[Asterisk-Users] sip tone question

[Don LeBlanc]

Hello All, 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 :   (no NAT) to 172.16.0.25:5060 NOTICE[12301]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support incomplete.  Turn off on client if possible NOTICE[12301]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support incomplete.  Turn off on client if possible   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.   Sincerely, Don LeBlanc