Difference between revisions of "How to make several connections from one IP"

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Line 24: Line 24:
  INVITE sip:3908155xxxxx@82.215.xxx.xxx SIP/2.0
  INVITE sip:3908155xxxxx@82.215.xxx.xxx SIP/2.0
  Via: SIP/2.0/UDP 82.215.xxx.xxx:5060;branch=z9hG4bK50ef8b9b;rport
  Via: SIP/2.0/UDP 82.215.xxx.xxx:5060;branch=z9hG4bK50ef8b9b;rport
  From: "39081761xxxxx" <sip:<big>'''1028'''</big>@82.215.xxx.xxx>;tag=as3b117c95
  From: "39081761xxxxx" <sip:<big><big><big>'''1028'''</big></big></big>@82.215.xxx.xxx>;tag=as3b117c95
  To: <sip:390815517304@82.215.xxx.xxx>
  To: <sip:390815517304@82.215.xxx.xxx>
  Contact <sip:1028@82.215.xxxx.xxx>
  Contact <sip:1028@82.215.xxxx.xxx>

Revision as of 12:51, 19 February 2010

Description

Sometimes it is necessary to make several connections from one IP (server usually) to MOR to separate traffic.

Main rules how to do that:

  1. Devices should be DYNAMIC and should REGISTER to MOR
  2. In INVITE packet from external server in line From: should be visible, which device is dialing. (If external server is Asterisk, then in MOR descriptions (in sip.conf file) on external server, should be fromuser= lines which points which device is dialing - check explanation in details below)

If devices are not configured properly, then call from external server to MOR is possible, but only 1 device will be billed all the time or only 1 trunk will be active (depends on misconfiguration).

Possible error on misconfiguration:

WARNING[2836]: chan_sip.c:8385 check_auth: username mismatch, have <XXXX>, digest has <YYYY>




SIP INVITE packet

As described previously correct SIP INVITE packet should be received in order for MOR to correcly authenticate user.

Good SIP INVITE packet

INVITE sip:3908155xxxxx@82.215.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 82.215.xxx.xxx:5060;branch=z9hG4bK50ef8b9b;rport
From: "39081761xxxxx" <sip:1028@82.215.xxx.xxx>;tag=as3b117c95
To: <sip:390815517304@82.215.xxx.xxx>
Contact <sip:1028@82.215.xxxx.xxx>
Call-ID: 2f6f09f8498065164a9aadae5ef123cf@82.215.xxx.xxx
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 19 Feb 2010 12:28:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 242

Bad SIP INVITE packet



Example