I’ve noticed a problem:
- Asterisk 188.8.131.52 (pretty old)
- SIP phone with ALAW codec
- SIP phone with G729a codec
- SIP phone with G729a codec (one more)
- no NAT
When dialing to each other everything works fine.
But when I dial from phone with one codec to the phone with another codec AND it transefers my call to another one (ALAW => G729a (OK) => G729a (problem!)), the remote party hears me, but I don’t hear anything.
I think it’s a bug of Asterisk (because when I just dial without transfers, everything works between phones with ALAW and G729a codec).
What is the best way to solve this problem? Does it appear in 1.8 and later releases?
Best regards, Alexey.