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| 06:31.43 | davidzag | hello anyone worked on whatsappcall with chan mobile ? |
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| 14:39.45 | Accord | hey guys, what is it called when a user types on the dialer during call? I know old telco bots used to do this |
| 14:40.31 | Accord | I want to implement something similar |
| 14:49.18 | Samot | You mean type words? |
| 14:49.28 | Samot | Can you clarify? |
| 14:57.49 | Accord | can't type words on the dialer, only numbers, # and * |
| 14:58.20 | Accord | I know bots would ask you to pick option 1 , 2, 3 , 4 and when you pressed that number on your dialer it would go to that next menu |
| 14:58.43 | Accord | so how does the bot know I pressed something? |
| 14:59.49 | *** join/#asterisk _ganapathi_ (~Ganapathi@157.50.117.222) |
| 15:00.37 | _ganapathi_ | how do i disable pjsip module in asterisk 16. |
| 15:06.19 | [TK]D-Fender | _ganapathi_, modules.conf <- |
| 15:06.39 | Samot | Why would you do that? |
| 15:06.46 | [TK]D-Fender | However you should make the transition to it if your intent was to use chan_sip instead... |
| 15:07.31 | _ganapathi_ | am using chan_sip. but i dont knw real usage of pjsip. so i dont required. so to gain performance by disabling |
| 15:07.42 | Accord | figured it out, it's called DTMF |
| 15:08.09 | seanbright | someone could have answered that for you if we understood what you were looking for |
| 15:08.12 | seanbright | it wasn't super clear |
| 15:08.26 | _ganapathi_ | i tried to disable pjsip through modules.conf but it's throws some error as dependency |
| 15:09.14 | _ganapathi_ | if i disable pjsip then rtp also disabled. but i required rtp on sip |
| 15:10.07 | seanbright | disabling pjsip does not disable rtp |
| 15:10.13 | seanbright | they are separate modules |
| 15:10.24 | seanbright | res_rtp_asterisk.so is the RTP engine implementation |
| 15:10.39 | seanbright | which does not know or care about chan_pjsip |
| 15:11.15 | Samot | There are very few reasons to say on Chan_SIP. |
| 15:12.00 | seanbright | the main reason people don't want to switch is that it requires learning something new |
| 15:12.00 | Samot | stay* |
| 15:12.35 | Samot | Well for me it's the lack of a full RFC 4235 support. |
| 15:12.46 | Samot | So there are some I have to leave on Chan_SIP. |
| 15:13.44 | Samot | Because the lack of RFC4235 makes BLF/Call Pickup a PITA. |
| 15:15.53 | seanbright | oh, there's an xml schema in this rfc... |
| 15:15.55 | seanbright | taps out |
| 15:16.21 | seanbright | does chan_sip support that RFC? |
| 15:17.14 | Samot | Fully. |
| 15:17.30 | seanbright | nifty |
| 15:17.41 | seanbright | that's surprising |
| 15:17.45 | seanbright | is CCSS a piece of that? |
| 15:19.42 | Samot | CCSS? |
| 15:19.53 | seanbright | call completion supplementary services |
| 15:20.17 | seanbright | s |
| 15:20.23 | seanbright | https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=5243096 |
| 15:20.25 | Samot | https://www.irccloud.com/pastebin/RBaEA46p/ |
| 15:20.48 | seanbright | so what doesn't PJSIP support exactly? |
| 15:20.53 | Samot | The <local></local> and the <remote></remote> sections are missing. |
| 15:20.58 | Samot | That's from Chan_SIP. |
| 15:21.23 | Samot | So with PJSIP I can see 100 is getting a call or that 100 is busy |
| 15:21.52 | Samot | But I can't see who is calling 100 and when I hit the BLF to pick it up, it doesn't have the 100 information for appending to the ** |
| 15:22.09 | Samot | So my phone still wants me to enter the extension to pick up.. |
| 15:22.33 | Samot | In Chan_SIP, I can see who is calling 100 and I can hit the BLF button and 100 gets appended to ** |
