AVAYA IP Office SIP trunk anonymous.invalid problem

[Swankytmc]

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

 

[Swankytmc]

I've spoken to the provider and it's definitely the realm field causing problems. Which settings in IPO can allows me to update this or at least alleviate the problem?

 

[Swankytmc]

Hi,

Please see the engineering doc from the SIP provider. Still struggling to get the fields correct.

Thanks,

Al

 

[G van Hamburg]

Hi Al,

I can’t see the problem from looking at the document. The provider does not seem to care about PAI etc. Once you can athenticate for an outside call you are good to go. That brings me back to how I know authentication. (Also described in the document)

When you make an outside call the invite is sent and the provider answers with a 407 and a nonce. IPO sends the invite again but this time with username and password plus a reponse on the nonce.

Can you make a new good call and a bad call and show the complete SIP trace from both calls? I’d like to see the initial invite and the 407.

PS: When this gettting crucial, you can the provider to change to IP based. That way the register is not needed.

Freelance Certified Avaya Aura Engineer

 

[Swankytmc]

After 6 days of settled activity, this problem has returned. Looks like it's the realm field. I really need to work out how to hard configure this to remain static. Any ideas?


09:39:51 3130593128mS SIP Tx: UDP 192.168.1.249:5060 -> y.y.y.y:5060
INVITE sip:++44xxxxxxxxxxx@childwall.hnsaccess.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.249:5060;rport;branch=z9hG4bK8a8d353d285b8cbc07e11cc813bc13f7
From: "2000@sipprovider" <sip:2000@sipprovider>;tag=b327e6c843ec97f7
To: <sip:+44xxxxxxxxx@sipprovider>
Call-ID: 63d79a8d26ec50c7cca5b57a9b9f8f8b
CSeq: 973518607 INVITE
Contact: "2000@sipprovider" <sip:*@sipprovider:5060;transport=udp>
Proxy-Authorization: Digest username="2000",realm="anonymous.invalid",nonce="b78e83f1-0ba8-46be-be4a-01315e89f085",response="94d9d236143f1db28b501c3a45ccb311",uri="sip:+44xxxxxxxxx@sipprovider",algorithm=MD5,qop=auth,nc=0

 

[Swankytmc]

noticed that when outbound fails, it affects all handsets for a specific time period. I checked logs and noticed failed calls had this set:

Pres=Restricted (1)

How do I get past this? Could it be adding an A to the shortcode to force calls to be non-restricted?

Thanks,

SW

 

[derfloh]

Anonymous is configured in the phone's feature menu. Not reflected in Manager.

IP Office remote service
IP Office certificate check
CLI based call blocking
SCN fallback over PSTN

 

[Swankytmc]

Derfloh, I can't see how that can be the case. All handsets are fine and then suddenly al handsets start having the problem. I've been through and number withheld not activated.

 

[critchey]

OK just noticed something... you are using "use internal data"... but your SIP tab is set as 2000? Why are you using internal data but not setting the SIP tab up correctly for the user? You should be presenting a CallerID that is present on your account not your extension number. Have you tried setting up the SIP tab correctly for a user and then testing with that user?

The truth is just an excuse for lack of imagination.

 

[Efixka]

Hello,
in the advanced tabs of your SIP line
just tick "envoyer <<from>> en clair" (sorry i don't know the english translate)
and your realm will become your domain