |
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20:45.29 | wk057 | Samot: just a note, after yesterdayâs convo I ended up settling on a setup where I have a BLF line button that has no function when pressed just for âpeople in queueâ 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 âfreeâ... then |
20:45.29 | wk057 | can get back to the parked call. |
20:49.34 | Samot | I see. |
20:50.39 | Samot | Basically what I said yesterday but using a parking lot to hold them. |
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