| 15:22.35 | seanbright | interesting |
| 15:23.05 | seanbright | i can implement it i guess for $15k or so |
| 15:23.10 | seanbright | let me know |
| 15:23.15 | seanbright | (heh) |
| 15:23.16 | Samot | There is RFC4235 support in PJSIP, it's just not a full implementation. |
| 15:23.33 | Samot | Which really does mess with BLF/Call Pickup support on numerous phones. |
| 15:25.05 | Samot | The end result, I have numerous offices still running on Chan_SIP while anyone else who doesn't need that feature/function is on PJSIP. |
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| 15:29.33 | file | I vaguely recall someone putting a patch on an issue but never putting it up on code review |
| 15:29.39 | Samot | Right. |
| 15:29.48 | Samot | Someone did and then never followed through as requested. |
| 15:31.52 | Samot | So that little piece is stopping me from fully moving to PJSIP. |
| 15:32.04 | seanbright | ok, ok... $14.5k |
| 15:32.12 | seanbright | but only because we're friends |
| 15:32.46 | Samot | Well I'm kind of hoping it gets added. |
| 15:33.11 | Samot | Otherwise I'm going to have to put something in place to deal with it. I don't want to stay on Chan_SIP much longer. |
| 15:35.15 | Samot | I'm going to guess it's also hampering others from a full migration to PJSIP as well. |
| 15:37.25 | Samot | And by that I mean numerous people in the FreePBX forums have complained about it and told by vendors such as Yealink, etc that the issue is with Asterisk lack of RFC support. |
| 15:38.21 | seanbright | i think we all understand what you are saying |
| 15:38.43 | Samot | So $14.5K? |
| 15:38.46 | seanbright | or you could restate it a 5th time with slightly different words |
| 15:38.52 | Samot | I could. |
| 15:39.00 | seanbright | by all means |
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| 15:42.15 | seanbright | if i can find some free time i might take a whack at it |
| 15:42.24 | seanbright | i'm sure it is non-trivial which is why it wasn't done in the first place |
| 15:43.11 | Samot | If I had any real experience in C I'd take a whack at it but I don't. |
| 15:43.34 | Samot | The XML schema is there with PJSIP, it's just not the full schema. |
| 15:52.41 | seanbright | right. so the pjsip module is just receiving an event and serializing it into dialog-info |
| 15:52.54 | seanbright | my guess is that it just doesn't have access to the remote party's info |
| 15:52.59 | seanbright | but i haven't look at it, so that is just a guess |
| 15:53.23 | Samot | https://community.asterisk.org/t/rfc4235-tags-and-attributes-in-pjsip-notify-event-dialog-xml-body/75775/4 |
| 15:53.46 | Samot | I'd try that but like I said, I have zero C knowledge and I don't trust myself to not break anything. |
| 15:54.45 | seanbright | i mean, have you tried just pasting that code into the module? or you don't feel comfortable with that? |
| 15:56.05 | Samot | There is that. |
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| 15:56.18 | seanbright | i can create a branch and add it for you to test |
| 15:56.22 | seanbright | what version of asterisk? |
| 15:56.29 | Samot | Plus it has the remote as a static 0 for the example. I wouldn't know how to get the remote party details for it. |
| 15:56.48 | seanbright | ah, right |
| 15:56.55 | _ganapathi_ | where i can find asterisk.spec file to build rpm |
| 15:56.57 | Samot | That is really my biggest hold up. |
| 15:57.15 | Samot | If I knew that I might try to patch a test box with it. |
| 15:59.25 | seanbright | it looks like you can get the remote uri (the <target> node i guess) but not the identity |
| 15:59.45 | seanbright | anyway, what version of asterisk are you running? |
