You have invalid configurating and this causes congestions or impossible to
forward the calls.
Please use those rules to configure correctly your mapping:
For URI Mapping (URI field):
SIP
1. 192.168.1.90
2. 192.168.1.90/12345
3. username@192.168.1.90
4. username@192.168.1.90/12345
Dial format:
SIP/192.168.1.90
SIP/192.168.1.90/12345
SIP/username@192.168.1.90
SIP/username@192.168.1.90/12345
IAX:
1. 192.168.1.90
2. 192.168.1.90/12345
3. username:password@192.168.1.90
4. username:password@192.168.1.90/12345
Dial format:
IAX2/192.168.1.90
IAX2/192.168.1.90/12345
IAX2/username:password@192.168.1.90
IAX2/username:password@192.168.1.90/12345
For Custom mapping:
SIP
1. IP or domain field: 192.168.1.90
Custom DID: nothing
2. IP or domain field: 192.168.1.90
Custom DID: Any other number (for ex: 12345)
3. IP or domain field: username@192.168.1.90
Custom DID: Any other number (12345)
Dial format:
SIP/192.168.1.90
SIP/192.168.1.90/12345
SIP/username@192.168.1.90/12345
IAX2
1. IP or domain field: 192.168.1.90
Custom DID: nothing
2. IP or domain field: 192.168.1.90
Custom DID: Any other number (12345)
3. IP or domain field: username:password@192.168.1.90
Custom DID: nothing
4. IP or domain field: username:password@192.168.1.90
Custom DID: Any other number (12345)
Dial format:
IAX2/192.168.1.90
IAX2/192.168.1.90/12345
IAX2/username:password@192.168.1.90
IAX2/username:password@192.168.1.90/12345
Username could be: word, account number, phone number
Custom DID is used like Caller ID. Instead of sending Callee number ( source)
the custom DID change the source to new value
For example, if the field Custom DID is empty then the dial format will be :
SIP/192.168.1.90/SOURCENUMBER