Sent this to grandstream support but has anyone else seen this issue. All of the previous sdp rtpmap are correct until the grandstream sends this. I have been using disallow=gsm and canreinvite=no to get around the problem.
----- Original Message ----- From: Glenn Dalgliesh To: [EMAIL PROTECTED] Sent: Thursday, May 20, 2004 4:34 PM Subject: rtpmap issue I am seeing ATA-286, running Program--1.0.4.55, send and INVITE message with rtpmap:3 PCMU/8000. 3 is a well-known port and should be mapped to GSM not PCMU. 162.33.165.203 = HandyTone ATA-286162.33.165.198 = Asterisk SIP MESSAGE 16 162.33.165.203:5060(3) -> 162.33.165.198:5060(2) UDP Frame 16 20/May/04 15:26:16.0653 TimeFromPreviousSipFrame=0.0043 TimeFromStart=2.4311 SIP/2.0 200 OK Via: SIP/2.0/UDP 162.33.165.198:5060;branch=z9hG4bK2f81a326 From: <sip:[EMAIL PROTECTED]>;tag=as4fb7ebf5 To: <sip:[EMAIL PROTECTED]>;tag=9ce2cb909d0a97ff Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream HT286 1.0.4.55 Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 147 v=0 o=5003 8000 8000 IN IP4 162.33.165.203 s=SIP Call c=IN IP4 162.33.165.203 t=0 0 m=audio 5004 RTP/AVP 3 a=rtpmap:3 PCMU/8000 a=ptime:20 _______________________________________________ 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
