Да нет, стандартные выводы правда должны быть видны при нормальной вербозности, может Вы её задвинули?
Установите
core set verbose 3

При любом звонке будет видно кто кому звонит типа
-- Executing Dial("SIP/1000-41a3ba70", "SIP/toCS1000E/5004") in new stack
Если у вас этот звонок конечно в этом контексте исполняется

'_1XXX' => 1. Dial(SIP/${EXTEN}) [pbx_config]
'_5XXX' => 1. Dial(SIP/toCS1000E/${EXTEN})

Стоял 0 уровень, включил как было сказано, вот результат:

-- Executing [5004@default:1] Dial("SIP/1000-00000006", "SIP/toCS1000E/5004") in new stack
-- Called toCS1000E/5004
-- Got SIP response 302 "Moved Temporarily" back from 10.21.2.203
-- Now forwarding SIP/1000-00000006 to 'SIP/5004' (thanks to SIP/toCS1000E-00000007)
[May 13 13:00:50] WARNING[1028]: chan_sip.c:3094 create_addr: No such host: 5004
[May 13 13:00:50] NOTICE[1028]: app_dial.c:535 wait_for_answer: Unable to create local channel for call forward to 'SIP/5004' (cause = 20)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/1000-00000006' status is 'CHANUNAVAIL'
a

да не в макросах дело... ну попробуйте уже наконец без ';phone-context...'. Астериск в данном коннкретном случае игнорирует всё, что идет после ';' в contact. Бага это или фича вот в чем вопрос...

Пока, я не могу его убрать, на базе этого контекста построена маршрутизация в сети в которую интегрируется Астериск...
Попробую вечером этим заняться и отпишу о результате.

Это 10.21.2.203 отвечает 302, если бы это был просто ИП телефон, то это была бы переадресация, включенная на нём AFAIK, типа СallForwardAll

В данном случае 10.21.2.203 это redirect сервер.

Попытался отключить phone-context
INVITE теперь выглядит так: INVITE sip:+5004@nrs.voip.telecom.lan SIP/2.0
+ убрать не могу...

Ниже трассировка...
приходит SIP/2.0 302 Moved Temporarily с Contact: <sip:+5004@nrs.voip.telecom.lan:5060;maddr=10.21.2.200;transport=udp;x-nt-redirect=redirect-server>

и как бы он должен, повторный INVITE слать на maddr=10.21.2.200, а судя по трассировке, он шлет опять на
nrs.voip.telecom.lan.



<--- SIP read from 10.21.0.50:5060 --->
INVITE sip:5004@10.21.0.114 SIP/2.0
Via: SIP/2.0/UDP 10.21.0.50;branch=z9hG4bK0501d7af000000794bed3e6a000036eb00000050;rport
From: "Andrey" <sip:1000@10.21.0.114>;tag=28b2d1e1eb
To: <sip:5004@10.21.0.114>
Contact: <sip:1000@10.21.0.50>
Call-ID: 91A617B00FA148B4A12C5AB8490516B80x0501d7af
CSeq: 2 INVITE
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 241
Content-Type: application/sdp
Supported: replaces,norefersub,timer
Proxy-Authorization: Digest username="1000",realm="telecom.lan",nonce="4e318ba9",uri="sip:5004@10.21.0.114",response="2df509385bda7bf580c8a6e3e4a9182f",algorithm=MD5

v=0
o=- 3482828010 3482828011 IN IP4 10.21.0.50
s=SJphone
c=IN IP4 10.21.0.50
t=0 0
m=audio 49172 RTP/AVP 8 101
c=IN IP4 10.21.0.50
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=setup:active
a=sendrecv

<--- Transmitting (NAT) to 10.21.0.50:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.21.0.50;branch=z9hG4bK0501d7af000000794bed3e6a000036eb00000050;received=10.21.0.50;rport=5060
From: "Andrey" <sip:1000@10.21.0.114>;tag=28b2d1e1eb
To: <sip:5004@10.21.0.114>
Call-ID: 91A617B00FA148B4A12C5AB8490516B80x0501d7af
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:5004@10.21.0.114>
Content-Length: 0

---
Audio is at 10.21.0.114 port 16812
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.21.2.203:5060:
INVITE sip:+5004@nrs.voip.telecom.lan SIP/2.0
Via: SIP/2.0/UDP 10.21.0.114:5060;branch=z9hG4bK161b5a20;rport
From: "Andrey-SIP" <sip:1000@nrs.voip.telecom.lan>;tag=as6802e106
To: <sip:+5004@nrs.voip.telecom.lan>
Contact: <sip:1000@10.21.0.114>
Call-ID: 72f77a975876f66c0ca1a25918eb8135@nrs.voip.telecom.lan
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="asterisk", realm="nrs.voip.telecom.lan", algorithm=MD5, uri="sip:+5004@nrs.voip.telecom.lan", nonce="5b20e5ab7d41a9034652d87e9f19e475", response="170e2b9831e58fdea17947ceb7d5f56f", opaque="edd0d8060aed140e33f6e60ec5277b30", qop=auth, cnonce="644bbf6a", nc=00000001
Date: Fri, 14 May 2010 16:13:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 234

