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| 03:31.14 | igcewieling | stays perfectly still and listens for the "h323....h323....h323...." call of the dangerous creature called a Time Sucker |
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| 05:23.02 | igcewieling | a Chupahora |
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| 15:48.38 | Kobaz | so, this is weird. https://dpaste.com/7R6RYWTNS |
| 15:48.49 | Kobaz | WaitForRing() never gets the ring |
| 15:51.02 | file | ring and ringing are two separate things, ring is from analog land if I recall |
| 15:51.23 | Kobaz | Aaah |
| 15:51.53 | Kobaz | oddly enough this works in older asterisk's |
| 15:52.19 | Kobaz | I changed the flow to use SendDTMF and WaitDigit, works well |
| 15:53.06 | Kobaz | oh wait, that's not the same one i used in 11 |
| 15:53.17 | Kobaz | Okay so, the analog thing makes sense |
| 15:53.33 | Kobaz | good to know |
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| 16:54.08 | gregf | Hi, How can I make it so my extension ring X number of times before going to voice VoiceMailMessage? I haven't figured that out yet, very new. |
| 16:54.19 | gregf | I have it now so it just goes directly to voicemail I did figure that much out |
| 16:54.52 | gregf | on asterisk 16 btw |
| 16:59.42 | igcewieling | You can't. |
| 16:59.59 | igcewieling | You can, however, make it ring X number of seconds. |
| 17:00.53 | gregf | Ok, Well ring period? |
| 17:01.13 | gregf | I think I need to use Dial? |
| 17:04.15 | gregf | yep, it was dial... that took me way to long to figure out. |
| 17:05.50 | gregf | and Dail does take ring-time in seconds btw... |
| 17:06.24 | Kobaz | USA is 6 seconds per ring |
| 17:06.36 | gregf | Kobaz: ty |
| 17:06.37 | Kobaz | core hsow application Dial |
| 17:06.49 | Kobaz | you can give a timeout parameter to Dial() |
| 17:06.53 | Kobaz | *core show |
| 17:07.36 | Kobaz | In North America (excluding Mexico, Central America and parts of the Caribbean), the standard audible ringing tone is a repeated cadence of a two-second tone and four seconds of silence. The signal is composed of the frequencies 440 Hz and 480 Hz.[3] |
| 17:07.41 | Kobaz | https://en.wikipedia.org/wiki/Ringing_tone |
| 17:12.17 | igcewieling | That doesn't really apply to VoIP. |
| 17:12.32 | Kobaz | Hmm? |
| 17:13.33 | igcewieling | With analog, the ring cycle is done by the local CO, which is pretty standard. |
| 17:13.40 | Kobaz | right |
| 17:14.00 | igcewieling | With VoIP, it is handled by the remote phone, the local phone, the remote carrier, or the local carrier. |
| 17:14.34 | Kobaz | well haha yeah, but the ring cadance is typical |
| 17:14.46 | Kobaz | across those, (for sane systems) |
| 17:15.58 | igcewieling | *shrug* My ring cadence is Triiplet-Low or something like that. |
| 17:16.23 | Kobaz | that's for receiving calls |
| 17:16.24 | igcewieling | I don't think the user actually specified Ring or Ringback |
| 17:16.37 | Kobaz | right, i'm talking about ringback |
| 17:17.05 | igcewieling | I was referring mostly to ringing. |
| 17:20.56 | csavinovich_ | Hello all, I have a question: How can I call a function immediately on lifting up the handset? Been a long time since I've used the dialplan. Off the top of my head I know I would do it using AMI and I would capture the triggered event, something "line Up" me thinks. Will anyone know of a much simpler solution? |
| 17:21.12 | igcewieling | I'm weird. When making a call I care about how many seconds it takes to go to Voicemail. When receiving a call about the number of times the phone rings so I know how much time I have to answer it. |
| 17:21.37 | igcewieling | csavinovich_: analog or SIP? |
| 17:22.35 | igcewieling | I'm weird. When making a call I care about how many seconds it takes to go to Voicemail. When receiving a call about the number of times the phone rings so I know how many rings I have to answer it. |
| 17:24.17 | Kobaz | some people want 'rings' |
| 17:24.35 | csavinovich_ | <PROTECTED> |
| 17:24.59 | Kobaz | actually a lot of people want rings, every manager I talk to at customer sites who helps design the systems we install in terms of call flow. they say something like "after 4 rings, this needs to go to voicemail" |
