Добрый день прошу помочь с настройкой (разобраться в проблеме) с DVG 7062. Не идут звонки по направлению с DVG 7062 на MyPBX 1600 а в обратную сторону всё ок.

MyPBX отписывает сл:

<--- SIP read from 192.168.17.252:5060 --->
INVITE sip:2104@192.168.21.252;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 192.168.17.252:5060;branch=z9hG4bK1438efee550e6525
From: <sip:1720@192.168.17.252>;tag=88757ccf-684903
To: <sip:2104@192.168.21.252;user=phone>
Call-ID: D1B9-221E-4668490331F038F9CBAB-002@SipHost
CSeq:2 INVITE
Contact: <sip:1720@192.168.17.252:5060>
Expires:90
Max-Forwards:70
Supported: replaces
User-Agent:dlink 12-38-43929367-0.9.5.1.1928-PBX6S
Content-Type: application/sdp
Content-Length: 328

v=0
o=1720 1792097970 1792097970 IN IP4 192.168.17.252
s=Session SDP
c=IN IP4 192.168.17.252
t=0 0
m=audio 9004 RTP/AVP 18 0 2 8 4 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

<------------->
--- (14 headers 14 lines) ---
Sending to 192.168.17.252 : 5060 (no NAT)
Using INVITE request as basis request - D1B9-221E-4668490331F038F9CBAB-002@SipHo st
Found peer 'trunk-sps-Pogranichnaya'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 192.168.17.252:9004
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x90d (g723|ulaw|ala w|g726|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone -event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.17.252:9004
Looking for 2104 in DID_inbound_trunk-sps-Pogranichnaya (domain 192.168.21.252)

<--- Reliably Transmitting (no NAT) to 192.168.17.252:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.17.252:5060;branch=z9hG4bK1438efee550e6525;received=192 .168.17.252
From: <sip:1720@192.168.17.252>;tag=88757ccf-684903
To: <sip:2104@192.168.21.252;user=phone>;tag=as2822da57
Call-ID: D1B9-221E-4668490331F038F9CBAB-002@SipHost
CSeq: 2 INVITE
User-Agent: asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
[Oct 21 08:17:14] NOTICE[321]: chan_sip.c:14166 handle_request_invite: Call from 'trunk-sps-Pogranichnaya' to extension '2104' rejected because extension not found.


При этом звонки с MyPBX и на неё идут отлично на этот же номер с Cisco 2821..

Прошу помочь


лог slmon

Line DialNum_New [212104]
15:23:58 [000785] 2: OutPinCode=<>, dial_no=<212104>
15:23:58 [000785] 2: PhoneBook has GwNo=2104, addr=192.168.21.252:5060
15:23:58 [000785] 2: ==7:Inviting
15:23:59 [000785] 2: VoIP CallOut, Invite <sip:2104@192.168.21.252;user=phone>
15:23:59 [000787] 2: 1720=DIALING
15:23:59 [000788] 2: 1720=DISCONNECT
15:23:59 [000788] 2: Release Active Dlg
15:23:59 [000789] 2: PlayFile(/opt/voc/NotifyVoip.pcm)=0
15:23:59 [000789] 2: ### NewStatus Overlap: PlayBusyTone->PlayDisconnected ###
15:23:59 [000789] 2: ==25:PlayDisconnected
15:24:00 [000804] 2: ==15:PlayBusyTone
15:24:03 [000832] DSP_ch2_check=0
15:24:03 [000832] 2: Fxs Hangup
15:24:03 [000832] 2: ==3:Idle
15:24:03 [000832] 2: SetFax(1)=0
15:24:03 [000835] 2: Fxs Pickup!
15:24:03 [000835] 2: ==4:GetDtmf
15:24:12 [000917] 2: DTMF Timeout #1
15:24:12 [000917] 2: End Input Route [1]
15:24:12 [000917] 2: Line DialNum_New [212104]
15:24:12 [000917] 2: OutPinCode=<>, dial_no=<212104>
15:24:12 [000917] 2: PhoneBook has GwNo=2104, addr=192.168.21.252:5060
15:24:12 [000917] 2: ==7:Inviting
15:24:12 [000917] 2: VoIP CallOut, Invite <sip:2104@192.168.21.252;user=phone>
15:24:12 [000919] 2: 1720=DIALING
15:24:12 [000920] 2: 1720=DISCONNECT
15:24:12 [000920] 2: Release Active Dlg
15:24:12 [000920] 2: PlayFile(/opt/voc/NotifyVoip.pcm)=0
15:24:12 [000920] 2: ### NewStatus Overlap: PlayBusyTone->PlayDisconnected ###
15:24:12 [000921] 2: ==25:PlayDisconnected
15:24:13 [000935] 2: ==15:PlayBusyTone
15:24:16 [000961] DSP_ch2_check=0