asterisk users - Nov 2007 - Grandstream GXP2020 + Asterisk 1.4.11

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

> I`m using several GXP2020 phones with newest Firmware 1.1.4.18.
I had issues with phone locking up using 1.1.4.18. I've now gone to 1.1.4.22
and have eliminated that.
> Asterisk Version: 1.4.11.
Me too. Still testing 1.4.13 on a non-production system.
> I use on every phone the 10000 as local port and in the rtp.conf