Try on the Grandstream DTMF via INFO. Also use uLaw for codec. If behind the NAT just say NAT=YES and REINVITE=NO. It works like a champ. Regards, Uriel
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lists Sent: Saturday, September 27, 2003 7:01 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP/ Grandstream Issues I just got a grandstream SIP phone Here is my sip.conf for the phone [mlh] type=friend insecure=yes username=mlh secret=mlh host=dynamic canreinvite=no The phone as the default config on it. If I use the phone to call a Zap interface (a tdm card) the voice sounds all choppy. If I use the phone to call a x100p card, it does not dial what I dial (no DTMF) I don't know what else to try.....should I change the vocoder (it is on PCMU at the momemnt) I am using the phone on a LAN so bandwidth is not an issue. Any Help would be great, Michael _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
