Не ходит с некоторых пиров звонки между 2-мя *

Всем привет. Есть два астериска. Между ними поднят sip транк. С некоторых пиров первого сервера не доходят звонки до второго. Настройки пиров идентичные. В логах при неосуществленном звонке видно вот что:

<--- SIP read from 10.0.0.1:5063 --->
INVITE sip:990001@sip.insit.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5063;branch=z9hG4bK1697633786
From: "10185" <sip:10185@sip.insit.ru>;tag=538964269
To: <sip:990001@sip.myserver.ru>
Call-ID: 554399431@10.0.0.1
CSeq: 1 INVITE
Contact: <sip:10185@10.0.0.1:5063>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, `SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE`
Max-Forwards: 70
User-Agent: Yealink SIP-T20P 9.60.14.13
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 290

v=0
o=- 20025 20025 IN IP4 10.0.0.1
s=SDP data
c=IN IP4 10.0.0.1
t=0 0
m=audio 11800 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (14 headers 14 lines) ---
Sending to 10.0.0.1 : 5063 (no NAT)
Using INVITE request as basis request - 554399431@10.0.0.1

<--- Reliably Transmitting (no NAT) to 10.0.0.1:5063 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.1:5063;branch=z9hG4bK1697633786;received=10.0.0.1
From: "10185" <sip:10185@sip.myserver.ru>;tag=538964269
To: <sip:990001@sip.myserver.ru>;tag=as389d3efb
Call-ID: 554399431@10.0.0.1
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4faddace"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '554399431@10.0.0.1' in 32000 ms (Method: INVITE)
Found user '10185'

<--- SIP read from 10.0.0.1:5063 --->
ACK sip:990001@sip.myserver.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5063;branch=z9hG4bK1697633786
From: "10185" <sip:10185@sip.myserver.ru>;tag=538964269
To: <sip:990001@sip.myserver.ru>;tag=as389d3efb
Call-ID: 554399431@10.0.0.1
CSeq: 1 ACK
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from 10.0.0.1:5063 --->
INVITE sip:990001@sip.myserver.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5063;branch=z9hG4bK996552644
From: "10185" <sip:10185@sip.myserver.ru>;tag=538964269
To: <sip:990001@sip.myserver.ru>
Call-ID: 554399431@10.0.0.1
CSeq: 2 INVITE
Contact: <sip:10185@10.0.0.1:5063>
Proxy-Authorization: Digest username="10185", realm="asterisk", nonce="4faddace", uri="sip:990001@sip.myserver.ru", response="7ea40e12ff68927ea48b5587791ec6b5", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T20P 9.60.14.13
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 290

v=0
o=- 20025 20025 IN IP4 10.0.0.1
s=SDP data
c=IN IP4 10.0.0.1
t=0 0
m=audio 11800 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (15 headers 14 lines) ---
Sending to 10.0.0.1 : 5063 (no NAT)
Using INVITE request as basis request - 554399431@10.0.0.1
Found user '10185'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x110c (ulaw|alaw|g729|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event), combined - 0x0 (nothing)
Peer audio RTP is at port 10.0.0.1:11800
Looking for 990001 in insit (domain sip.myserver.ru)
list_route: hop: <sip:10185@10.0.0.1:5063>

<--- Transmitting (no NAT) to 10.0.0.1:5063 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.1:5063;branch=z9hG4bK996552644;received=10.0.0.1
From: "10185" <sip:10185@sip.myserver.ru>;tag=538964269
To: <sip:990001@sip.myserver.ru>
Call-ID: 554399431@10.0.0.1
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:990001@217.64.140.138>
Content-Length: 0


<------------>
    -- Executing [990001@insit:1] NoOp("SIP/10185-0000fcbc", "Звонок на Факс 10185") in new stack
    -- Executing [990001@insit:2] Dial("SIP/10185-0000fcbc", "SIP/990001@SIPPhone|6000|tT") in new stack
    -- Called 990001@SIPPhone
    -- SIP/SIPPhone-0000fcbd is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/10185-0000fcbc' status is 'CONGESTION'

<--- Reliably Transmitting (no NAT) to 10.0.0.1:5063 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.0.0.1:5063;branch=z9hG4bK996552644;received=10.0.0.1
From: "10185" <sip:10185@sip.myserver.ru>;tag=538964269
To: <sip:990001@sip.myserver.ru>;tag=as0e5cc2fd
Call-ID: 554399431@10.0.0.1
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21
Content-Length: 0


<------------>

<--- SIP read from 10.0.0.1:5063 --->
ACK sip:990001@sip.myserver.ru SIP/2.0
Via: SIP/2.0/UDP 10.0.0.1:5063;branch=z9hG4bK996552644
From: "10185" <sip:10185@sip.myserver.ru>;tag=538964269
To: <sip:990001@sip.myserver.ru>;tag=as0e5cc2fd
Call-ID: 554399431@10.185.1.9
CSeq: 2 ACK
Content-Length: 0

На втором астериске не видно этого звонка. У кого-нить есть идеи по этому поводу.