Hi! You need to add this to EACH and EVERY sip user, not just in [general]:
disallow=all allow=ulaw allow=alaw See also: http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone Cheers, Philipp > [200] > type=friend > username=200 > host=dynamic > context=home > reinvite=no > canreinvite=no > > [201] > type=friend > username=201 > host=dynamic > context=home > reinvite=no > canreinvite=no > > I turned on sip debug, and noticed the following in the output: > > v=0 > s=SIP Call > c= IN IP4 192.168.2.29 > m= audio 5004 RTP/AVP 0 > a=rptmap:0 PCMU/8000 > a=ptime:20 > > Found audio format UNKN > Found description format PCMU > Capabilities: us - 4, them 4/0, combined - 4 > Non-codec capabilities: us - 1, them - 0, combined 0 > > Does anyone know why this could be happening? Thanks, > > Ron > > > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
