Great video, thank you, please in a dialog, how to identify a transaction? I know that a dialog can be identified with the Call-Id header, To and From tags

This script doesn't work on Debian 11 :(

Anybody have experince with Sonicwall NSA 2700 & SIP with NAT ? We tried using Meraki MX 85 but got one way audio and coud not get it resolved. Looking for any informatio if Sonicwall NSA 2700 would handle SIP NAT better ?

more than half of the video is litterally the download lol!

I still don't understand....even I cannot make my sip-ua status become registered......poor me

very clear ! good video

Hi I’m having a issue with forwarding calls from a landline to zoiper phone with sip trunking wandering if I can email you for guidance.

Beautiful!!

is it possible to be bypassed?

Thanks for the video

Excellent video. Thank you

Hy Keith, I have a final project for my college. I'm buliding a interconnection between asterisk, openSIPS, and Kamailio. Im facing a little problem while connecting kamailio to asterisk. Would you give me help ? I'm ready for paying for your help. you can contact me at [email protected]. Thanks

thank you so much for the useful explanation I have a question if I have python already installed in my host machine (mac) can I access those packages without install them?

So what if you dont have 2 way audio, user (A) in Arizona home with VPN site to site to HQ, user (B) at home in Georgia with VPN site to site to HQ, User(A) calls User (B), call timer is running but neither user can hear each other.. User(B) calls User(A) , call timer is running but neither user can hear each other. Important to understand .... User(A) can call HQ just fine, User(B) can all HQ just fine can have hours of conversations from remote to HQ, HQ can call User(A) or User(B) just fine. users within HQ can call internally just fine. its just the remote to remote seems to be the issue. User(A) has DynamicDNS (no-ip.com) on firewall, modem is passing public IP to firewall, User(B) has DynamicDNS as well, and too has public IP passing to firewall

Anyone can answer ? Does proxies have capability to carry Media ???

Sir, need training on Session Border Controller. I am taking your coaching in Udemy. But I couldn't fin your session on SBC(session BorderController) Please reply once you see this comment. I would like to be a part of your training sessions. Please provide the link of SBC if you have any . Thank you so much . you were too gooooood!!!

I do not think that response OK packet can reach the phone. The response follows the via header and it would try to send the response to private IP.

Great video and explanation, you are a talanted teacher, thanks!

Thanks, this script fixed my issue Can you also do a video of how to use sipp with askterisk pbx?

Thank you so much for this. Very helpful. Can you do one for Kamailio?

Informative

Can you explain the "payload" and how it works?

This video just turned on my brain, I found my problem, I need to enable NAT transversal in the SonicWall?

Please upload more videos on SIP protocol and SIPp. Appreciate for all efforts made for this community.

Need to tell usage how to communicate two pjsip in different architecture of processor

If you do not want strange problems, never use STUN. This can be useful only for short tests.

Hey thanks for the script and the video. You should make a good video on usage. Assuming this will also work for Ubuntu 18.04

I have a one audio problem on vpn client remote extention. Do you have any tutorial about this.

Great video. Thanks. Didn't mention if the public ip address of the router is static or dynamic. What about if the call originates from outside?

very very nice its right and true talk

Awesome !!! Thank you !!!

This video was great, and clarity of your explication are very good. Thanks.

Thank u for your Video i can't find The Elastix 4 or Elastix MT iso anywhere please share it with me Best Regards

very helpful video,Really learned a lot helped me to configure server that i bought from datasoft datasoft.ws/

thx for your great explaination! Is it possible that oneway audio is caused by the nat layer if it is only a temporary issue?

Thanks...any experience with asterisk? I have server running on ubuntu with dd-wrt as gateway and am having these issues. ICMP packets are not being routed back to the sip client properly on remote extensions