exten => _6XX.,1,Dial(SIP/sip/{EXTEN:1}) отправит на второй сервер двузначный номер. А судя по
-- Called sip:asterisk/105 там у Вас трёхзначные, типа 105.

asterisk- это пароль? А почему он у Вас в строке диал присутствует? Если есть возможность всё вписать в настройки пира?
secret=asterisk

отправляетесь читать мануалы и примеры.

да нет, трехзначный. набираю 6105, первую отрубает- набирает 105. ну он в некоторых примерах в диале присутствует

Connecting the dial plans
Example 1

extensions.conf:
exten => _7XXX,1,Dial(IAX2/myserver:passwordA@IAXserverA/${EXTEN:1},30,r)
exten => _7XXX,2,Dial(SIP/myserver:passwordA@SIPserverA/${EXTEN:1},30,r)
exten => _7XXX,3,Congestion

exten => _8XXX,1,Dial(IAX2/myserver:passwordB@IAXserverB/${EXTEN:1},30,r)
exten => _8XXX,2,Dial(SIP/myserver:passwordB@SIPserverB/${EXTEN:1},30,r)
exten => _8XXX,3,Congestion

взято отсюда
http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers

Те же примеры, но может будет понятней.
http://asterisk.ru/knowledgebase/Asterisk+dual+servers

Пример №2

хорошо, я посмотрю. ded, с вами можно как-нибудь более производительно связаться завтра днем например? мой icq 367-839-045

А зачем?
У меня нет icq.

Небольшой апдейт: звонок проходит в обе стороны, но в динамиках тишина симметричная.

1) Дебаг на уровне tcpdump host 62.141.104.184
2) CLI>sip debug ip 62.141.104.184

самый последний апдейт: сервер_2 успешно принимает звонок по директиве answer (тестовый экстеншн) и играет в него сообщения, но телефон на сервере_1 этого не видит и продолжает в режиме Dial

Debug звонка на сервер_2 с сервера_1


CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:243@62.141.104.184>
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (NAT) to 62.141.104.184:55101:
OPTIONS sip:s@62.141.104.184:55101 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bK22bdb154;rport
From: "asterisk" <sip:asterisk@192.168.1.245>;tag=as37a2c45b
To: <sip:s@62.141.104.184:55101>
Contact: <sip:asterisk@192.168.1.245>
Call-ID: 375bf1523fdf4ada4004d68366448101@192.168.1.245
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 29 Jun 2009 07:12:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
localhost*CLI>
<--- SIP read from 62.141.104.184:55101 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.245:5060;branch=z9hG4bK22bdb154;received=79.141.238.1 73;rport=5060
From: "asterisk" <sip:asterisk@192.168.1.245>;tag=as37a2c45b
To: <sip:s@62.141.104.184:55101>;tag=as4b8231ea
Call-ID: 375bf1523fdf4ada4004d68366448101@192.168.1.245
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:62.141.104.184>
Accept: application/sdp
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '375bf1523fdf4ada4004d68366448101@192.168.1.245' Me thod: OPTIONS
Really destroying SIP dialog '5f3478c0091d79f40e3ea61e5813e922@62.141.104.184' M ethod: OPTIONS
Really destroying SIP dialog '7e9e725a52bf1ba60b88fc8457593ee5@192.168.1.245' Me thod: INVITE
[Jun 29 11:13:07] WARNING[2695]: pbx.c:2538 __ast_pbx_run: Timeout, but no rule 't' in context 'start-out'
-- Executing [h@start-out:1] Hangup("SIP/90-0947d118", "") in new stack
== Spawn extension (start-out, h, 1) exited non-zero on 'SIP/90-0947d118'
[Jun 29 11:13:08] NOTICE[2557]: chan_sip.c:7487 sip_reregister: -- Re-registr ation for spa400@192.168.1.2
[Jun 29 11:13:08] NOTICE[2557]: chan_sip.c:12628 handle_response_register: Outbo und Registration: Expiry for 192.168.1.2 is 120 sec (Scheduling reregistration i n 105 s)
localhost*CLI>
<--- SIP read from 62.141.104.184:55101 --->
REGISTER sip:sip SIP/2.0
Via: SIP/2.0/UDP 62.141.104.184:55101;branch=z9hG4bK17145749;rport
From: <sip:sip@79.141.238.173>;tag=as214b5073
To: <sip:sip@79.141.238.173>
Call-ID: 546944302c35f05b7c2b682825f28282@127.0.0.1
CSeq: 114 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="sip", realm="asterisk", algorithm=MD5, uri="sip: sip", nonce="6f8ab473", response="93fac73ac6edba7efc14a4f076c3a369"
Expires: 120
Contact: <sip:s@62.141.104.184:55101>
Event: registration
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 62.141.104.184 : 55101 (NAT)

<--- Transmitting (NAT) to 62.141.104.184:55101 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 62.141.104.184:55101;branch=z9hG4bK17145749;received=62.141.104 .184;rport=55101
From: <sip:sip@79.141.238.173>;tag=as214b5073
To: <sip:sip@79.141.238.173>
Call-ID: 546944302c35f05b7c2b682825f28282@127.0.0.1
CSeq: 114 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:sip@192.168.1.245>
Content-Length: 0


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

<--- Transmitting (NAT) to 62.141.104.184:55101 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 62.141.104.184:55101;branch=z9hG4bK17145749;received=62.141.104 .184;rport=55101
From: <sip:sip@79.141.238.173>;tag=as214b5073
To: <sip:sip@79.141.238.173>;tag=as3ee7365f
Call-ID: 546944302c35f05b7c2b682825f28282@127.0.0.1
CSeq: 114 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4d739835"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '546944302c35f05b7c2b682825f28282@127.0.0.1 ' in 32000 ms (Method: REGISTER)
localhost*CLI>
<--- SIP read from 62.141.104.184:55101 --->
REGISTER sip:sip SIP/2.0
Via: SIP/2.0/UDP 62.141.104.184:55101;branch=z9hG4bK7ffde703;rport
From: <sip:sip@79.141.238.173>;tag=as09b5378a
To: <sip:sip@79.141.238.173>
Call-ID: 546944302c35f05b7c2b682825f28282@127.0.0.1
CSeq: 115 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="sip", realm="asterisk", algorithm=MD5, uri="sip: sip", nonce="4d739835", response="2af4f36e3d088a6a2032db6d1cfabd7d"
Expires: 120
Contact: <sip:s@62.141.104.184:55101>
Event: registration
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 62.141.104.184 : 55101 (NAT)

<--- Transmitting (NAT) to 62.141.104.184:55101 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 62.141.104.184:55101;branch=z9hG4bK7ffde703;received=62.141.104 .184;rport=55101
From: <sip:sip@79.141.238.173>;tag=as09b5378a
To: <sip:sip@79.141.238.173>
Call-ID: 546944302c35f05b7c2b682825f28282@127.0.0.1
CSeq: 115 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:sip@192.168.1.245>
Content-Length: 0


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

<--- Transmitting (NAT) to 62.141.104.184:55101 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.141.104.184:55101;branch=z9hG4bK7ffde703;received=62.141.104 .184;rport=55101
From: <sip:sip@79.141.238.173>;tag=as09b5378a
To: <sip:sip@79.141.238.173>;tag=as3ee7365f
Call-ID: 546944302c35f05b7c2b682825f28282@127.0.0.1
CSeq: 115 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: <sip:s@62.141.104.184:55101>;expires=120
Date: Mon, 29 Jun 2009 07:13:12 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '546944302c35f05b7c2b682825f28282@127.0.0.1 ' in 32000 ms (Method: REGISTER)
localhost*CLI>
<--- SIP read from 62.141.104.184:55101 --->
OPTIONS sip:79.141.238.173 SIP/2.0
Via: SIP/2.0/UDP 62.141.104.184:55101;branch=z9hG4bK45649993;rport
From: "asterisk" <sip:asterisk@62.141.104.184>;tag=as6423e3f7
To: <sip:79.141.238.173>
Contact: <sip:asterisk@62.141.104.184>
Call-ID: 22a29e76100f48877e7b60aa6b5cc9cb@62.141.104.184
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 24 Jul 2009 07:03:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Looking for s in start-out (domain 79.141.238.173)

<--- Transmitting (no NAT) to 62.141.104.184:55101 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 62.141.104.184:55101;branch=z9hG4bK45649993;received=62.141.104 .184;rport=55101
From: "asterisk" <sip:asterisk@62.141.104.184>;tag=as6423e3f7
To: <sip:79.141.238.173>;tag=as5da0bdf0
Call-ID: 22a29e76100f48877e7b60aa6b5cc9cb@62.141.104.184
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:192.168.1.245>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '22a29e76100f48877e7b60aa6b5cc9cb@62.141.10 4.184' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '546944302c35f05b7c2b682825f28282@127.0.0.1' Method : REGISTER
localhost*CLI> sip set debug off
SIP Debugging Disabled

Это - Debug звонка на сервер_2 с сервера_1 не нужно, это же работало с самого начала?
Крайне неудачные алиасы, настолько путанно всё!

REGISTER sip:sip SIP/2.0 - ещё не хватает пароля, тоже sip.

Да и сам дебаг - это Вы для себя делаете, а не для меня. Старайтесь сами разобраться.