[asterisk-users] Grandstream GXP2020 + Asterisk 1.4.11

[Erik Wartusch]

Hi,

I`m using several GXP2020 phones with newest Firmware 1.1.4.18.

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?

My sip.conf:

[test1]
        type=friend
        context=outgoing
        username=test1
        secret=987454
        qualify=yes
        host=dynamic
        nat=yes
        canreinvite=no
        disallow=all
        allow=alaw
        allow=ulaw
        callerid=Test <0>
        insecure=very

Kind Regards,

Erik