SIP Contact URI showing sip:asterisk@ipaddress

Asterisk Asterisk SIP
1

Hi,
I have two sip endpoints 6000 and 60001. Usually ip and port of contact field gets changed by asterisk but when I am trying to call from 6000 to 6001 the username also getting changed. Is this normal behaviour of asterisks or there is any configuration issue.

I am using kamailio and asterisk and BYE request is not getting forwarded to asterisk. After searching in internet I found out this issue occurs if there is a mismatch in contact field of INVITE and BYE request. I suspect this is the reason for loss of BYE message.

This is the INVITE request sent from browser:

INVITE sip:6000@34.215.xxx.xx SIP/2.0
Record-Route: sip:34.215.xxx.xx:5060;ftag=rTRvLtczwe9uTRiARptL;lr=on
Via: SIP/2.0/UDP 34.215.xxx.xx:5060;branch=z9hG4bK800a.ac7dd2aba9fc7384bdb4e7852f8cc6b9.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;received=103.211.xx.xx;branch=z9hG4bKSdgIzvVFw4ARJZk2JRNvySg77Ww88hQP;rport=33336
From: "6001"sip:6001@34.215.xxx.xx;tag=rTRvLtczwe9uTRiARptL
To: sip:6000@34.215.xxx.xx
Contact: "6001"sips:6001@df7jal23ls0d.invalid;alias=103.211.xx.xx~33336~6;rtcweb-breaker=no;click2call=no;transport=wss;+g.oma.sip-im;language=“en,fr”
Call-ID: 2d5ec1a6-5f07-967c-4969-59354e6bca75
CSeq: 61376 INVITE
Content-Type: application/sdp
Content-Length: 2053
Max-Forwards: 69
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

This is the INVITE request sent by asterisk:

INVITE sip:6000@34.215.xxx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.xx.xxx:5060;rport;branch=z9hG4bKPjae24d281-7715-46fa-bdc0-ee8286d4db62
From: “6001” sip:6001@172.31.xx.xxx;tag=3733e901-f1e7-4b19-af0b-f2f3224c4b41
To: sip:6000@34.215.xxx.xx
Contact: sip:asterisk@172.31.xx.xxx:5060
Call-ID: eef214cf-2fb9-4503-be5b-942a0acef0ba
CSeq: 20322 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 15.5.0
Content-Type: application/sdp
Content-Length: 1003

As we can see in contact field asterisk is present.Is it not supposed to be from user(6001).

2

There is nothing wrong in your logs, as far as the user part of the Contact header goes. You may have a NAT misconfiguration for the IP part.

The suer part is not supposed to be anything in particular. Asterisk is the only system that should care about it, as its only purpose is to get the request to the right place in Asterisk. Whilst chan_sip used to send a caller ID or extension number, chan_pjsip sends Asterisk. They could, quite legally, send some random string.