| 16:00.11 | Samot | It's FreePBX based boxes so it's all RPM. That said, 16.3.0 |
| 16:04.49 | seanbright | https://github.com/seanbright/asterisk/commit/bd22c3a7fb105c90a0ebfd8a044d85e92500ceb5 |
| 16:04.58 | seanbright | so that is just with his patch integrated\ |
| 16:05.03 | seanbright | compiles for me |
| 16:05.59 | Samot | I take it the remote info would be in the state_data obj? |
| 16:06.41 | seanbright | https://github.com/seanbright/asterisk/commit/b358cb117216b323c9275f5dd0b569c72a8dc24d |
| 16:06.53 | seanbright | and that patch just prints the remote dialog uri to the log as an error |
| 16:07.00 | seanbright | so we can see if it is useful |
| 16:08.02 | seanbright | oh, and then there is this: https://issues.asterisk.org/jira/browse/ASTERISK-24601 |
| 16:08.25 | seanbright | which has a patch (11-dialog-info.patch) that does the heavy lifting |
| 16:08.56 | Samot | I think that maybe what file was referring to. |
| 16:09.22 | seanbright | yes |
| 16:09.34 | seanbright | so there is a patch there |
| 16:09.39 | seanbright | you could test it? |
| 16:13.51 | Samot | I can this weekend. |
| 16:14.04 | *** join/#asterisk alexandre9099 (~alexandre@unaffiliated/alexandre9099) |
| 16:14.07 | Samot | I'm looking at that patch file right now. |
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| 16:16.34 | Samot | So this should be fine even though it looks like a patch in 15.2.2? |
| 16:17.12 | Samot | Because I think this patch might do the trick. |
| 16:25.18 | seanbright | it's not the cleanest patch |
| 16:25.25 | seanbright | and it might not apply cleanly to 16 |
| 16:26.35 | seanbright | wget -O - https://issues.asterisk.org/jira/secure/attachment/58314/11-dialog-info.patch | patch -p1 |
| 16:26.42 | seanbright | that works for me with a fresh checkout of 16 |
| 16:26.44 | _ganapathi_ | Error loading module 'res_rtp_asterisk.so', missing dependency: res_pjproject |
| 16:27.04 | seanbright | _ganapathi_: then load res_pjproject.so |
| 16:27.16 | seanbright | you don't need the other pjsip modules |
| 16:28.15 | _ganapathi_ | then am getting some other error while loading res_pjproject |
| 16:28.21 | seanbright | which is? |
| 16:28.36 | seanbright | i'm going to guess that you aren't loading res_sorcery_* |
| 16:29.42 | _ganapathi_ | yes. |
| 16:29.54 | seanbright | ok, so load them. i think there are 4. |
| 16:30.00 | _ganapathi_ | need to enable all sorcery module ? |
| 16:30.10 | seanbright | yes. |
| 16:30.18 | seanbright | technically no... but yes |
| 16:31.26 | seanbright | so how does call pickup work when multiple callers are trying to reach a device? |
| 16:31.33 | _ganapathi_ | ok thnks. now rtp loaded. |
| 16:31.40 | seanbright | so A and B are both calling C and on D you want to pickup C |
| 16:32.38 | _ganapathi_ | so sorcery necessary on multiple caller calling SIP ?. |
| 16:32.50 | seanbright | _ganapathi_: no, i am talking about something else now |
| 16:33.09 | _ganapathi_ | ooh. |
| 16:35.02 | Samot | seanbright: Sorry was on a call. But it's the same as just doing **D |
| 16:35.35 | Samot | The phone reads the XML and prepends the callee's extension to the call pick command. |
| 16:35.54 | seanbright | right, but what if there are multiple incoming calls? do you get multiple soft keys? |
| 16:36.00 | Samot | No. |
| 16:36.08 | Samot | It just sends **<extension> |
| 16:36.24 | Samot | So it will grab the last channel that was presented. |
| 16:36.28 | seanbright | gotcha |
| 16:36.44 | Samot | I still have to tell the phone what the Call Pickup code is.. |
| 16:37.13 | Samot | The BLF functionality just prepends and sends it as one-touch. |
