I am using asterisk with the Amportal tool and it works fine between SIP (soft) phones using linux. Now I’m trying to create a SIP <> PSTN gateway. It does work, but I have an audio delay of +/- 500 ms.
If I call from a SIP phone to a normal phone number, the outgoing voice has this delay.
If I call from a normal number to the asterisk box who routes the call to a SIP phone, again the SIP phone has the delay.
The normal phone does not create a delay. Calling between SIP phones does not create a delay. I have a I4L compatible ISDN card (I load it with modprobe hisax id=HiSax type=35 protocol=2)
Some info about my configs:
device => /dev/ttyI0
port = 5060
bindaddr = 0.0.0.0
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
Any idea what the problem is?