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2005 Jul 14
0
Seperate RTP server, voice not being received.
...| RTP \-----------> <PacWest RTP> I can make or receive calls from my SIP Phone, however voice only works in one direction. I see the following when a call happens on Asterisk2: Sip read: INVITE sip:XXXXXXXXXXXXXX4@<Asterisk2_IP>:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP <PacWest SIP>:5060;branch=z9hG4bKad7c753a29b-a9fbbac5 Via: SIP/2.0/UDP <PacWest RTP>;branch=z9hG4bKac483280508 To: <sip:9167243434@<PacWest SIP>;user=phone> From: sip:7079953217@<PacWest RTP>;tag=1c483274524...