I am new to this forum and voip so please be patient with me.
I just added a second DID to a provider and it seems to work except the caller does not hear a ring back and will terminate after a given amount of time (18 seconds). I have been working with the provider and they insist that it is my problem. I decided to get a tcpdump of the handshaking to make sure everything is good. I compared the handshaking of the second DID with that of the first DID (which works perfectly) and first I noticed that the INVITE packets from the provider are pretty different which tells me that the configuration of the two DIDs are different. I then noticed in the Session Progress handshake gets truncated because the limit size of the packet is reached.
I tried to include the truncated packet capture but I couldn’t post it.
Does anyone know why the Session Progress handshake packet would be larger than the max packet size? Is there anyway that I can control it?
I am curretnly usingFPBX-188.8.131.52 (Asterisk PBX 13.22.0)
Thanks for any help