Hi,
have a setup which should not be unknown to others;
Asterisk behind wall doing NAT, and out in the wild world behind linksys
router a Polycom phone. The Polycom phone is on DMZ. It should register
with my server.
sip conf:
[4031]
type=friend
context=main
callerid="HJEMME" <4031>
secret=4031
nat=yes
canreinvite=yes
qualify=yes
host=dynamic
dtmfmode=rfc2833
username=4031
mailbox=4031@main
disallow=all
allow=ulaw
allow=gsm
progressinband=no
I can dial and the phone rings,caller ID comes up, both ways this is
working.
Now answering, gives no sound, or just a fraction of a second of sound.
Voicemail, greetings etc appears to use ports above 10000, these seems
to pass ok.
Debugging RTP, which I read is where the audio is passed,and which has
been opened on server side gateway to ports 10000-20000, showed packets
to the Polycom address on ports 29xxx, whereas I opened rtp ports upto
30000.
No change, below the initiating sequence of a call being answered.
-- SIP/4031-2264 answered SIP/4030-1c2e
-- Attempting native bridge of SIP/4030-1c2e and
SIP/4031-2264
Got RTP packet from 10.1.0.51:10080 (type 0, seq 39101, ts
368481408, len 160)
Sent RTP packet to xx.xx.x.xxx:29256 (type 0, seq 35268, ts -32,
len 160)
Got RTP packet from 10.1.0.51:10080 (type 0, seq 39102, ts
368481568, len 160)
Sent RTP packet to xx.xx.x.xxx:29256 (type 0, seq 35269, ts 128,
len 160)
Got RTP packet from xx.xx.x.xxx:29258 (type 0, seq 7541, ts
1008244723, len 160)
Sent RTP packet to 10.1.0.51:10080 (type 0, seq 19971, ts -80,
len 160)
I am a bit confused about the settings of RTP on the Polycom 600, could
be something there.
Anyone that could get me on track?
regards
Frank
On Tue, 2005-05-03 at 10:02 +0200, list wrote:> Hi, > have a setup which should not be unknown to others; > Asterisk behind wall doing NAT, and out in the wild world behind linksys > router a Polycom phone. The Polycom phone is on DMZ. It should register > with my server. > sip conf: > [4031] > type=friend > context=main > callerid="HJEMME" <4031> > secret=4031 > nat=yes > canreinvite=yes<snip> I have a similar setup, which works fine, but I use "canreinvite=no" -Tor
Hei, thanks that was it... Could bet I did try that earlier, but now its working. Actually I did manage to join the 2 phones in conference but not directly. Interesting. thanks again On Tue, 2005-05-03 at 17:34, Tor Setane wrote:> On Tue, 2005-05-03 at 10:02 +0200, list wrote: > > Hi, > > have a setup which should not be unknown to others; > > Asterisk behind wall doing NAT, and out in the wild world behind linksys > > router a Polycom phone. The Polycom phone is on DMZ. It should register > > with my server. > > sip conf: > > [4031] > > type=friend > > context=main > > callerid="HJEMME" <4031> > > secret=4031 > > nat=yes > > canreinvite=yes > <snip> > > I have a similar setup, which works fine, but I use "canreinvite=no" > > -Tor > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users