Thank you very much! Very good presentation, highly useful . Only good stuff. Also I like to see Linux Mint OS used on presentation laptop. It is what I am using on my working laptop too.
Thank you, and yes I love linux mint! however I started using dropbox in place of google drive because drive (grive) is not reliable in linux and not supported by google??!!!
Absolutely agree. I used Gdrive and after Onedrive. Onedrive updates just crash when I do not have time to look why is crashed. I was saw dropbox supported by Ubuntu. So it will be my next step for personal documentation storage.
I have several customers using OOMA Voip and OOMA doesn't manage the SiP the same way you describe in the video where I can specify the SiP Destination IP. From OOMA support, "Unlike most other phone providers, we securely tunnel all SIP communications to our ooma SIP Servers." I am not sure how to mark the connection without the SiP IP. BTW: I have about 6 customers using OOMA and they are all experiencing choppy outgoing call quality (heard from their connected customer but not heard by them.) I want to set up queues but am stuck at the mark connection step. Any Tips? Awesome video BTW. Learned a lot.
@@yeo1397 The only time you should specify egress port (because you can only queue packets leaving an interface) is if you can not specify the traffic by other means. In the case of SIP and RTP packets, there are very well defined ways to specify those traffic types.
@@SkibidiWaPaPaPaPaThanks for the reply! Also, I have another question. I know that this is an irrelevant question in your presentation, but this is bugging me for a long time. When I use PCQ, does the limit is being measured by Packets or KiB in the latest RoS version? I was very confused about this parameter. Queue Size in KiB, but how do I convert KiB into a number of packets just like the default fifo queue type?
Hello, callers cannot hear me when they call in (I can hear them), and on another VoIP app, the calls are oftentimes dropping. I was told that SIP ALG needs to be disabled, however, I work from on a phone data plan that I tether to my laptop. Will the phone company be able to disable SIP ALG in this regard, or what other options may I look into? Thank you!
hey i really need some help i am doin dual wan load balancing with mikrotik and my clients whatsapp calls are not connecting nor the voip box dont register? can you help me what exactly i should do.. please and thank you
I'm not even sure that whatsapp uses SIP protocols. I would start with finding out what protocols whatsapp uses OR find out how whatsapp server subnets are, that way you can prioritize by subnet / ip destination rather than protocol. If it is using SIP protocols please reference my video on SIP ALG as thats where I talk about SIP problems, one of which the problem you are describing. ru-vid.com/video/%D0%B2%D0%B8%D0%B4%D0%B5%D0%BE-tM7wyKdnIKA.html
I use 2 isp, and use 2 Mikrotik with different segments, segment 192.168.216.0/24 on Mikrotik 1 isp 1, and segment 192.168.223.0/24 on Mikrotik 2 isp 2, but when one isp has a problem, for example ISP 1 down, I move the segment path 192.168.216.0/24 through mikrotik 2, with this configuration there is one way call, can you tell me how to fix it ??
