I'm working on migrating from TDM to SIP providers and built up my SBCs a few months back. I set up a predot (94) to send calls out as 10-digit for the VISP to route to a special DoD Phone network back in December and it works great.
This week we are working to get our commercial lines off TDM and onto the SBCs, so I set up another predot (97) to send calls out as 12-digit for the VISP to route to their commercial carrier (AT&T).
Now, this is the odd part. 94 works great with clear 2-way audio. 97 calls over the exact same SBCs, same Dial-Peers, the call connects, the onsite endpoint gets audio from the distant end just fine, but the distant end cannot hear the on-site audio. BUT, the on-site endpoint hears itself about a fraction of a second later. Like I can say hello, and by the time I finish I am already hearing "..llo" almost like the stream is curving back around on me.
What the heck am I missing? I can't imagine it is a codec issue if 94 works fine. dial-peer config issue? sip-profiles? I am a total greenhorn on this SBC / SIP so I am just pounding through documentation trying to figure out what to look for.
Any insight would be immensely helpful. Have a TAC case open, but would like to figure it out before they get back to me.
PS - may be part of it? we have SRTP configured endpoint to VISP. both 94. & 97. calls connect with the "padlock" icon next to the call timer. and I swear the first day I tested this in early January I got 2 way audio.
Rob
