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 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users