Updated to current a couple days ago, default dialplan, extensions are registered as 1001 and 1002. tcpdump on the freeswitch host shows RTP going both ways to/from both phones. This is running on CentOS, so your log and debug instructions don't make much sense, but I do have SIP tracing enabled... see http://pastebin.freeswitch.org/6694 The phones are on the 10.10.155.0/24 subnet, the switch is on 198.202.199.1, but there's no NAT between, just a router... from the POV of the switch, it can see all the phones on the 10.10.155.0/24 network directly. Matthew Kaufman Michael S Collins wrote:
What FS revision and is this a default Dialplan? Please pastebin the output of the CLI while making test calls. Be sure to press F8 to enable debug messages. Also, if you can do so turn on SIP messages by launching FreeSWITCH with TPORT_LOG=1.
-MC
Sent from my iPhone
On Jan 6, 2009, at 10:22 PM, Matthew Kaufman <matthew@matthew.at> wrote:
I have two Polycom phones (one 550 and one 650) successfully registered to the switch. If I call from either extension to the other and answer, audio flows from the calling party to the called party, but audio does not flow from the called party back to the calling party. Even more strange, the called party does not answer, then when the call is sent to voicemail, the calling party *also* does not hear any of the voicemail greeting, though they are recorded successfully. Calling the voicemail box directly from either phone *does* work, and the calling party can hear the prompts just fine in that case.
There is no NAT between the phones and FreeSWITCH.
Suggestions?
Matthew Kaufman matthew@matthew.at
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