[Asterisk-Users] rtpmap issue w/Grandstreams

Glenn Dalgliesh asterisk at techhat.com
Thu May 20 13:47:44 MST 2004


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: support at grandstream.com
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:1113 at 162.33.165.198>;tag=as4fb7ebf5 To:
<sip:5003 at 162.33.165.198>;tag=9ce2cb909d0a97ff Call-ID:
c6cfa0a62bef9d7a at 162.33.165.203 CSeq: 102 INVITE User-Agent: Grandstream
HT286 1.0.4.55 Contact: <sip:5003 at 162.33.165.203> 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




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