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asterisk1_ip
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
Call-ID: 4832738...