| 16:37.32 | seanbright | seems inflexible, but it is what it is |
| 16:37.37 | seanbright | i guess that situation arises rarely |
| 16:37.48 | Samot | Like I said, it's no different than dialing **100 |
| 16:37.55 | Samot | What does Asterisk do then? |
| 16:38.01 | seanbright | shrugs |
| 16:38.07 | seanbright | i've never used asterisk before |
| 16:38.14 | seanbright | where am i? |
| 16:38.23 | Samot | 3Cx. |
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| 18:28.57 | _ganapathi_ | res_rtp_asterisk is required for ? |
| 18:29.07 | seanbright | RTP |
| 18:29.42 | _ganapathi_ | chan_rtp.so - alone not enough to handle rtp ? |
| 18:29.50 | seanbright | no |
| 18:30.03 | seanbright | chan_sip uses res_rtp_asterisk |
| 18:30.10 | seanbright | chan_pjsip uses res_rtp_asterisk |
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| 18:32.00 | seanbright | chan_rtp uses res_rtp_multicast |
| 18:32.33 | seanbright | chan_rtp is used to send RTP without SIP signaling. it's probably not what you want. |
| 18:33.05 | _ganapathi_ | ooh. |
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| 18:55.12 | _ganapathi_ | <PROTECTED> |
| 18:55.27 | _ganapathi_ | using websocket . |
| 18:55.49 | _ganapathi_ | enabled websocket,http,http post module. but any other module dependent for this ? |
| 19:05.49 | sawgood | In the older days of (chan_sip): we could run: sip show peer 1000 (and) see details about the "phone type" (like say Yealink and see cool thigns like firmware versions) ... using PJSIP: is there a command like this (I tried) pjsip show endpoint 1000 |
| 19:06.14 | seanbright | pjsip show contact xyz |
| 19:06.15 | seanbright | maybe? |
| 19:06.30 | sawgood | looking |
| 19:07.52 | sawgood | not that one, sir ... but on the same topic: using pjsip show auth 1000-auth (we can see the secret) which is excellent |
| 19:08.25 | sawgood | I really dig that ... seeing the secret |
| 19:08.37 | seanbright | heh |
| 19:12.44 | seanbright | probably should fix that |
| 19:13.08 | seanbright | but if you have CLI access, you probably have access to the credentials anyway |
| 19:15.28 | Samot | sawgood; In PJSIP endpoints have nothing to do with contacts. |
| 19:15.40 | Samot | Contacts are an AOR thing. |
| 19:17.57 | Samot | I don't think anything in the cli output shows the user agent though. |
| 19:18.21 | Samot | I know it stores it somewhere as you can pull the UA with dialplan functions. |
| 19:19.52 | seanbright | database show registrar |
| 19:20.03 | seanbright | assuming they register and aren't static |
| 19:22.22 | _ganapathi_ | <PROTECTED> |
| 19:22.28 | Samot | Well there wouldn't be any UA information for a static. |
| 19:22.30 | _ganapathi_ | while connecting |
| 19:32.08 | _ganapathi_ | openssl minimum version for asterisk 16 ? |
| 20:09.23 | _ganapathi_ | anybody pls help me on openssl error |
| 20:09.53 | _ganapathi_ | using 14 earlier..it was working fine. after upgraded to 16 not working |
| 20:11.02 | seanbright | if it's compiling, it has a sufficient version |
| 20:19.34 | *** join/#asterisk Penguin (~xwQ5kwYl6@the.penguins.got.out.of.the.systems.at.penguinsystems.net) |
| 20:25.03 | alexandre9099 | hi, does anyone know how a POTS phone works "electrically wise"? I am trying to run a modem as a voice gateway (which i'm getting mild success), but i don't quite understand how POTS/PSTN works |
| 20:25.28 | alexandre9099 | for example, if i try to dial a number using DTMF tones nothing happens |
| 20:26.11 | alexandre9099 | (this has nothing to do with asterisk though) |
| 20:28.02 | Samot | First you have to understand a modem is not an FXS/FXO module. |
| 20:28.13 | Samot | It's not designed to expect or use voice. |
