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у меня несколько регистраций на телфине, а все входящие звонки валятся через 1... Вот мои настройки:

[74951111111]

defaultuser=1111111
secret=qwerty
type=peer
disallow=all
allow=ulaw,alaw
qualify=yes
dtmfmode=rfc2833
canreinvite=no
registersip=yes
nat=yes
promiscredir=yes
insecure=port,invite
host=sip.telphin.com
port=5068
fromuser=2222222
fromdomain=sip.telphin.com
context=from-trunk

1111111:qwerty@sip.telphin.com:5068/1111111

[74952222222]
type=2222222
secret=qwerty
registersip=yes
qualify=yes
promiscredir=yes
port=5068
nat=yes
insecure=port,invite
host=sip.telphin.com
fromuser=2222222
fromdomain=sip.telphin.com
dtmfmode=rfc2833
disallow=all
defaultuser=peer2
context=from-trunk
canreinvite=no
allow=ulaw,alaw
username=2222222

2222222:qwerty@sip.telphin.com:5068/2222222

sip.telphin.com:5068                    N      2222222         1785 Registered           Tue, 03 Mar 2015 17:47:56
sip.telphin.com:5068                    N      1111111         1785 Registered           Tue, 03 Mar 2015 17:47:56

Звоню я с номера +74953333333 на номер +74952222222

<--- SIP read from UDP:213.170.92.166:5068 --->
INVITE sip:peer2@A.B.C.D:5060 SIP/2.0
Record-Route: <sip:74952222222@213.170.92.166:5068;r2=on;lr=on;ftag=1c1891928713;ldi=62a.1488;tfs=AAAAABMBCwQCCwQAAQ90BHFAR0EZREtVQUZbXhxjb206NTA2OA-->
Record-Route: <sip:74952222222@213.170.92.166;r2=on;lr=on;ftag=1c1891928713;ldi=62a.1488;tfs=AAAAABMBCwQCCwQAAQ90BHFAR0EZREtVQUZbXhxjb206NTA2OA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1891928713;did=62a.4aa3ba9>
Via: SIP/2.0/UDP 213.170.92.166:5068;branch=z9hG4bKd5a.d59656d869beafa388b6a9563b648183.0
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bKd5a.ee674594.0
Via: SIP/2.0/UDP 213.170.91.202:5060;branch=z9hG4bKac1891945065
Max-Forwards: 69
From: +74953333333 <sip:+74953333333@sip.telphin.com:5068>;tag=1c1891928713
To: <sip:74952222222@213.170.100.150;user=phone>
P-Preferred-Identity: <sip:+74953333333@sip.telphin.com:5068>
Call-ID: 1891927311169201113169@213.170.91.202
CSeq: 1 INVITE
Contact: <sip:84953333333@213.170.91.202:5060>
Supported: em,timer,replaces,path,resource-priority,sdp-anat
Content-Type: application/sdp
Content-Length: 308

v=0
o=AudiocodesGW 1891879912 1891879911 IN IP4 213.170.91.202
s=Phone-Call
c=IN IP4 213.170.91.202
t=0 0
m=audio 8780 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 14 lines) ---
Sending to 213.170.92.166:5068 (no NAT)
Sending to 213.170.92.166:5068 (no NAT)
Using INVITE request as basis request - 1891927311169201113169@213.170.91.202
Found peer '74951111111' for '+74953333333' from 213.170.92.166:5068
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5

Где я делаю ошибку в настройках?

у меня несколько регистраций на телфине, а все входящие звонки валятся через 1... Вот мои настройки:

[74951111111]

defaultuser=1111111
secret=qwerty
type=peer
disallow=all
allow=ulaw,alaw
qualify=yes
dtmfmode=rfc2833
canreinvite=no
registersip=yes
nat=yes
promiscredir=yes
insecure=port,invite
host=sip.telphin.com
port=5068
fromuser=2222222
fromdomain=sip.telphin.com
context=from-trunk

1111111:qwerty@sip.telphin.com:5068/1111111

[74952222222]
type=2222222
secret=qwerty
registersip=yes
qualify=yes
promiscredir=yes
port=5068
nat=yes
insecure=port,invite
host=sip.telphin.com
fromuser=2222222
fromdomain=sip.telphin.com
dtmfmode=rfc2833
disallow=all
defaultuser=peer2
context=from-trunk
canreinvite=no
allow=ulaw,alaw
username=2222222

2222222:qwerty@sip.telphin.com:5068/2222222

sip.telphin.com:5068                    N      2222222         1785 Registered           Tue, 03 Mar 2015 17:47:56
sip.telphin.com:5068                    N      1111111         1785 Registered           Tue, 03 Mar 2015 17:47:56

Звоню я с номера +74953333333 на номер +74952222222

<--- SIP read from UDP:213.170.92.166:5068 --->
INVITE sip:peer2@A.B.C.D:5060 sip:2222222@A.B.C.D:5060 SIP/2.0
Record-Route: <sip:74952222222@213.170.92.166:5068;r2=on;lr=on;ftag=1c1891928713;ldi=62a.1488;tfs=AAAAABMBCwQCCwQAAQ90BHFAR0EZREtVQUZbXhxjb206NTA2OA-->
Record-Route: <sip:74952222222@213.170.92.166;r2=on;lr=on;ftag=1c1891928713;ldi=62a.1488;tfs=AAAAABMBCwQCCwQAAQ90BHFAR0EZREtVQUZbXhxjb206NTA2OA-->
Record-Route: <sip:213.170.100.150;lr;ftag=1c1891928713;did=62a.4aa3ba9>
Via: SIP/2.0/UDP 213.170.92.166:5068;branch=z9hG4bKd5a.d59656d869beafa388b6a9563b648183.0
Via: SIP/2.0/UDP 213.170.100.150:5060;branch=z9hG4bKd5a.ee674594.0
Via: SIP/2.0/UDP 213.170.91.202:5060;branch=z9hG4bKac1891945065
Max-Forwards: 69
From: +74953333333 <sip:+74953333333@sip.telphin.com:5068>;tag=1c1891928713
To: <sip:74952222222@213.170.100.150;user=phone>
P-Preferred-Identity: <sip:+74953333333@sip.telphin.com:5068>
Call-ID: 1891927311169201113169@213.170.91.202
CSeq: 1 INVITE
Contact: <sip:84953333333@213.170.91.202:5060>
Supported: em,timer,replaces,path,resource-priority,sdp-anat
Content-Type: application/sdp
Content-Length: 308

v=0
o=AudiocodesGW 1891879912 1891879911 IN IP4 213.170.91.202
s=Phone-Call
c=IN IP4 213.170.91.202
t=0 0
m=audio 8780 RTP/AVP 8 0 18 101
a=ptime:20
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 14 lines) ---
Sending to 213.170.92.166:5068 (no NAT)
Sending to 213.170.92.166:5068 (no NAT)
Using INVITE request as basis request - 1891927311169201113169@213.170.91.202
Found peer '74951111111' for '+74953333333' from 213.170.92.166:5068
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5

Где я делаю ошибку в настройках?