zanewoodfield
IS-IT--Management
I have an issue with a provider change over. The system is working fine for inbound calls, but our provider is trying to turn on source based routing of SIP at their end, but because the AVAYA presents the wrong IP address in the SIP FROM header, the outbound call breaks.
I have tried modifying ARS, short codes, LAN2 network topology firewall settings all to no avail. Does anyone know how I can get the correct IP address (the LAN2 address of the PBX) to show in the SIP FROM header?
See trace below:
Caller id and contact are fine but from header shows the ISTP Proxy address as the from as well as the TO.
53447mS SIP Call Tx: 17
INVITE sip:021612941@161.29.64.108 SIP/2.0
Via: SIP/2.0/UDP 161.29.192.153:5060;rport;branch=z9hG4bK3f7b2a53b2631df69002684c55e2e6e8
From: "039779340" <sip:039779340@161.29.64.108>;tag=6d206a27c8e94c25
To: <sip:021612941@161.29.64.108>
Call-ID: f573217d42b143c5ed913269a44836f2@161.29.192.153
CSeq: 1245346865 INVITE
Contact: "039779340" <sip:039779340@161.29.192.153:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Content-Length: 302
v=0
o=UserA 191639824 3596332152 IN IP4 161.29.192.153
s=Session SDP
c=IN IP4 161.29.192.153
t=0 0
m=audio 49152 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
I have tried modifying ARS, short codes, LAN2 network topology firewall settings all to no avail. Does anyone know how I can get the correct IP address (the LAN2 address of the PBX) to show in the SIP FROM header?
See trace below:
Caller id and contact are fine but from header shows the ISTP Proxy address as the from as well as the TO.
53447mS SIP Call Tx: 17
INVITE sip:021612941@161.29.64.108 SIP/2.0
Via: SIP/2.0/UDP 161.29.192.153:5060;rport;branch=z9hG4bK3f7b2a53b2631df69002684c55e2e6e8
From: "039779340" <sip:039779340@161.29.64.108>;tag=6d206a27c8e94c25
To: <sip:021612941@161.29.64.108>
Call-ID: f573217d42b143c5ed913269a44836f2@161.29.192.153
CSeq: 1245346865 INVITE
Contact: "039779340" <sip:039779340@161.29.192.153:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Content-Type: application/sdp
Supported: timer
Content-Length: 302
v=0
o=UserA 191639824 3596332152 IN IP4 161.29.192.153
s=Session SDP
c=IN IP4 161.29.192.153
t=0 0
m=audio 49152 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15