| 17:25.35 | Kobaz | csavinovich_: you have an ATA like a cisco SPA or something like that, you need to set up the hot line feature, where picking up the handset dials an extension on the sip server |
| 17:25.59 | igcewieling | csavinovich_: If analog on Digium or Sangoma, that is configured in chan_dahdi.conf immediate=yes. |
| 17:26.22 | Kobaz | igcewieling: ..."most of the phones are actually analog attached to one of those boxes that do analog to digital"... |
| 17:26.33 | igcewieling | for SIP or analog connect to a SIP adapter, that is handled by the SIP device. Usually called "hotline" or "warmline". |
| 17:26.39 | Kobaz | yup |
| 17:30.16 | Samot | "After 4 rings" roughly equals a 24 second ringtime. |
| 17:30.46 | Kobaz | right |
| 17:30.54 | Kobaz | ...just replying to: <igcewieling> I'm weird. When making a call I care about how many seconds it takes to go to Voicemail. |
| 17:31.14 | Kobaz | the flip side is most office people I've run into count rings, not seconds |
| 17:31.53 | Samot | SIP systems do not do it in "rings" |
| 17:31.59 | Kobaz | We know this |
| 17:32.07 | Samot | So how you present that to the end user is up to you. |
| 17:32.16 | Samot | Yes, I have users that say "rings" |
| 17:33.04 | Kobaz | ringalingaling |
| 17:33.36 | Samot | I mainly service hotels, so having special dialing rules for various types of phones is common place. |
| 17:35.07 | Kobaz | It's always really interesting to get a feel for the use cases. We get involved in custom app builds and really complex routing and tracking systems, which is why I run into all this weird stuff I do |
| 17:35.30 | Samot | Along with "Comes from outside, ring one way", "comes from guest, ring another way", "comes from the outside doors, ring another way" |
| 17:35.44 | Kobaz | steps back and waits for Samot's response to "weird" |
| 17:35.59 | Samot | Shrug. |
| 17:36.01 | Kobaz | hehe |
| 17:36.13 | Samot | I've serviced all sorts of industries. |
| 17:36.21 | Samot | I'm use to special use cases. |
| 17:36.48 | Kobaz | Speaking of special, I may have found a DoS for Asterisk-16, requires AMI |
| 17:37.04 | Samot | I mean, University of Michigan and pretty much 80% of the universities in Michigan at one point. |
| 17:37.11 | Kobaz | Ah |
| 17:37.23 | igcewieling | Swap out the admin with a different one. AMI should not be exposed on the internet. 7-) |
| 17:37.28 | Samot | Hospitals, state buildings (reps/congress)..etc. etc. |
| 17:37.59 | Kobaz | igcewieling: i mean like, if someone has developer access via AMI, and sends requests in a certain way, it's more like accidental DoS |
| 17:39.15 | Kobaz | All I know so far is that I broke my asterisk pretty badly just by using cli and ami together. More details to follow once I get the info in place and I read up on the current security discussion process |
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| 18:06.52 | csavinovich_ | @igcewieling: It is an Adtran SIP device with analog phones connected to it. It seems to send out an off hook message, anyone knows how Asterisk receives this message? |
| 18:07.12 | igcewieling | show us the message it sends. |
| 18:07.47 | igcewieling | without hotline or wamrline setup, the Adtran won't send anything to the SIP server until it has collected all expected digits. |
| 18:09.03 | csavinovich_ | Don't know it yet :(... I guess I will have to wait until I receive the documentation for it. I would venture guess it will be a SIP message :( |
| 18:14.04 | igcewieling | um, Adtran has extensive documentation online. |
| 18:16.01 | csavinovich_ | yes, I will be receiving some logs soon. Thank you igcewieling, you cleared it up :) |
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| 18:28.00 | igcewieling | Just ssh/telnet into it or access it with a web browser. you can't do what you want to do without access to the device. |
| 18:33.30 | Kobaz | Is it normal that module reload res_pjsip.so blocks new calls? |
| 18:33.36 | Kobaz | about 800 endpoints |
