When I make a simple phone call from one Budgetone 101 to another, the speech sounds slurred and slow, sort of like the person is talking under water. Both phones and the Asterisk server are on the same subnet.
Both phones are configured to use the PCMU (ulaw) codec as first choice, and the Voice Frames per TX parameter is set to 2. Incidentally, if I directly IP dial from one phone to the other (bypassing Asterisk) the speech sounds excellent. I'm running a CVS build from Feb. 1, 2004, and there is a Digium X100P card with one incoming CO line in my machine. The first part of my sip.conf looks like this: [general] port=5060 binaddr=0.0.0.0 disallow=all allow=ulaw [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
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