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| 02:32.33 | drmessano | file: Is this syntax correct for the 'i' option? |
| 02:32.36 | drmessano | MulticastRTP/linksys/239.10.1.1:5004/239.10.1.1:6061/i(10.255.255.1) |
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| 06:04.47 | *** topic/#asterisk is AstriCon 2019 in Atlanta! http://www.astricon.net/ -- #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.29.1 (2019/10/16) 16.6.1 (2019/10/16) Standard: 17.0.0 (2019/10/28); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
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| 14:28.52 | seanbright | drmessano: that appears to be correct |
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| 15:22.27 | drmessano | seanbright: Thanks! |
| 15:22.48 | seanbright | i'm basing that on looking at code, i have no idea if it is "correct" |
| 15:22.53 | seanbright | i don't know anything about multicast |
| 15:23.11 | drmessano | Right. Understood. Just wanted a syntax conformation. |
| 15:23.19 | drmessano | Confirmation |
| 15:24.24 | drmessano | I'm trying to implement this in the worst way possible, so having some pieces confirmed helps a lot |
| 15:26.15 | Samot | drmessano: Oh, using the Lanny Method approach. Daring. |
| 15:26.36 | drmessano | And Brave |
| 15:31.19 | drmessano | I need to Asterisk installed at home and make a few multicast test calls first, to confirm the endpoints and my VLC client can receive audio |
| 15:31.28 | drmessano | get* |
| 15:32.05 | drmessano | After that I can nail down the multicast routing |
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| 15:36.02 | Samot | Cool, let me know how it goes. |
| 15:36.26 | Samot | It's something I don't see a need of doing (for me) but I'm interested if it does work. |
| 15:37.40 | Samot | But who knows. We've been messing around doing IP cameras so I could see a door buzzer with video or something being needed to alert people in the the possible future. |
| 15:40.15 | igcewieling | Algol has some interesting products, we use them for overhead paging, but they have doorphone (and video doorphones) with SIP. I don't know if they do multicast. http://www.algosolutions.com/products/doorphones-security.html |
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| 16:14.47 | wonderworld | hey all. i remember using a very nice ncurses gui to visualize sip traffic. forgot the name. ;) |
| 16:19.15 | igcewieling | sngrep |
| 16:24.09 | igcewieling | Wow, 1178 IPs blocked by fail2ban for ssh attack |
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| 16:36.48 | wonderworld | igcewieling: hey thank you thats it. very nice tool. |
| 16:41.42 | wonderworld | i have a problem with zombie sip channels. sip softphones register via WSS and are bridged in a confbridge. when users close their browser, etc the sip channels stay alive and the count of participants in confbridge doesn't go down. |
| 16:42.32 | wonderworld | i guess I'd need to set a timeout that checks if sip or websocket signalling has been received in the last x seconds, but I am not sure where to do it. |
| 16:45.27 | igcewieling | there are session-timers and rtp keepalive options for pjsip and chan_sip. |
| 16:46.10 | wonderworld | thank you i will read that up |
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| 17:11.34 | Samot | wonderworld: Is the WebRTC phone sending a BYE when the browser closes? |
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| 17:25.02 | wonderworld | Samot: it should but it sometimes doesn't have enough time to do it when the browser is being closed. |
| 17:25.28 | wonderworld | the sip channels then just stay alive forever and i am looking for a way to detect the right ones and kill them |
| 17:25.53 | wonderworld | this is a pretty old asterisk though, maybe i should update it first to the latest version and try again? |
| 17:26.06 | wonderworld | i use 14.3.0 |
| 17:28.08 | Samot | OK so the issue is that the SIP client (WebRTC) isn't actually disconnecting the call or sending anything to indicate a BYE should happen. |
| 17:28.19 | Samot | These are not "zombie" calls to Asterisk. |
| 17:29.39 | igcewieling | We don't live in a perfect world. If the PBX can't handle a loss of connectivity with a client, then it is broken. |
| 17:29.48 | wonderworld | maybe my terminology is wrong. |
| 17:30.03 | Samot | You have an active call via WebRTC. |
| 17:30.16 | wonderworld | this would be one of the channels without SIP client shown by "sip show channels" |
| 17:30.29 | wonderworld | 176.22.196.21 sq siirlcoiq3oqosd (alaw) No Rx: ACK sq |
| 17:30.32 | Samot | You're closing the browser and the WebRTC client isn't sending a BYE. |
| 17:30.37 | igcewieling | have you considered using PJSIP instead? |
| 17:31.10 | Samot | This is just like if a hardphone hung up their handset and the phone didn't send a BYE to the PBX |
| 17:32.19 | wonderworld | Samot: is there a way to work around this? |
| 17:32.36 | Samot | Outside of fixing your WebRTC client? I'm not really sure. |
| 17:32.47 | wonderworld | igcewieling: if it would solve my problems, i would try port it to pjsip |
| 17:32.53 | Samot | How do you know which channels in the bridge are not active anymore because a browser was closed? |
| 17:32.59 | Samot | To Asterisk the call is still up. |
| 17:33.20 | wonderworld | Samot: i am testing by myself with the browser just spawning new sessions |
| 17:33.36 | igcewieling | wonderworld: chan_sip support session timers, but I'd look at rtpkeepalive first. |