| 20:29.54 | alexandre9099 | but it does work for voice kinda well, sure it's not dedicated for that task, but it works |
| 20:30.14 | alexandre9099 | (it get's recognized as a sound card on ALSA) |
| 20:30.37 | alexandre9099 | anyhow, what i would like to know is how the phone works "inside the line" when dialing a number |
| 20:31.09 | alexandre9099 | maybe i'm not searching right, but i find absolutely nothing about POTS/PSTN insides |
| 20:31.12 | Samot | That's DTMF. |
| 20:32.01 | alexandre9099 | hmm maybe i'm doing something wrong then, is it literally just the DTMF tones when off-hook? |
| 20:32.07 | Samot | DTMF is either sent inband (in the audio payload) or outofband (its own audio payload) |
| 20:32.28 | seanbright | for traditional telco (a copper pair) it will always be inband |
| 20:32.34 | Samot | That. |
| 20:33.10 | seanbright | the other side "listens" for the dtmf and acts on them |
| 20:33.17 | seanbright | the other side generates dial tone and ringback |
| 20:33.21 | seanbright | etc, etc. |
| 20:33.27 | alexandre9099 | so in theory if i generate DTMF tones on audacity and play them to the line it should work, right? |
| 20:33.33 | seanbright | yes |
| 20:33.49 | seanbright | google for 'captain crunch whistle' |
| 20:33.55 | seanbright | enjoy the rabbit hole |
| 20:34.33 | Samot | Specially since there are numerous FXS/FXO options to achieve what you are trying to do on something not designed for this. |
| 20:34.34 | alexandre9099 | what would be the normal audio rate for a POTS line? maybe the rate is not right and so it fails to send the DTMF tones right :) |
| 20:34.40 | alexandre9099 | i'll give a look at it |
| 20:35.08 | Samot | The problem with your questions are that POTS settings are regional. |
| 20:35.17 | alexandre9099 | Samot, i would also like to understand how it works, i do not really intend to use this as a "production" device, but more as a learning experience |
| 20:35.30 | Samot | Learning POTS now is a waste of time. |
| 20:35.49 | Samot | All telecos have been transitioning from it. |
| 20:36.00 | seanbright | 8k |
| 20:36.08 | alexandre9099 | Samot, i know, but i can be curious on how it works, no? :) |
| 20:36.08 | Samot | In most cases now the copper is just in the last mile. |
| 20:36.25 | Samot | Where are you in the world? |
| 20:36.26 | alexandre9099 | seanbright, i will give it a try, i was doing with 16k or 9k |
| 20:36.29 | alexandre9099 | portugal |
| 20:36.32 | seanbright | oh |
| 20:36.41 | Samot | OK, you have different standards then us. |
| 20:36.44 | Samot | We're in the US. |
| 20:36.46 | seanbright | well, i'm talking north america, so... |
| 20:37.10 | Samot | So anything we would tell you is not going to be accurate in regards to POTS. |
| 20:37.25 | alexandre9099 | i see, anyhow i'll try 8k :) |
| 20:38.19 | alexandre9099 | yey, got it through :) |
| 20:38.27 | alexandre9099 | it was indeed 8k |
| 20:38.51 | seanbright | wonderful |
| 20:39.01 | seanbright | 9k isn't a thing that i have ever heard of |
| 20:39.22 | seanbright | multiples of 8k, yes. 9k, no. |
| 20:39.39 | alexandre9099 | no idea, i was trying to use slmodemd (to create an AT interface with the sound card) and it was talking something about the card only supporting 16k and not 9k... |
| 20:41.08 | alexandre9099 | btw, this thing i'm doing is kinda stupid and just to prove a point i did with an ISP technician that told me that either i got to pay or i simply couldn't use my landline phone number over voip (i have fiber, so the POTS/PSTN i'm talking about is over voip) |
| 20:41.23 | _ganapathi_ | seanbright: any idea about openssl error. moment wss connection creation then am getting this error. |
| 20:41.50 | seanbright | _ganapathi_: i don't. you aren't providing a ton of information other than "it doesn't work" |
| 20:42.38 | seanbright | if you open a browser and go to https://asterisk-server:port/wss do you see the "Upgrade Required" message? |
| 20:42.48 | _ganapathi_ | i have no idea what else need to provide for this error. sorry.. will u pls let me know any other area need to look into |
| 20:42.51 | seanbright | is res_http_websocket.so loaded? |
| 20:43.02 | seanbright | is it snowing in space? |
| 20:43.11 | _ganapathi_ | 500 error |
| 20:43.22 | seanbright | is res_http_websocket.so loaded? |
| 20:43.23 | _ganapathi_ | yes. res_http_websocket loaded |
| 20:43.54 | seanbright | ok, turn on debug output, hit the page again from your web browser, and then pastebin the debug output |
| 20:44.05 | seanbright | core set debug 10 |
| 20:44.37 | seanbright | assuming you have debug output going to a file (like 'full'). you'll have to confirm that in logger.conf |
| 20:46.28 | _ganapathi_ | asterisk/message / |
| 20:46.30 | _ganapathi_ | ? |
| 20:46.38 | seanbright | huh? |
| 20:46.54 | _ganapathi_ | <PROTECTED> |
| 20:47.06 | seanbright | what is at the beginning of that line? |
| 20:47.09 | seanbright | full => |
| 20:47.10 | seanbright | ? |
| 20:48.28 | seanbright | i have to leave in 12 minutes so let's get moving |
| 20:51.21 | _ganapathi_ | https://pastebin.com/u1ZFp9Sb |
| 20:52.55 | seanbright | your certificate, is it self-signed or legit? |
| 20:53.43 | seanbright | openssl's error message are garbage, obviously |
| 20:53.48 | _ganapathi_ | self-signed |
| 20:56.04 | seanbright | yeah, i dunno |
| 20:57.27 | seanbright | what's weird is that even after that error it looks like the WS is still opened\ |
| 20:57.37 | seanbright | res_http_websocket.c: WebSocket connection from '10.11.2.2:59117' for protocol 'sip' accepted using version '13' |
| 20:58.41 | _ganapathi_ | hmm |
| 21:00.07 | seanbright | anyway. i have to leave. hopefully 1 of the other 200 people in here wake up. |
| 21:00.09 | seanbright | adios |
| 21:00.45 | _ganapathi_ | ha ha. |
| 21:00.57 | _ganapathi_ | am awake only for this. |
| 21:09.07 | *** part/#asterisk _ganapathi_ (~Ganapathi@49.207.182.134) |
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| 21:26.35 | rgpx | hi - so I've been searching a ton about bad UDP checksums in tcpdump output, a few will be checksum OK, then ill see a bad one etc. Generally, is this a kernel-level setting people running asterisk in production use? Or rather it's indicative of another problem? |
| 21:27.11 | rgpx | (kernel level setting meaning does anyone disable checksums via ethtool or another method) |
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| 23:00.40 | Slade | anyone here have an opinion on a solid conferencing/bridge line for meetings? nothing self hosted or anything, some 'cloud' offering |
| 23:01.43 | Slade | i know amazons got something. i'm assuming microsoft too |
| 23:02.50 | Samot | What do you need? |
| 23:03.07 | Samot | If you have Asterisk and just need basic conf calls, that covers it. |
| 23:03.31 | Slade | yea dont have asterisk setup yet. I guess i could spin it up quickly. wanted to see whats out there tho |
| 23:03.45 | Samot | GoToMeeting and others like it. |
| 23:03.46 | Slade | I do just need very basic calls tho |
| 23:04.10 | Slade | yea theres gotomeeting, amazon, ciscos thing.. |
| 23:06.35 | Slade | i've seen a few that'll call participants to get them to join, seems handy |
| 23:17.42 | Slade | https://www.twilio.com/voice/conference probably doesnt stand up quickly |
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