I am very new to all of this so forgive me if some of my questions don’t make sense
I experience 1-way audio when I place calls. I can hear them but they can not hear me.
If I bypass asterisk and call through the gsm gateway directly, there is absolutely no problem with the audio.
Ubuntu 12.04 LTS
Portech mv-378 gsm gateway
I want to call in my asterisk server and have asterisk direct the call to the appropriate port depending on the number dialed. Instead of carrying 4 phones, I can leave the other sims at the office and take advantage of in-network dialing using Asterisk routes & trunks for the logic and the gsm gateway to make the calls using a sim from the appropriate network. I initiate the calls from a sip client using an extension that I created in the asterisk server and that registers fine.
Here is my configuration:
The asterisk server and the portech gateway are connected via a switch directly into the isp cable connection, not a router and each has device has its own public IP so there should be no firewall and no NAT.
I created an extension (11116666) that registers into my asterisk from the cellular network.
I have a feeling the problem might have to do with my incoming trunk settings, but I do not know how to set them up properly. Right now, the incoming settings fo my trunks are totally empty!
Please let me know if there is any file or outputs I could provide to help you guys help me… Thank you!