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How to set up beeline sip in FreePBX?
Good afternoon, colleagues! There is an Internet service from Beeline, in addition to it, a sip-number is given by a separate vilan.
Sip works great on the Grandstream GXP1400 phone. And on FreePBX I can not find the correct settings. The maximum that was achieved is an outgoing call and a sound in one direction.
The provider is impenetrable in terms of recommended settings for a virtual PBX ..
From the connection data we have:
IP - 10.30.6.178
Gateway - 10.30.6.177
Mask - 255.255.255.252
Port - 5060
Sip server - 10.30.0.1
Login without password 1111234567
Phone number 234-567
The route on FreePBX is static, the host pings, and as I wrote above, there is an outgoing call and sound in the same direction.
How to properly analyze what ATS-ke is missing?
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Beeline example
PEER DETAILS
username=SIPblabla - SIP USER !!! Differs from fromuser always goes separately.
type=peer
transport=tcp
secret=password
port=5060
nat=yes
insecure=port,invite
host=195.239.174.100 - sip proxy
fromuser=SIPblabla Always comes in the form [email protected]
fromdomain=mpbx.sip.beeline.ru - this is domain not proxy Bla .ru
dtmfmode=inband
disallow=all
context=from-trunk
allow=alaw
USER DETAILS
type=user
secret=password
context=from-trunk
Registration. Always comes first [email protected], password, and then [email protected]
[email protected]:password:[email protected]:5060/SIPblabla
Provider provided
beeline
SIP settings User ID SIPblabla
domain mpbx.sip.beeline.ru
Authorization User ID [email protected]
SIP proxy 195.239.174.100
Password:
this is with authorization, as without authorization, xs
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