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.