I have problem with calls from SJphone to Linksys SPA-922. When I make call in this direction I cannot hear on SPA the caller but caller can hear me. When I initiate call from SPA, then everything go well. There is no firewall in between SJphone and asterisk nor SPA and asterisk. After a long investigation I found the problem.
Problem is that when SJphone initiates the call it results that voice from SJphone is encoded using GSM codes (which SPA doesn’t support). Other directin is encoded using G711 supported by both sides.
I am using SIP protocol.
canreinvite=no (I prefer that due to complexity of network.)
All voice traffic goes through asterisk.
My first question is: Is it expected behavior that call is established using codecs that booth sides do not accept? Or am I doing something wrong?
Second question: Can asterisk translate voice from one codec to another?
Of course, I can workaround by adding disallow=gsm for SJphone to sip.conf but this is not what I want. I want to disallow gsm when somebody calls SPA.
Or better, I want to recode gsm to something understandable by SPA while allow one SJphone talking to another SJphone using GSM codec.
PS: If I am asking wrong forum, please let me know.