Hi,
This is first time I've had to ask for help so be gentle. We have configured a registered SIP trunk but are intermittently sending out calls with anonymous@anonymous.invalid in the from field as well as anonymous.invalid in the realm field interchangeably. I can't for the life of me figure out why it seems to happen randomly as users are configured with a SIP user and the trunk is set to use internal data. Due to this the ITSP will reject calls with a 403 reply. Software is release 11. Any ideas/suggestions?
09:11:01 18186446mS SIP Tx: UDP 192.168.1.249:5060 -> 178.255.63.75:5060
INVITE siputbound-number@blah.blah SIP/2.0
Via: SIP/2.0/UDP 192.168.1.249:5060;rport;branch=z9hG4bKb296f257e323accbc47538176a2f9a30
From: "user" <sip:user@blah.blah>;tag=fdf23f007344a9ad
To: <siputbound-number@blah.blah>
Call-ID: 7dbd1068906422b4b18b9c63d289d82a
CSeq: 2117248170 INVITE
Contact: "2000" <sip:user@192.168.1.249:5060;transport=udp>
Proxy-Authorization: Digest username="user",realm="[highlight #FCE94F]anonymous.invalid[/highlight]",nonce="6b3c188c-90cd-483c-bb5d-dc7da1f3ae46",response="9efcaababe06b2276307c665d8beeffe",uri="siputbound-number@blah.blah",algorithm=MD5,qop=auth,nc=0000
0004,cnonce="b832f157ec7d52bbf9e4"
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 11.0.4.4.0 build 6
Content-Type: application/sdp
Content-Length: 301
v=0
o=UserA 3224913659 1783336038 IN IP4 192.168.1.249
s=Session SDP
c=IN IP4 192.168.1.249
t=0 0
m=audio 46750 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
Thanks,
Al
This is first time I've had to ask for help so be gentle. We have configured a registered SIP trunk but are intermittently sending out calls with anonymous@anonymous.invalid in the from field as well as anonymous.invalid in the realm field interchangeably. I can't for the life of me figure out why it seems to happen randomly as users are configured with a SIP user and the trunk is set to use internal data. Due to this the ITSP will reject calls with a 403 reply. Software is release 11. Any ideas/suggestions?
09:11:01 18186446mS SIP Tx: UDP 192.168.1.249:5060 -> 178.255.63.75:5060
INVITE siputbound-number@blah.blah SIP/2.0
Via: SIP/2.0/UDP 192.168.1.249:5060;rport;branch=z9hG4bKb296f257e323accbc47538176a2f9a30
From: "user" <sip:user@blah.blah>;tag=fdf23f007344a9ad
To: <siputbound-number@blah.blah>
Call-ID: 7dbd1068906422b4b18b9c63d289d82a
CSeq: 2117248170 INVITE
Contact: "2000" <sip:user@192.168.1.249:5060;transport=udp>
Proxy-Authorization: Digest username="user",realm="[highlight #FCE94F]anonymous.invalid[/highlight]",nonce="6b3c188c-90cd-483c-bb5d-dc7da1f3ae46",response="9efcaababe06b2276307c665d8beeffe",uri="siputbound-number@blah.blah",algorithm=MD5,qop=auth,nc=0000
0004,cnonce="b832f157ec7d52bbf9e4"
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
User-Agent: IP Office 11.0.4.4.0 build 6
Content-Type: application/sdp
Content-Length: 301
v=0
o=UserA 3224913659 1783336038 IN IP4 192.168.1.249
s=Session SDP
c=IN IP4 192.168.1.249
t=0 0
m=audio 46750 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
Thanks,
Al