hi, When calling from Sipura to softphone (either SJphone or the sip client in my e61 nokia phone), with the Sipura set to Prefered codec: g723 Use prefered codec only: no and the softphone only supporting g711, I get this reply from the softphone: U xxxx:11474 -> xxxxx:5060 SIP/2.0 488 Not Acceptable Here. Via: SIP/2.0/UDP xxxx;branch=z9hG4bK23e.248c6cd1.1,SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. To: <sip:1@test.com>;tag=ujk6mpqhbhhc6kj68irr. From: sipura line1 <sip:2@test.com>;tag=d0404120874d710eo0. Call-ID: e5052808-575f7746(a)192.168.1.52. CSeq: 102 INVITE. Warning: 304 192.168.1.3 Media type not available. Content-Length: 0. the invite was: U xxxx:5060 -> xxxx:11474 INVITE sip:1@test.co SIP/2.0. Record-Route: <sip:xxxx;lr=on;ftag=d0404120874d710eo0>. Via: SIP/2.0/UDP xxxx:branch=z9hG4bK23e.248c6cd1.1. Via: SIP/2.0/UDP xxxx:10001;branch=z9hG4bK-3e8ce192;rport=10001. From: sipura line1 <sip:2@test.com>;tag=d0404120874d710eo0. To: <sip:1@test.com>. Call-ID: e5052808-575f7746(a)192.168.1.52. CSeq: 102 INVITE. Max-Forwards: 69. Contact: sipura line1 <sip:2@xxx:10001>. Expires: 240. User-Agent: Sipura/SPA2002-3.1.5. Content-Length: 419. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 36143 36143 IN IP4 xxxx. s=-. c=IN IP4 xxxxxx. t=0 0. m=audio 62238 RTP/AVP 4 0 2 8 18 96 97 98 100 101. a=rtpmap:4 G723/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. If I set prefered codec to g711 on the Sipura it works normally. What is the best way to handle this on the sip proxy? Thanks, Richard _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users