Finish the configuration of FreeBPX Asterisk server

Hi, I have 18 years of experience in VOIP and asterisk. I offer my services for your project. Please have a look at my profile to see all my experience details. I assure you of a high quality professional service. We can chat out details. I can check now , you may award only once the issue is resolved.

This might be an issue with: NAT Sip registry RTP time-outs Re-invite I am able to fix this for you, please go through my past PBX projects and customer feedback over nine years

Hello, I'm a voip specialist with over 10 years of experience working on the TS department of a US voip company; I can install, troubleshoot and customize many SIP devices including: softphones, ATAs, IP-phones and PBX systems like Asterisk/FPBX. I read your project description, and I can certainly help you fix this problem that you're experiencing. Feel free to contact me at any time to get this project started. Thank you.

I have 10 years of experience in asterisk, freepbx, Vitalpbx, Elastix, Incredible pbx, fusionpbx, freeswitch, A2billing, ASTPP and voip solutions. please check my profile for details

Greetings It looks like trunk configured for chan_sip driver, but incoming calls are handled by chan_pjsip. I can solve it easly. Regards.

I can help you with troubleshooting your freepbx and also configure g729. I am able to start immediately contact through chat.

I am certified in sangoma freepbx solutions, write me and I can give you some help with your project

Hi. I'm Jonatan. I work with asterisk /freePbx since 2009. I can help you with the configuration.

I can easily resolve that problem, I get some of this errors before when trying to setup incoming trunks.

Hello, I can resolve your problem quickly and deliver it to you. Codec error does not appear according to the problem. Your configuration will need to be checked from the beginning. Submit it to your evaluation, we wish you good work.