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[matthew at matthew.at]

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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[msc at freeswitch.org]

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:

[matthew at matthew.at]

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:

[brian at freeswitch.org]

What svn rev and what firmware rev on the phone?


On Jan 7, 2009, at 12:33 PM, Matthew Kaufman wrote:

[matthew at matthew.at]

Brian West wrote:

[brian at freeswitch.org]

Please update the phone to 3.1.1.0137

http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_3_1_1_release_sig.zip

We can't support anything but the latest firmware... the same applies
to most phones since we routinely report, fix and work around issues.
2.1.2 is very old.

/b



On Jan 7, 2009, at 12:46 PM, Matthew Kaufman wrote:

[matthew at matthew.at]

Brian West wrote:

[matthew at matthew.at]

Matthew Kaufman wrote:

[anthony.minessale at g...]

could you use a pcap util like tcpdump or wireshark to capture the traffic
so we can see the rtp too that may help to figure it out.

On Wed, Jan 7, 2009 at 1:44 PM, Matthew Kaufman <matthew@matthew.at (matthew@matthew.at)> wrote:

[matthew at matthew.at]

Anthony Minessale wrote:

[anthony.minessale at g...]

is it possible to get a binary pcap?
That way we can look at it in wireshark.
you can email it direct to me and brian


On Wed, Jan 7, 2009 at 3:21 PM, Matthew Kaufman <matthew@matthew.at (matthew@matthew.at)> wrote:

[matthew at matthew.at]

Matthew Kaufman wrote:

[matthew at matthew.at]

Matthew Kaufman wrote:

[anthony.minessale at g...]

Did you ever find out if the rtp was making it to your phone?
Did you get around to testing the echo exten? That is the most basic call you can do
it is 1 leg call just playing your own audio back. Also 9998 plays the tetris song with the tone generator.



We for sure can see rtp packets in the pcap bound for your phone.
This is a very unique problem as many people get this basic situation working daily so
it must be a network issue of some sort.

Can we rule out your network where the phones live by testing some phones on the same network as FS?
Do you have a hub you could put the phones on so you can packet sniff the traffic to them?





On Thu, Jan 8, 2009 at 12:51 AM, Matthew Kaufman <matthew@matthew.at (matthew@matthew.at)> wrote: