Создать внутренний номер на астериске . С данными этого внутреннего номерами зарегестрировать шлюз на астериске. Куда уж понятней? или за Вас это надо сделать?

Мне бы конфиги показать ...

конфиги чего? Вы же через FreePBX настраиваете , если уж надо то /etc/asterisk/sip.conf .
Конфиг шлюза , если только скриншоты, и то не могу предоставить. Но уж шлюз настроить уж точно проблем не должно быть, мне как помниться там настраивается всё довольно просто. Как говориться ответы в мануалах.

=)
и в каком же мануале искать ошибку AcSIPParser: Unrecognized Header was detected at line: 10

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
Audio is at 10.0.0.251 port 12844
Video is at 10.0.0.251 port 14472
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding video codec 0x80000 (h263) to SDP
Adding video codec 0x200000 (h264) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.0.1.251:5060:
INVITE sip:101@10.0.1.251 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.251:5060;branch=z9hG4bK4f9afc80;rport
Max-Forwards: 70
From: "IPhone" <sip:trixbox@10.0.0.251>;tag=as5cdfd98f
To: <sip:101@10.0.1.251>
Contact: <sip:trixbox@10.0.0.251>
Call-ID: 0f6619456e3036567c5974210131611d@10.0.0.251
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Date: Fri, 16 Apr 2010 10:16:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 390

v=0
o=root 383523131 383523131 IN IP4 10.0.0.251
s=Asterisk PBX 1.6.0.22-samy-r60
c=IN IP4 10.0.0.251
b=CT:384
t=0 0
m=audio 12844 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/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
m=video 14472 RTP/AVP 34 99
a=rtpmap:34 H263/90000
a=rtpmap:99 H264/90000
a=sendrecv

---
-- Called MP-118/101
trixbox*CLI>
<--- SIP read from UDP://10.0.1.251:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.0.0.251:5060;branch=z9hG4bK4f9afc80;rport
From: "IPhone" <sip:trixbox@10.0.0.251>;tag=as5cdfd98f
To: <sip:101@10.0.1.251>;tag=1c2103695731
Call-ID: 0f6619456e3036567c5974210131611d@10.0.0.251
CSeq: 102 INVITE
Contact: <sip:10.0.1.251>
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-MP-118 FXS/v.4.80A.014.006
Reason: Q.850 ;cause=3
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Transmitting (NAT) to 10.0.1.251:5060:
ACK sip:101@10.0.1.251 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.251:5060;branch=z9hG4bK4f9afc80;rport
Max-Forwards: 70
From: "IPhone" <sip:trixbox@10.0.0.251>;tag=as5cdfd98f
To: <sip:101@10.0.1.251>;tag=1c2103695731
Contact: <sip:trixbox@10.0.0.251>
Call-ID: 0f6619456e3036567c5974210131611d@10.0.0.251
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Content-Length: 0


---
-- SIP/MP-118-0000005e is circuit-busy
...................
Reliably Transmitting (NAT) to 10.0.1.251:5060:
OPTIONS sip:10.0.1.251 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.251:5060;branch=z9hG4bK27e5a2b9;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.0.0.251>;tag=as1bd53fb4
To: <sip:10.0.1.251>
Contact: <sip:Unknown@10.0.0.251>
Call-ID: 174136ac2e0cc3400a1accf16087870a@10.0.0.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Date: Fri, 16 Apr 2010 10:16:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
trixbox*CLI>
<--- SIP read from UDP://10.0.1.251:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.251:5060;branch=z9hG4bK27e5a2b9;rport
From: "Unknown" <sip:Unknown@10.0.0.251>;tag=as1bd53fb4
To: <sip:10.0.1.251>;tag=1c5332752
Call-ID: 174136ac2e0cc3400a1accf16087870a@10.0.0.251
CSeq: 102 OPTIONS
Contact: <sip:10.0.1.251>
Supported: em,100rel,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 234
X-Resources: telchs=8/8;mediachs=0/0

v=0
o=AudiocodesGW 5335185 5335068 IN IP4 10.0.1.251
s=Phone-Call
c=IN IP4 10.0.1.251
t=0 0
m=audio 6000 RTP/AVP 18 4
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=fmtp:4 annexa=no
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 12 lines) ---
Really destroying SIP dialog '174136ac2e0cc3400a1accf16087870a@10.0.0.251' Method: OPTIONS
Reliably Transmitting (NAT) to 10.0.1.251:5060:
OPTIONS sip:10.0.1.251 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.251:5060;branch=z9hG4bK4c50d00a;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@10.0.0.251>;tag=as05234aac
To: <sip:10.0.1.251>
Contact: <sip:Unknown@10.0.0.251>
Call-ID: 10b4433317ae13173c3c3e0d7d90dbc6@10.0.0.251
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.22-samy-r60
Date: Fri, 16 Apr 2010 10:17:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
trixbox*CLI>
<--- SIP read from UDP://10.0.1.251:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.251:5060;branch=z9hG4bK4c50d00a;rport
From: "Unknown" <sip:Unknown@10.0.0.251>;tag=as05234aac
To: <sip:10.0.1.251>;tag=1c155397948
Call-ID: 10b4433317ae13173c3c3e0d7d90dbc6@10.0.0.251
CSeq: 102 OPTIONS
Contact: <sip:10.0.1.251>
Supported: em,100rel,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Content-Type: application/sdp
Content-Length: 238
X-Resources: telchs=8/8;mediachs=0/0

v=0
o=AudiocodesGW 155400377 155400255 IN IP4 10.0.1.251
s=Phone-Call
c=IN IP4 10.0.1.251
t=0 0
m=audio 6000 RTP/AVP 18 4
a=rtpmap:18 g729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 g723/8000
a=fmtp:4 annexa=no
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 12 lines) ---
Really destroying SIP dialog '10b4433317ae13173c3c3e0d7d90dbc6@10.0.0.251' Method: OPTIONS

Что еще не понятно ... в настройках шлюза, конфиги [SIP Params]:
TRUNKGROUP = 1-1,Tehno1,0
TRUNKGROUP = 2-2,Tehno2,0
TRUNKGROUP = 3-3,Tehno3,0
TRUNKGROUP = 4-4,Tehno4,0
TRUNKGROUP = 5-5,Tehno5,0
TRUNKGROUP = 6-6,,0
TRUNKGROUP = 7-7,,0
TRUNKGROUP = 8-8,,0
PROXYIP = 86.110.4.148
AUTHENTICATION_0 = Tehno1,ghb8
AUTHENTICATION_1 = Tehno2,ghb8
AUTHENTICATION_2 = Tehno3,ghb8
AUTHENTICATION_3 = Tehno4,ghb8
AUTHENTICATION_4 = Tehno5,ghb8

со шлюза ошибки:
1d:17h:44m:3s ( lgr_TrnkGrp)(1680 ) !! [ERROR] #0:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number 101 [File: Line:-1]

1d:17h:44m:3s ( lgr_psbrdif)(1681 ) !! [ERROR] AcBoard::GetEndPoint- Can't find EndPoint for Dest:101 Source:trixbox SourceIp:a0000fb [File: Line:-1]

1d:17h:44m:3s ( lgr_call)(1682 ) !! [ERROR] Call::GetEndPoint- Can't find endpoint for phone number 101 [File: Line:-1]

давайте еще логов почтового сервера , апача и думаю все логи logger'а сюда же.
У вас криво настроен сип аккаунт , обратитесь к документации по шлюзу.

Гугл выдает множество результатов по данной теме, даже с картинками и даже для боксовых астерисков..