| 18:34.04 | Samot | Havent seen it and I got more then 1000 |
| 18:34.26 | Kobaz | for me it will not process a new call until the reload is done |
| 18:34.44 | Samot | How long does that take? |
| 18:34.57 | Kobaz | about 5 seconds |
| 18:35.01 | Kobaz | sometimes 10 |
| 18:35.27 | Samot | I dont think it should take that long. |
| 18:35.39 | Samot | Id have to test |
| 18:40.06 | Samot | Yeah, it was basically instant. |
| 18:40.24 | Kobaz | hmm |
| 18:40.41 | Kobaz | let me compare reading plain text vs extconfig |
| 18:41.04 | Samot | Oh you're using realtime? |
| 18:41.15 | Kobaz | semi realtime |
| 18:41.30 | Kobaz | it's loading a static config by selecting from a db function |
| 18:41.32 | Samot | You have stuff in the database you're calling with realtime, yes? |
| 18:42.00 | Samot | So reloading res_pjsip.so is loading data from a database? |
| 18:42.10 | Kobaz | correctr |
| 18:42.30 | Samot | So you're using the pjsip_endpoints via realtime? |
| 18:42.34 | Kobaz | no |
| 18:42.42 | Kobaz | it references them in memory once it's loaded |
| 18:43.05 | Samot | So how is reloading res_pjsip.so calling from a database? |
| 18:43.11 | Kobaz | via extconfig |
| 18:43.26 | *** topic/#asterisk by gtjoseph -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 18.2.2, 16.16.2 (2021/03/04) Final Bugfix: 13.38.2, 17.9.3 (2021/03/04); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 18:43.28 | Kobaz | [settings] pjsip_wizard.conf => pgsql,pbx,asterisk.asterisk_exten_config('PJSIP_WIZARD') |
| 18:43.55 | Samot | So yes, realtime. |
| 18:44.29 | Samot | I use static only, so it was almost instant. |
| 18:45.06 | Kobaz | # explain analyse select * from asterisk.asterisk_exten_config('PJSIP_WIZARD'); Execution Time: 241.959 ms |
| 18:45.25 | Kobaz | so, that should be the only overhead |
| 18:47.07 | Samot | Well can't really help. realtime never had an impactful benefit for me. |
| 18:47.26 | Kobaz | i just like not relying on real files |
| 18:47.44 | Kobaz | but, something I did notice, is that the query is run 10 times, during the cource of the reload |
| 18:47.53 | Kobaz | *course |
| 18:48.03 | Kobaz | so that's going to add 2.4 seconds to whatever else it's doing |
| 18:48.07 | Samot | Then you wouldn't be "semi-realtime" |
| 18:48.17 | Kobaz | yeah |
| 18:48.27 | Kobaz | In Asterisk-11 it was the same as loading a static file |
| 18:48.30 | Kobaz | it would query it once |
| 18:48.31 | Samot | See realtime has a performance cost. |
| 18:48.45 | Samot | I don't think that cost is worth it. |
| 18:48.55 | Kobaz | i'm going to run this through my exporter and make a static file and compare |
| 18:49.10 | file | and if you were using chan_sip it may do the same, but chan_sip is not sorcery and PJSIP, and thus is different |
| 18:49.12 | Kobaz | Well, this was never al issue, even with 2000 endpoints, it would be, instant, like you said |
| 18:49.15 | Samot | I'm reloading 100% more endpoints at a fraction of what you are doing. |
| 18:49.20 | Kobaz | file: yeah I'm finding that out |
| 18:49.49 | Kobaz | Samot: i essentially have a ~200ms reload time using this method and chan_sip, so i never had a performance issue |
| 18:51.53 | Kobaz | hence, never having to do anything differently |
| 18:52.04 | Kobaz | But yeah, I might need to change this around then |
| 18:53.21 | Kobaz | here's the issue i'm having |
| 18:53.43 | Kobaz | res_pjsip is taking so lone to reload, and while it's loading it hangs onto a lock on module_list |
| 18:54.05 | Kobaz | and then I have other things in AMI that check for certain thigns on startup, like which modules are loaded, and then they block |
| 18:54.25 | Samot | I'm just going to say moving from chan_sip to chan_pjsip is not a parallel move. |
| 18:54.26 | Kobaz | and then other things block like pj subscription lists, and then everything grinds to a halt sometimes |
| 18:54.32 | Kobaz | Samot: of course |
| 18:54.40 | Kobaz | I'm just saying why i never had performance issues |
| 18:55.30 | Samot | Well I doubt that. Just at a different level. |
| 18:55.42 | Kobaz | ? |
| 18:55.44 | Samot | But yeah, you're going to be hunting these things down for a bit. |
