[Asterisk-Users] Cannot reach a SIP device

[Sudhir Kumar]

I am trying to call a my friend who has GS HandyTone-486 behind a
firewall but it goes to his voicemail straightway. Surprisingly, he can
call me fine. I also see that his device is properly registered. Can
anyone help me resolve this problem.

In my sip.conf I do have canreinvite=no and nat=yes.
In the GS HandyTone, he has set "use random port = yes" and 
"NAT traversal = yes"

When he calls me, there is no problem at all, audio is fine too.

Thanks,
-- sudhir


Here are some debug messages from the Server:
------------------------------------------------
cequip2*CLI> database show
     ......
/SIP/Registry/3110     : 168.243.154.92:63210:300:3110
     .....


------------------------------------------------
cequip2*CLI> sip debug ip 168.243.154.92
SIP Debugging Enabled for IP: 168.243.154.92

------------------------------------------------
After I call him from my extension:

Peer RTP is at port 192.168.2.4:0
    -- Executing Dial("SIP/4390-8620", "SIP/3110|15|rt") in new stack
We're at 66.251.6.188 port 10502
12 headers, 7 lines
Reliably Transmitting:
INVITE sip:3110 at 168.243.154.92:63210 SIP/2.0
Via: SIP/2.0/UDP 66.251.6.188:5060;branch=z9hG4bK3ac3b9e8
From: "Sudhir Kumar" <sip:4390 at 66.251.6.188>;tag=as009f251d
To: <sip:3110 at 168.243.154.92:63210>
Contact: <sip:4390 at 66.251.6.188>
Call-ID: 090d79152d43bf100374a2f630282ea3 at 66.251.6.188
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 15 Oct 2004 16:50:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 136

v=0
o=root 26933 26933 IN IP4 66.251.6.188
s=session
c=IN IP4 66.251.6.188
t=0 0
m=audio 10502 RTP/AVP
a=silenceSupp:off - - - -
 (NAT) to 168.243.154.92:63210
    -- Called 3110
cequip2*CLI> 

Sip read: 
SIP/2.0 415 
Via: SIP/2.0/UDP 66.251.6.188:5060;branch=z9hG4bK3ac3b9e8
From: "Sudhir Kumar" <sip:4390 at 66.251.6.188>;tag=as009f251d
To: <sip:3110 at 168.243.154.92:63210>;tag=9a40e4e5aafd25aa
Call-ID: 090d79152d43bf100374a2f630282ea3 at 66.251.6.188
CSeq: 102 INVITE
User-Agent: Grandstream HT486 1.0.5.10
Content-Length: 0


8 headers, 0 lines
    -- Got SIP response 415 "" back from 168.243.154.92
Transmitting:
ACK sip:3110 at 168.243.154.92:63210 SIP/2.0
Via: SIP/2.0/UDP 66.251.6.188:5060;branch=z9hG4bK3ac3b9e8
From: "Sudhir Kumar" <sip:4390 at 66.251.6.188>;tag=as009f251d
To: <sip:3110 at 168.243.154.92:63210>;tag=9a40e4e5aafd25aa
Contact: <sip:4390 at 66.251.6.188>
Call-ID: 090d79152d43bf100374a2f630282ea3 at 66.251.6.188
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (NAT) to 168.243.154.92:63210
  == No one is available to answer at this time
    -- Executing VoiceMail("SIP/4390-8620", "3110") in new stack
    -- Playing 'vm-intro' (language 'en')
Destroying call '090d79152d43bf100374a2f630282ea3 at 66.251.6.188'
  == Spawn extension (default, 3110, 2) exited non-zero on
'SIP/4390-8620'