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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