acollard83
IS-IT--Management
We have an Asterisk server behind NAT. The nat router is a Cisco 2851. We are having one-way and no audio issues. Ports are forwarded correctly. The below SIP Trace shows something changing the ports. Any ideas where to look?
U 2013/04/23 22:26:31.411121 4.31.X.2:5060 -> 216.82.224.202:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKbc0e.c9dc9fd3.0;received=216.82.224.202.
Via: SIP/2.0/UDP 4.31.X.2:1195;branch=z9hG4bKbc0e.cba48a23.0.
Via: SIP/2.0/UDP 4.31.X.2:1197;branch=z9hG4bK0aB3e613c07c2399b58.
Record-Route: <sip:216.82.224.202;lr;ftag=gK0a019743>.
Record-Route: <sip:4.31.X.2:1195;lr=on;ftag=gK0a019743>.
From: "Unavailable" <sip:+19193225173@4.31.X.2:1197>;tag=gK0a019743. The system is adding this port 1197 onto the call and that is why we have no two way audio plus the calls drop.
To: <sip:+15175136709@4.31.X.2:1195>;tag=as56f12d93.
Call-ID: 353021783_1501442@4.55.10.97.
CSeq: 16852 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Contact: <sip:%2b15175136709@4.31.X.2:1680>.
Content-Type: application/sdp.
Content-Length: 236.
.
v=0.
o=root 12990 12990 IN IP4 4.31.X.2.
s=session.
c=IN IP4 4.31.X.2.
t=0 0.
m=audio 19486 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSuppff - - - -.
a=ptime:20.
a=sendrecv.
U 2013/04/23 22:26:31.411121 4.31.X.2:5060 -> 216.82.224.202:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKbc0e.c9dc9fd3.0;received=216.82.224.202.
Via: SIP/2.0/UDP 4.31.X.2:1195;branch=z9hG4bKbc0e.cba48a23.0.
Via: SIP/2.0/UDP 4.31.X.2:1197;branch=z9hG4bK0aB3e613c07c2399b58.
Record-Route: <sip:216.82.224.202;lr;ftag=gK0a019743>.
Record-Route: <sip:4.31.X.2:1195;lr=on;ftag=gK0a019743>.
From: "Unavailable" <sip:+19193225173@4.31.X.2:1197>;tag=gK0a019743. The system is adding this port 1197 onto the call and that is why we have no two way audio plus the calls drop.
To: <sip:+15175136709@4.31.X.2:1195>;tag=as56f12d93.
Call-ID: 353021783_1501442@4.55.10.97.
CSeq: 16852 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO.
Supported: replaces.
Contact: <sip:%2b15175136709@4.31.X.2:1680>.
Content-Type: application/sdp.
Content-Length: 236.
.
v=0.
o=root 12990 12990 IN IP4 4.31.X.2.
s=session.
c=IN IP4 4.31.X.2.
t=0 0.
m=audio 19486 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSuppff - - - -.
a=ptime:20.
a=sendrecv.