I have a user that just got a broadband connection so she could have an
extension off our pbx. The service is DSL and uses a speedstream 5200
dsl router. I sent her a Polycom IP300. At first it would not access the
config files via ftp so I had tech support walk her through setting the
phone's internal IP to be the dmz. This allowed me to set up the phone
using the web interface and now it registers. We had NAT problems so I
set the NAT features of the phone:
IP Address: 67.136.nnn.nnn
Signalling Port : 5060
Media Port Start: 10000
In sip.conf, I have
nat=yes
externalip=67.136.nnn.nnn
qualify=yes
I can call the user and she can hear me. If she calls me, no voice can
be heard either way. When I run sip show channels, I see:
Peer User/ANR Call ID Seq (Tx/Rx) Format Last Msg
192.168.0.169 805 02fe2b2a684 00102/00000 g729 Tx: ACK
67.136.nnn.nnn 893 8dae34ea-ae 00101/00001 g729 Rx: INVITE
67.136.nnn.nnn (None) 3926de51-a1 00101/00001 unknow Rx:
REGISTER
and it just stays like that until the call is terminated. I would think
it was an rtp / nat problem, any ideas how to fix?
--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int: (305) 704-7249 Fax: (815)301-9759
Cell: 264-235-5670
Yahoo IM: netconcepts_anguilla@yahoo.com
