[asterisk-users] Grandstream GXP2020 + Asterisk 1.4.11
we at deuromedia.at
Mon Nov 12 05:12:26 CST 2007
I`m using several GXP2020 phones with newest Firmware 126.96.36.199.
Asterisk Version: 1.4.11.
It happens several times that users complain that the caller cannot hear the
transmitted voice from the phones....
Also now it happens quite often that callers on hold beeing dropped.
Environment: ISDN with chan_misdn and SIP internal calls. No NAT no DNS name
(only IPS configured).
I configured in sip.conf and on the phone now that "alaw" is preferred. As I
saw in the FMW Bug list that GSM is not a good option.... Also I set the
canreinvite=no as it is also configured in a Grandstream manual.
I use on every phone the 10000 as local port and in the rtp.conf I allowed a
range from 10000 - 50000. As far my SIP knowledge is up to date the local
port has not to differ from phone to phone or I´m wrong?
Any idea or useres which had the same problems and fixed it?
More information about the asterisk-users