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 thisInt<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
--
Serge S. Yuriev
Senior VoIP engineer
____________________________________________________________ _____________
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____________________________________________________________ _____________
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