Hi everyone, I' m breaking my head trying to figure this out.
I have a Fortigate 60B and it seems to be modifying parts of the SIP Headers. I tried disabling " sip-helper" and " sip-nat-trace" but it does not seem to be helping. I also ensured that there is no protection profile on the firewall rule.
The specific headers that I' m talking about are the " From" " Contact" and " Anonymity" headers.
My provider (flowroute.com) claims that they sent the packet looking like this and anything missing must be being stripped out by my Firewall:
INVITE sip:************@************** SIP/2.0.
Record-Route: <sip:70.167.153.130;lr>.
Record-Route: <sip:216.115.69.142;lr>.
Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bK73c3.e613564d11025903295c92119be9b271.0.
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK73c3.883356619b22bdd3d8c07c5ecf0503bc.0.
Via: SIP/2.0/UDP 216.115.69.142;branch=z9hG4bK73c3.3d845433f085edc1e0ede5538e244ce9.0.
Via: SIP/2.0/UDP 4.55.14.163:5060;branch=z9hG4bK05B640b19b831a7f023.
From: " Private" <sip:+**************@4.55.14.163:5060;isup-oli=62>;tag=gK05565c95.
To: <sip:***************@216.115.69.142:5060>.
Call-ID: 1342517982_39619543@4.55.14.163.
CSeq: 4659 INVITE.
Max-Forwards: 14.
P-Asserted-Identity: <sip:+***************@pstn>.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: " Private" <sip:+***************@4.55.14.163:5060>.
Anonymity: name.
Content-Length: 228.
Content-Type: application/sdp.
.
v=0.
o=- 18884 1819 IN IP4 4.55.14.130.
s=-.
c=IN IP4 4.55.14.130.
t=0 0.
m=audio 7796 RTP/AVP 0 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=maxptime:20.
I am receiving it on the PBX looking like:
Notice that the parts in bold above are stripped from the INVITE
INVITE sip:************@************** SIP/2.0.
Record-Route: <sip:70.167.153.130;lr>.
Record-Route: <sip:216.115.69.142;lr>.
Via: SIP/2.0/UDP 70.167.153.130;branch=z9hG4bK73c3.e613564d11025903295c92119be9b271.0.
Via: SIP/2.0/UDP 216.115.69.131;branch=z9hG4bK73c3.883356619b22bdd3d8c07c5ecf0503bc.0.
Via: SIP/2.0/UDP 216.115.69.142;branch=z9hG4bK73c3.3d845433f085edc1e0ede5538e244ce9.0.
Via: SIP/2.0/UDP 4.55.14.163:5060;branch=z9hG4bK05B640b19b831a7f023.
From: <sip:+**************@4.55.14.163:5060;isup-oli=62>;tag=gK05565c95.
To: <sip:***************@216.115.69.142:5060>.
Call-ID: 1342517982_39619543@4.55.14.163.
CSeq: 4659 INVITE.
Max-Forwards: 14.
P-Asserted-Identity: <sip:+***************@pstn>.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed.
Contact: <sip:+***************@4.55.14.163:5060>.
Content-Length: 228.
Content-Type: application/sdp.
.
v=0.
o=- 18884 1819 IN IP4 4.55.14.130.
s=-.
c=IN IP4 4.55.14.130.
t=0 0.
m=audio 7796 RTP/AVP 0 18 101.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=maxptime:20.
If anyone can help me figure this out it would be greatly appreciated.
Thank You
