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<DIV><FONT face=Arial size=2>Hello All,</FONT></DIV>
<DIV><FONT face=Arial size=2>We are running Asterisk on a linux server as a SIP
proxy with Cisco ATA 186's at the subscriber end. For long distance we
have iax2 connectivity with a ip carrier. For local calls we are routing
out through a commercial VEGA voicestream pots unit to an adtran channel bank
and then from there to our class 5 soft switch. The sip to sip calls
and the long distance calls work great. The problem is with the local
calls going out the pstn gateway (vega to channel bank to soft switch).
When I dial a local call from one of my ATA186 units, I hear a sound that is
like someone pressing down a digit on the phone key pad and temporary
dialtone. After about 1 second of this, the call proceeds & terminates
normally. Below is part of what comes up in the sip debug log
:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>(no NAT) to 172.16.0.25:5060<BR>NOTICE[12301]: File
rtp.c, Line 263 (process_rfc3389): RFC3389 support incomplete. Turn off on
client if possible<BR>NOTICE[12301]: File rtp.c, Line 263 (process_rfc3389):
RFC3389 support incomplete. Turn off on client if possible</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Does anyone know if it is possible to disable rtp
support ONLY to my local calls through the pstn ? My vendor with the Vega
product says that the calls going through it, should allow the Vega to do all of
the rtp. Any help would be greatly appreciated.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sincerely, Don LeBlanc</FONT></DIV></BODY></HTML>