On Wed, Oct 12, 2016 at 3:51 PM, Serge Yuriev <me@nevian.org> wrote:

From perspective of server it is normal route via desired interface.
So VPN somewhere outside and server not aware of it.

--
Wbr, Serge via mobile

12.10.2016, 21:17, "Brian West" <brian@freeswitch.org>:

How are the two networks connected? VPN?

On Wed, Oct 12, 2016 at 12:01 PM, Serge S. Yuriev <me@nevian.org> wrote:
Hi,

How I can debug this issue to move it further?
I feel much more comfortable with FS than Asterisk which works out-of-box :)

Proxy mode doesn't work also because of 3pcc.
--
Wbr, Serge via mobile

11.10.2016, 14:25, "Serge S. Yuriev" <me@nevian.org>:

Hi

Is anyone have had chance to check the logs?

I tried to include 172.17.2.3 as local-network on external while excluding it from internal - no joy :(

<list name="lan" default="deny">
<node type="deny" cidr="172.17.2.3/32"/>
<node type="deny" cidr="172.17.2.4/32"/>
<node type="allow" cidr="192.168.0.0/16"/>
<node type="allow" cidr="10.0.0.0/8"/>
<node type="allow" cidr="172.16.0.0/12"/>
</list>

<list name="wan" default="deny">
<node type="allow" cidr="172.17.2.
3/32"/>
<node type="allow" cidr="172.17.2.
4/32"/>
<node type="allow" cidr="83.y.y.128/25"/>
</list>

--
Wbr, Serge via mobile

09.10.2016, 13:03, "Serge Yuriev":
Bad one
https://pastebin.freeswitch.org/view/5a6b306c

Good one
https://pastebin.freeswitch.org/view/5b1ca4e3

On 8 Oct 2016, at 04:23, Anthony Minessale <anthony.minessale@gmail.com> wrote:

Too terse.

You probably need to produce full traces on pastebin with the full debug to get any idea.

On Fri, Oct 7, 2016 at 6:13 PM, Serge Yuriev <me@nevian.org> wrote:
As mentioned before I tried to play with local-network-acl but no joy. Maybe it’s just not right? On which profile I should tune?

<list name="lan" default="deny">
<node type="deny" cidr="172.17.2.3/32"/>
<node type="deny" cidr="172.17.2.4/32"/>
<node type="allow" cidr="192.168.0.0/16"/>
<node type="allow" cidr="10.0.0.0/8"/>
<node type="allow" cidr="172.16.0.0/12"/>
</list>

On both profiles I have like this
Int
<param name="rtp-ip" value="$${inside_bind_ipv4}"/>
<param name="sip-ip" value="$${inside_bind_ipv4}"/>
<param name="ext-rtp-ip" value="$${inside_bind_ipv4}"/>
<param name="ext-sip-ip" value="$${inside_bind_ipv4}"/>

Ext
<param name="rtp-ip" value="$${outside_bind_ipv4}"/>
<param name="sip-ip" value="$${outside_bind_ipv4}"/>
<param name="ext-rtp-ip" value="$${outside_bind_ipv4}"/>
<param name="ext-sip-ip" value="$${outside_bind_ipv4}"/>

On 8 Oct 2016, at 00:48, Brian West <brian@freeswitch.org> wrote:

you have to fix your local-network-acl in each system probably to do the right thing, do you have the ext-rtp-ip set with the automat: prefix?

On Fri, Oct 7, 2016 at 1:23 PM, Serge S. Yuriev <me@nevian.org> wrote:
Hello,

Two SIP profiles:
External 83.хх
Internal 10.23.154.0/24

Via external we are receiving/send calls from/to 172.17.2.0/29
For some reason if we call outside FS sends unmodified addresses in SDP.
So we have unroutable address in SDP and one-way audio. If call flows
ext to int all working correct.
Tried local-network-acl on inside (10.хх) with excluded 172.хх,
apply-nat-acl with included 172.xx on either int and ext. Nothing helps :(

"Bad one" SDP - from internal to external
send 960 bytes to udp/[10.23.154.63]:6060 at 18:16:22.226984:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.23.154.63:6060;branch=z9hG4bKe433fa68b81
From: "IT, Юрьев Сергей"
<sip:12550@10.23.154.63>;tag=195594~27154efa-6325-45a2-9e47-67e5d9302ebc-237816120
To: <sip:62987%236546@10.23.154.100>;tag=66NUXXHvB6HBp
Call-ID: 86c80-7f71bc46-c44e-3f40000a@10.23.154.63
CSeq: 101 INVITE
Contact: <sip:mod_sofia@10.23.154.100:6060>
User-Agent:
FreeSWITCH-mod_sofia/1.7.0+git~20160707T165535Z~be13536ac9~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
PRACK, NOTIFY
Require: timer
Supported: precondition, 100rel, timer, path, replaces
Allow-Events: talk, hold, conference, refer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 180

v=0
o=- 1475853382 2 IN IP4 172.17.2.3
s=-
>> c=IN IP4 172.17.2.4
b=AS:64
t=0 0
m=audio 3040 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20

And a good one - external to internal
send 1162 bytes to udp/[10.23.154.65]:5060 at 12:34:15.132027:
------------------------------------------------------------------------
INVITE sip:12550@10.23.154.65 SIP/2.0
Via: SIP/2.0/UDP 10.23.154.100:6060;rport;branch=z9hG4bKUXyFjDmg8rtmB
Max-Forwards: 69
From: "Абонент"
<sip:$(caller_id_number)@10.23.154.100>;tag=1agg8aZ7FUUBK
To: <sip:12550@10.23.154.65>
Call-ID: d8367628-0fc1-4325-998f-3f32f9d3a05b
CSeq: 97580363 INVITE
Contact: <sip:gw+cucm-65@10.23.154.100:6060;transport=udp;gw=cucm-65>
User-Agent:
FreeSWITCH-mod_sofia/1.7.0+git~20160707T165535Z~be13536ac9~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
PRACK, NOTIFY
Supported: precondition, 100rel, timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 268
X-FS-Support: update_display,send_info
Remote-Party-ID: "Абонент"
<sip:$(caller_id_number)@10.23.154.100>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1475804423 1475804424 IN IP4 10.23.154.100
s=FreeSWITCH
>> c=IN IP4 10.23.154.100
t=0 0
m=audio 28432 RTP/AVP 8 18 101 13
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:20