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| 17:33.49 | Samot | So each time you open the browser the WebRTC is making a new call? |
| 17:34.19 | wonderworld | igcewieling: i did that. but when setting rtp debug to on, i still see asterisk pushing out the rtp stream to the dead clients and the timeout doesn't trigger |
| 17:34.31 | wonderworld | Samot: yes |
| 17:34.45 | Samot | OK so you have an issue with your WebRTC client. |
| 17:34.48 | igcewieling | wonderworld: I think it timesout on no received rtp |
| 17:34.52 | Samot | It's not disconnecting calls to the other side |
| 17:34.57 | Samot | So it just keeps making new calls. |
| 17:35.12 | Samot | Because once you close the broswer, the client thinks there are no active calls. |
| 17:35.16 | wonderworld | Samot: basicly it's the RTP implementation in Firefox and CHrome |
| 17:35.26 | Samot | This has nothing to do with RTP |
| 17:35.32 | igcewieling | wonderworld: test disconnecting by removing the network connection. that way nobody will get hungup (pun intended) on the "closing the browser" part. |
| 17:35.34 | Samot | A BYE doesn't have RTP in it. |
| 17:35.58 | Samot | Again this is like a regular phone never sending a BYE but still making new calls on the line like it's not in use. |
| 17:36.12 | Samot | Your WebRTC client needs to be fixed. |
| 17:36.25 | igcewieling | You problem appears to me to be a general issue with the PBX losing connection to the softhone, I can't imagine how closing the app is any different from a connectivity outage. |
| 17:36.40 | Samot | This isn't a PBX problem |
| 17:36.53 | Samot | This is a WebRTC client problem. It's not ending calls properly. |
| 17:37.01 | Samot | And then it just creates new ones. |
| 17:37.31 | igcewieling | just like if the network connection dropped? |
| 17:37.53 | Samot | Well I haven't seen any debugs or data. |
| 17:38.02 | wonderworld | Samot: sorry i meant webrtc implementaion, not RTp of course |
| 17:38.16 | Samot | But sure, the PBX isn't seeing the dead audio stream.. |
| 17:38.33 | Samot | Unless the WebRTC is background while the browser is closed. |
| 17:38.38 | Samot | I don't know. |
| 17:39.34 | Samot | wonderworld: So you close a browser and after a minute you still see the same amount of channels in the conf bridge? |
| 17:39.51 | wonderworld | would it be of any use if the softphone would use different usernames to start calls? |
| 17:40.00 | wonderworld | Samot: yes exactly |
| 17:40.22 | Samot | And it will stay up in the conf bridge even after all the other calls have properly disconnected? |
| 17:40.25 | wonderworld | Samot: it works sometimes, but 20-30% of channels stay there until i kill them manualy |
| 17:40.27 | Samot | It just will never drop? |
| 17:40.53 | wonderworld | yes, confbridge showed 12 participants with zero ppl in it |
| 17:41.18 | Samot | And you had 12 active calls and channels in the PBX? |
| 17:41.57 | wonderworld | i was thinking about detecting the time, the last sip package arrived from a client and script myself a timeout around it |
| 17:42.11 | wonderworld | but i couldn't find a way to get this information yet |
| 17:42.14 | Samot | OK, humor me on this. |
| 17:42.50 | Samot | Replicate the issue. Close the browser, have the call still active in the PBX...then reboot the computer. |
| 17:43.42 | wonderworld | reboot the client or the pbx? |
| 17:43.53 | igcewieling | wonderworld: are you blocking ANY ICMP? |
| 17:44.05 | wonderworld | igcewieling: this might be |
| 17:44.22 | wonderworld | i think i am just forwarding the needed ports for sip/webrtc/rtp |
| 17:45.36 | wonderworld | i can ping the pbx on it's public ip |
| 17:45.42 | wonderworld | should that be sufficient? |
| 17:47.10 | wonderworld | let me try to reboot my box as samot requested. back in a minute |
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| 17:49.28 | wonderworld | ok, replicated the issue and rebooted the client machine. channels stimm there |
| 17:49.36 | wonderworld | still |
| 17:49.59 | wonderworld | maybe i will first update to the latest asterisk and try again |
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| 20:32.56 | overyander | can you specify multiple e-mail addresses in the e-mail field for a voice mail box? |
| 20:33.30 | igcewieling | I'm reasonably sure you can't. |
| 20:33.53 | Samot | email1@domain.com|email2@domain.com |
| 20:34.17 | Samot | You need to use a | just like the options. |
| 20:36.10 | overyander | thanks Samot. Is there a space between the addresses and the | character? |
| 20:36.23 | Samot | No. |
| 20:38.17 | igcewieling | nifty. |
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| 21:09.24 | igcewieling | Samot: any idea when the GUI started changing , to | when writing out the e-mail field ? I'm reasonably sure FreePBX 2.9 did not do that. |
| 21:10.39 | Samot | I have no clue about the GUI but the | has been a voicemail option for a very very long time. |
| 21:14.40 | igcewieling | I suspect | for voicemail e-mail field was supported at least as far back as 1.4. |
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| 22:01.13 | nny | i feel stupid, what's the command to show all channel variables for a give channel vs. what (ex: sip show channel) shows? |
| 22:01.19 | nny | given* |
| 22:01.23 | nny | globals, etc |
| 22:01.36 | seanbright | core show channel <channel> |
| 22:03.14 | igcewieling | Dumpchan if you want to do it in the dialplan |
| 22:06.12 | nny | ty to both of ya |
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