| 18:55.52 | Samot | chan_sip had limitations. |
| 18:55.54 | Kobaz | What level would that be? |
| 18:55.55 | Kobaz | Well yeah |
| 18:56.04 | Samot | It can't handle the number of sessions pjsip can. |
| 18:56.18 | Kobaz | fair enough, I never had that high of a session load |
| 18:56.18 | Samot | Or with the CPU/memory performance pjsip can |
| 18:56.20 | file | chan_sip was a known factor, chan_pjsip isn't |
| 18:57.11 | Kobaz | Yeah, I know our own caveats of chan_sip very well, and can easily design around that |
| 18:57.27 | Kobaz | Samot: yup, reloading 800 endpoints with a static file is, almost instand |
| 18:57.31 | Samot | I think part of this is the fact that there were so many kludges for chan_sip performance they became standard. |
| 18:57.38 | Kobaz | and no long-term locks being held |
| 18:58.06 | Samot | So people by default were already accounting for chan_sip issues when doing setups because that's just what had to be done. |
| 18:58.08 | seanbright | so just use chan_sip |
| 18:58.16 | Kobaz | allrightey, refactoring here we go |
| 18:58.26 | Kobaz | seanbright: yes! that's the answer! </kidding> |
| 18:58.49 | seanbright | is it not? |
| 18:59.11 | Kobaz | At least I have all the hooks already for extension changes and writing a file would be literally adding one line of code |
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| 19:42.13 | TandyUK | hey guys, Im interested which android apps you use for sip calls, and maybe why.. (and yes i know theres like a billion writeups i could google) |
| 19:42.27 | Kobaz | either linphone or cloud soft phone |
| 19:44.05 | TandyUK | do you know if they integrate properly when using bluetooth? (ie while driving - auto answer and/or voice dial, vi the bluetooth device) |
| 19:44.20 | Kobaz | no idea |
| 19:44.26 | Kobaz | linphone is free, you can try it |
| 19:44.36 | TandyUK | fair enough :) |
| 19:45.12 | TandyUK | it bugs the hell out of me that samsung can nerf the native sip functionality and google let them |
| 19:45.30 | TandyUK | https://support.google.com/phoneapp/answer/2811843? << Thats how hard it _should_ be lol |
| 20:13.41 | Samot | It's an open source OS |
| 20:13.53 | Samot | By that sheer fact, Google let them. |
| 20:14.24 | Samot | And honestly, everything I've read on the native SIP stack in Android ain't great. |
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| 21:28.52 | TandyUK | Samot: at least it worked in the past though, without samsung causing me to jump through hoops to undo whatever it is theyve done to stop it working |
| 21:29.12 | TandyUK | serves me right for buying a samsung device i guess |
| 21:33.42 | Samot | I'm sure Samsung is just one of the manufacturers who disable the native SIP dialer. |
| 21:34.12 | Samot | Who knows, depending on the carrier they could request it be disabled. |
| 21:35.02 | Samot | Samsung+ATT != Samsung+T-Mobile when buying from ATT or T-Mobile. |
| 21:38.59 | igcewieling | *shrug* Use Bria, for a softphone it is pretty good. |
| 21:50.08 | TandyUK | fwiw, this is on a bought-direct-from-samsung phone, which I was hoping would not have such nerfs |
| 21:50.29 | TandyUK | but yeah, playing with various softphone apps is what im doing now :) |
| 21:51.00 | TandyUK | C Sip Simple i think was the last android one I used |
| 22:05.58 | Kobaz | oh |
| 22:06.05 | Kobaz | random question. account.$reg_num.codec.opus.para= |
| 22:06.08 | Kobaz | what's para ? |
| 22:06.12 | Kobaz | ...for yealink phones |
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| 23:03.27 | Kobaz | i broke it |
| 23:03.37 | Kobaz | https://dpaste.com/5T7K32CMM |
| 23:07.53 | Kobaz | guess i'll need more core debug |
| 23:26.47 | Kobaz | aaaaah |
| 23:26.50 | Kobaz | you need to match dialplan |
| 23:26.55 | Kobaz | i added debugging to res_parking |
| 23:27.06 | Kobaz | ERROR[9793][C-0000003a]: Parking lot 'c10000' -- Failed to find extension '701' in context 'parkedcalls_c10000' |
| 23:27.21 | Kobaz | i guess you need to populate the context with your parking destinations ahead of time |
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