v=0
o=root 714 715 IN IP4 10.21.0.114
s=session
c=IN IP4 10.21.0.114
t=0 0
m=audio 16812 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
asterisk*CLI>
<--- SIP read from 10.21.2.203:5060 --->
SIP/2.0 302 Moved Temporarily
Via: SIP/2.0/UDP 10.21.0.114:5060;branch=z9hG4bK161b5a20;rport;received=10.21.0.114
From: "Andrey-SIP" <sip:1000@nrs.voip.telecom.lan>;tag=as6802e106
To: <sip:+5004@nrs.voip.telecom.lan>;tag=63485
Call-ID: 72f77a975876f66c0ca1a25918eb8135@nrs.voip.telecom.lan
CSeq: 103 INVITE
Contact: <sip:+5004@nrs.voip.telecom.lan:5060;maddr=10.21.2.200;transport=udp;x-nt-redirect=redirect-server>
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 10.21.2.203:5060:
ACK sip:+5004@nrs.voip.telecom.lan SIP/2.0
Via: SIP/2.0/UDP 10.21.0.114:5060;branch=z9hG4bK161b5a20;rport
From: "Andrey-SIP" <sip:1000@nrs.voip.telecom.lan>;tag=as6802e106
To: <sip:+5004@nrs.voip.telecom.lan>;tag=63485
Contact: <sip:1000@10.21.0.114>
Call-ID: 72f77a975876f66c0ca1a25918eb8135@nrs.voip.telecom.lan
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 10.21.0.114 port 12510
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.21.2.203:5060:
INVITE sip:+5004@nrs.voip.telecom.lan:5060 SIP/2.0
Via: SIP/2.0/UDP 10.21.0.114:5060;branch=z9hG4bK417f555e;rport
From: "Andrey-SIP" <sip:1000@10.21.0.114>;tag=as3cae0cc4
To: <sip:+5004@nrs.voip.telecom.lan:5060>
Contact: <sip:1000@10.21.0.114>
Call-ID: 2e072db944ec7ed06da12062022366bc@10.21.0.114
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 14 May 2010 16:13:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 714 714 IN IP4 10.21.0.114
s=session
c=IN IP4 10.21.0.114
t=0 0
m=audio 12510 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Really destroying SIP dialog '72f77a975876f66c0ca1a25918eb8135@nrs.voip.telecom.lan' Method: INVITE
asterisk*CLI>
<--- SIP read from 10.21.2.203:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.21.0.114:5060;branch=z9hG4bK417f555e;rport;received=10.21.0.114
From: "Andrey-SIP" <sip:1000@10.21.0.114>;tag=as3cae0cc4
To: <sip:+5004@nrs.voip.telecom.lan:5060>;tag=16032
Call-ID: 2e072db944ec7ed06da12062022366bc@10.21.0.114
CSeq: 102 INVITE
Contact: <sip:1000@10.21.0.114>
Proxy-Authenticate: Digest realm="10.21.0.114", nonce="35870965027d4ff2865428e8d899b83b", opaque="edd0d8060aed140e33f6e60ec5277b30", stale=false, algorithm=MD5, qop="auth"
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 10.21.2.203:5060:
ACK sip:+5004@nrs.voip.telecom.lan:5060 SIP/2.0
Via: SIP/2.0/UDP 10.21.0.114:5060;branch=z9hG4bK417f555e;rport
From: "Andrey-SIP" <sip:1000@10.21.0.114>;tag=as3cae0cc4
To: <sip:+5004@nrs.voip.telecom.lan:5060>;tag=16032
Contact: <sip:1000@10.21.0.114>
Call-ID: 2e072db944ec7ed06da12062022366bc@10.21.0.114
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
[May 14 16:13:26] NOTICE[720]: chan_sip.c:13043 handle_response_invite: Failed to authenticate on INVITE to '"Andrey-SIP" <sip:1000@10.21.0.114>;tag=as3cae0cc4'
asterisk*CLI>
<--- Reliably Transmitting (NAT) to 10.21.0.50:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.21.0.50;branch=z9hG4bK0501d7af000000794bed3e6a000036eb00000050;received=10.21.0.50;rport=5060
From: "Andrey" <sip:1000@10.21.0.114>;tag=28b2d1e1eb
To: <sip:5004@10.21.0.114>;tag=as6e86b1bd
Call-ID: 91A617B00FA148B4A12C5AB8490516B80x0501d7af
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21


<------------>
asterisk*CLI>
<--- SIP read from 10.21.0.50:5060 --->
ACK sip:5004@10.21.0.114 SIP/2.0
Via: SIP/2.0/UDP 10.21.0.50;branch=z9hG4bK0501d7af000000794bed3e6a000036eb00000050;rport
From: "Andrey" <sip:1000@10.21.0.114>;tag=28b2d1e1eb
To: <sip:5004@10.21.0.114>;tag=as6e86b1bd
Call-ID: 91A617B00FA148B4A12C5AB8490516B80x0501d7af
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: SJphone/1.65.377a (SJ Labs)
Content-Length: 0
Proxy-Authorization: Digest username="1000",realm="telecom.lan",nonce="4e318ba9",uri="sip:5004@10.21.0.114",response="2df509385bda7bf580c8a6e3e4a9182f",algorithm=MD5


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '2e072db944ec7ed06da12062022366bc@10.21.0.114' Method: INVITE
Really destroying SIP dialog '91A617B00FA148B4A12C5AB8490516B80x0501d7af' Method: ACK
asterisk*CLI>
<--- SIP read from 10.21.0.50:5060 --->


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

Я так понял, Asterisk игнорирует поле maddr в Contact (Contact: <sip:5004;phone-context=cdp.udp@nrs.voip.telecom.lan:5060;maddr=10.21.2.200;transport=udp)
При повторном INVITE он использует опять адрес nrs.voip.telecom.lan, а не пришедший в maddr=10.21.2.200.

Проверил на версии 1.6.2, тот же результат.