Hi,

But the INVITE seams to go twice through some NAT logic as I see the 
"a=nortpproxy:yes" header twice. Can you upload somewhere the entire SIP 
trace (inbound and outbound part) ?

Regards,
Bogdan

troxlinux wrote:
> Hi Romanov , I upgrade my opensips with the last version 1.5 but i
> have the same problem
>
> in asterisk cli,  I see that the opensips writes the address ip twice
>
>
> <--- SIP read from 192.168.10.3:5060 --->
> INVITE sip:[email protected]:5070 SIP/2.0
> Record-Route: <sip:192.168.10.3;lr=on;ftag=26f16c5a110a5bd9>
> Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bK3bba.4fa31861.1
> Via: SIP/2.0/UDP
> 192.168.10.30:5064;rport=5064;received=192.168.10.30;branch=z9hG4bKdbd6939c968835d1
> From: "Opensips-14x" <sip:[email protected]>;tag=26f16c5a110a5bd9
> To: <sip:[email protected]>
> Contact: <sip:[email protected]:5064;transport=udp>
> Supported: replaces, timer, path
> Call-ID: [email protected]
> CSeq: 47478 INVITE
> User-Agent: Grandstream GXP2020 1.1.6.44
> Max-Forwards: 69
> Allow: 
> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
> Content-Type: application/sdp
> Content-Length: 302
> P-hint: inbound->inbound
>
> v=0
> o=201 8000 8001 IN IP4 192.168.10.3192.168.10.3
> s=SIP Call
> c=IN IP4 192.168.10.3192.168.10.3
> t=0 0
> m=audio 3500635006 RTP/AVP 18 0 101
> a=sendrecv
> a=rtpmap:18 G729/8000
> a=rtpmap:0 PCMU/8000
> a=ptime:20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
> a=nortpproxy:yes
> a=nortpproxy:yes
>
>
>
> any idea , how solve this problem ?
>
> regardss
>
> 2009/4/11 Romanov Vladimir <[email protected]>:
>   
>> This is bug. I solve this problem by upgrading to 1.5.0
>>
>> -----------------
>>     
>
>   
> ------------------------------------------------------------------------
>
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