Difference between revisions of "How to make several connections from one IP"
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(New page: = 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 an...) |
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Main rules how to do that: | Main rules how to do that: | ||
# Devices should be DYNAMIC and should REGISTER to MOR | |||
# 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> | |||
<br><br> | |||
= 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:<big>'''1028'''</big>@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 == | |||
<br><br> | |||
= Example = |
Revision as of 12:50, 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:
- Devices should be DYNAMIC and should REGISTER to MOR
- 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