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[lei.tlfly at gmail.com]

Hi all, I run FS on a machine with two net interface, each interface has a ip addr, one of the them connect to public network(has ip addr A), the other  connect to a private network(has ip addr B), FS server as a SIP server for public through A, all outbound call will bridge to a softswitch in private network through B. here is my sofia config file and diaplan config:

sofia internal.xml
....
<param name="rtp-ip" value="A"/>
<param name="sip-ip" value="A"/>
 ....

sofia external.xml
....
<param name="rtp-ip" value="B"/>
<param name="sip-ip" value="B"/>....

dialplan
......
<extension name="OUTBOUND">
    <condition field="destination_number" expression="^(\d+)$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
        <action application="set" data="effective_caller_id_number=xxxxxxx"/>  <!--here change the caller number -->
        <action application="bridge" data="sofia/external/${destination_number}@xxxxx"/>
      </condition>
    </extension>
.....

then call seq is
sipAgent --> [internal -->(bridge)-->external] -->softswith
                          FREESWITCH

the question is, when sipAgent make a outbound call, FS can't recevie the caller's up audio stream, I traced the SIP packets, found that FS has return addr B in SDP when ack the invite request from sipAgent, the ack packet is
===============
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP xxxxx:12208;branch=z9hG4bK-d8754z-dc750d57652c7c51-1---d8754z-;rport=12208
From: "1000" <sip:xxxx@A>;tag=cb4d3c4e
To: "65960581" <sip:xxxx@A>;tag=DtvSc0QX01yKN
Call-ID: ZTI2NmIwZGZiYzlhOGNkNTdiYWUzMzkzZTMwYzgxZjI.
CSeq: 2 INVITE
Contact: <sip:xxxxxx@B:5060;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 245

v=0
o=FreeSWITCH 1256598185 1256598186 IN IP4 B   ;>>>>wrong this is the ip addr of the adapter connect to the private network
s=FreeSWITCH
c=IN IP4 B ;>>>>wrong this is the ip addr of the adapter connect to the private network
t=0 0
m=audio 31066 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
================
I think FS should return A in SDP, not the external binding addr (B), does somebody known how to solve this problem?

--
Lei.Tang
lei.tlfly@gmail.com (lei.tlfly@gmail.com)

[egable+freeswitch at g...]

Try setting ext-rtp-ip and ext-sip-ip on both profiles.

On Tue, Oct 27, 2009 at 4:49 AM, Lei Tang <lei.tlfly@gmail.com> wrote:

[lei.tlfly at gmail.com]

Thanks Eliot, It works.

2009/10/27 Eliot Gable <egable+freeswitch@gmail.com ([email]egable%2Bfreeswitch@gmail.com[/email])>

[brian at freeswitch.org]

you should have just needed to set the rtp-ip and the sip-ip on both
profiles to their values and it would have worked fine... their would
have been no need to set the ext-*-ip equiv.

/b

On Oct 27, 2009, at 8:14 PM, Lei Tang wrote:

[lei.tlfly at gmail.com]

Hi Brian, It doesn't work if I only set rtp-ip and sip-ip. when  I set the ext-rtp-ip, it works fine. 

2009/10/28 Brian West <brian@freeswitch.org (brian@freeswitch.org)>

[brian at freeswitch.org]

Ok then something is broken badly... and that makes NO sense. Because it works on my box.

/b

On Oct 27, 2009, at 8:34 PM, Lei Tang wrote: