I am running FreePBX for longer time with SIP phones and Skype gateway. Works fine. Now, I installed ISDN card Asuscom P-IN100-ST-D with drivers dahdi_hfcs and finally, it started to work, I am able to make a call SIP->ISDN and also ISDN->SIP. Unfortunatelly, audio works one way only - from SIP to ISDN works fine, but no audio from ISDN to SIP. Call establishing works fine, hangup also. I tried dahdi_monitor tool and I see both Rx and Tx audio works nice. Also when I redirect ISDN call to voice mail and leave a message there, when I call from SIP phone to voice mail and hear the message, audio is excellent. Please, what I should check? Any special SIP phone settins is necessary?
I am running FreePBX 188.8.131.52 and Asterisk Version: 13.9.1 on CentOS.
Thanx a lot for your help.
Solved, finally! With wireshark I analysed communication. When two SIP phones was connected together, both uses PCMU codec and works. But when I made a call with ISDN, SIP phone used PCMU, but FreePBX used PCMA. When I disables PCMA in Asterisk SIP settings, everything works fine. I thought, FreePBX should negotiate the same codec for both sides, doesn’t it? And during call, must both sides use the same codec?