Hi,

I’m using Kamailio with presence enabled and Asterisk PJSIP and outbound-publish. My problem is happening when I place 2 consecutive calls from Asterisk :

When I make a first call Asterisk sent the following:

PUBLISH sip:201@192.168.100.37 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e69723a61082

From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf

To: sip:201@mydomain.com

Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183

CSeq: 10697 PUBLISH

Event: dialog

Expires: 180

Max-Forwards: 70

User-Agent: Asterisk PBX 14.6.0

Content-Type: application/dialog-info+xml

Content-Length: 247

<?xml version="1.0" encoding="UTF-8"?> early……

Kamailio replies :

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e69723a61082;received=192.168.100.37

From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf

To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-ff39

Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183

CSeq: 10697 PUBLISH

Expires: 180

SIP-ETag: a.1518775074.19863.16.0

Server: kamailio (5.0.5 (x86_64/linux))

Content-Length: 0

When the call is done, Asterisk sent another PUBLISH telling that the call if terminated :

PUBLISH sip:201@192.168.100.37 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97ab8c752

From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce

To: sip:201@mydomain.com

Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183

CSeq: 10698 PUBLISH

Event: dialog

SIP-If-Match: a.1518775074.19863.16.0

Expires: 180

Max-Forwards: 70

User-Agent: Asterisk PBX 14.6.0

Content-Type: application/dialog-info+xml

Content-Length: 230

<?xml version="1.0" encoding="UTF-8"?> terminated….

And Kamailio replies :

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97ab8c752;received=192.168.100.37

From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce

To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-48b4

Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183

CSeq: 10698 PUBLISH

Expires: 180

SIP-ETag: a.1518775074.19873.18.1

Server: kamailio (5.0.5 (x86_64/linux))

Content-Length: 0

Here, the SIP ETag is a.1518775074.19873.18.1.

The problem is if I make a new call before the expiration of the previous SUBSCRIBE, Asterisk reuse this SIP ETag according to the RFC :

PUBLISH sip:201@192.168.100.37 SIP/2.0

Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj9d13bb82-31d9-48db-9672-bd4b6b4f22f0

From: sip:201@mydomain.com;tag=33e6b028-0444-4b3a-8bc2-4a987a291528

To: sip:201@mydomain.com

Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183

CSeq: 10699 PUBLISH

Event: dialog

SIP-If-Match: a.1518775074.19873.18.1

Expires: 180

Max-Forwards: 70

User-Agent: Asterisk PBX 14.6.0

Content-Type: application/dialog-info+xml

Content-Length: 247

<?xml version="1.0" encoding="UTF-8"?> early…

Kamailio refuse it with this error : “Trying to update an already terminated state. Skipping update.” because the call is considered as terminated.

The RFC is stating :

When updating previously published event state, PUBLISH requests MUST

contain a single SIP-If-Match header field identifying the specific

event state that the request is refreshing, modifying or removing.

This header field MUST contain a single entity-tag that was returned

by the ESC in the SIP-ETag header field of the response to a previous

publication.

Why Kamailio is acting like that?

Best regards,

Cyrille