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
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: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Lists Sent: Saturday, September 27, 2003 7:01 PM To: asterisk-users@lists.digium.com 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 Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users