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Добрый день!
Существует ли возможность на аддпаке сделать следующую вещь:
допустим есть два разных потсовских порта, нужно чтобы один порт использовал для исходящих вызовов стандартные воипные диал пиры. А второй порт использовал уже другие диал пиры?

Как это можно организовать?

http://www.addpac.su/translation.htm

Пример 13

Сделал так как в примере:

! FXS
voice-port 0/0
translate-incoming called-number 2
fax-early-detect

dial-peer voice 0 pots
destination-pattern 7ХХХХХХХХХХ
port 0/0

dial-peer voice 9 voip
destination-pattern 4546T
session target Y.Y.Y.Y 5060
session protocol sip
voice-class codec 0
dtmf-relay h245-alphanumeric
no vad
translate-outgoing called-number 3
fax protocol t38 redundancy 2
fax rate 9600

translation-rule 2
rule 1 T 4546T
!
translation-rule 3
rule 1 4546T T

вот debug rta ipc:
[2492.950] VM(0/0/0) vmOffHook
[2493.010] VM(0/0/0) vmTmoOffHook
[2493.010] VM(0/0/0) Rx OffHook
[2493.010] VM(0/0/0) vopp enable
[2493.010] VM(0/0/0) Tx OFFHOOK_IND
[2493.010] VM(0/0/0) play Dial tone
[2493.010] VM(0/0/0) set T38 mode STD
[2493.010] VM(0/0/0) Fax rate 9600
[2493.065] RTA(0/0/0) Rx RS_LISTEN_REQ callId=36 ssId=1 G711U
peer=0.0.0.0 mp=23066/23067 hp=0/0
[2493.065] VM(0/0/0) codec same G711U
[2493.125] RTA(0/0/0) Rx RS_CLOSE_REQ callId=36 ssId=1 dir=reve
[2493.125] RTA(0/0/0) close Media socket
[2493.125] RTA(0/0/0) close RTCP socket
[2493.125] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[2493.125] VM(0/0/0) play mute
[2493.125] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[2493.125] VM(0/0/0) play Reorder tone
[2495.290] VM(0/0/0) vmOnHook
[2495.340] VM(0/0/0) vmTmoOnHook
[2495.390] VM(0/0/0) vmTmoOnHook
[2495.440] VM(0/0/0) vmTmoOnHook
[2495.490] VM(0/0/0) vmTmoOnHook
[2495.540] VM(0/0/0) vmTmoOnHook
[2495.590] VM(0/0/0) vmTmoOnHook
[2495.640] VM(0/0/0) vmTmoOnHook
[2495.690] VM(0/0/0) vmTmoOnHook
[2495.740] VM(0/0/0) vmTmoOnHook
[2495.790] VM(0/0/0) vmTmoOnHook
[2495.840] VM(0/0/0) vmTmoOnHook
[2495.890] VM(0/0/0) vmTmoOnHook
[2495.940] VM(0/0/0) vmTmoOnHook
[2495.990] VM(0/0/0) vmTmoOnHook
[2495.990] VM(0/0/0) Rx OnHook
[2495.990] VM(0/0/0) vopp idle
[2495.990] VM(0/0/0) Tx DISCONN_CNF


на софтсвич прилетают пустые инвайты, он дает отбой сразу. и в результате клиент сразу слышит короткие гудки.

В чем может быть затык?

Покажите вместе
deb voip call
deb voip sip
deb rta ipc

Вместе:
deb voip call
deb voip sip
deb rta ipc


(config)# [83475.430] VM(0/0/0) vmOffHook
[83475.490] VM(0/0/0) vmTmoOffHook
[83475.490] VM(0/0/0) Rx OffHook
[83475.490] VM(0/0/0) vopp enable
[83475.490] VM(0/0/0) Tx OFFHOOK_IND
[83475.490] VM(0/0/0) play Dial tone
1 <CEP 000000> : Call Received
2 <CEP 000000> : Call Initiated : calledNumber() crv(0) total(1)
3 <Call 214> : ****************** Call Created status(InitiatedByFXS) *******************
4 <CEP 000000> : Calling number(Номер А)
5 <CEP 000000> : Call id(ed26724b-c0c6-c97f-8146-0002a402230e) callNum(214)
6 <Call 214> : Match check (MatchedAll)
7 <Call 214> : MatchAllProcess After Sorted
<0> id(9) dest(4546T) prefer(0) selected(12)
8 <Call 214> : Initiate callee with dial-peer(4546T) status(CalleeDeterminedAll) id(ed26724b-c0c6-c97f-8146-0002a402230e)
9 <NetEP 214> : InitiateOutCall: calledNum(4546) callingNum(Номер А) target(Y.Y.Y.Y:5060)
10 <NetEP 214> : DoCall: calledAddr(sip:Y.Y.Y.Y:5060) callingAddr(Номер А)
[83475.495] VM(0/0/0) set T38 mode STD
[83475.495] VM(0/0/0) Fax rate 9600
11 <SIP 0> : No authentication information available
12 <SIP 214> : Send INVITE Request

Sending SIP PDU to ( Y.Y.Y.Y:5060 ) from 5060
INVITE sip:Y.Y.Y.Y SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKed4be347a413
From: <sip:Номер А@x.x.x.x>;tag=ed4be347a4
To: <sip:Y.Y.Y.Y>
Call-ID: ed26724b-19da-e35f-8147-0002a402230e@x.x.x.x
CSeq: 13 INVITE
Supported: timer, replaces, early-session
Min-SE: 1800
Date: Wed, 10 Feb 2010 03:24:29 GMT
User-Agent: AddPac SIP Gateway
Contact: <sip:Номер А@x.x.x.x>
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO
Content-Type: application/sdp
Content-Length: 236
Max-Forwards: 70

v=0
o=73432165180 1265772269 1265772269 IN IP4 x.x.x.x
s=AddPac Gateway SDP
c=IN IP4 x.x.x.x
t=1265772269 0
m=audio 23392 RTP/AVP 18 8 0
a=rtpmap:18 G729/8000/3
a=rtpmap:8 PCMA/8000/3
a=rtpmap:0 PCMU/8000/3
a=ptime:20

[83475.550] RTA(0/0/0) Rx RS_LISTEN_REQ callId=214 ssId=1 G711U
peer=0.0.0.0 mp=23392/23393 hp=0/0
[83475.550] VM(0/0/0) codec same G711U

Received SIP PDU from ( Y.Y.Y.Y:5060 )
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKed4be347a413;received=x.x.x.x
From: <sip:Номер А@x.x.x.x>;tag=ed4be347a4
To: <sip:Y.Y.Y.Y>;tag=as74d28835
Call-ID: ed26724b-19da-e35f-8147-0002a402230e@x.x.x.x
CSeq: 13 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

13 <SIP 214> : Receive 404 Not Found
14 <SIP 214> : Transaction (13 INVITE) completed
15 <SIP 214> : Send ACK Request

Sending SIP PDU to ( Y.Y.Y.Y:5060 ) from 5060
ACK sip:Y.Y.Y.Y SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKed4be347a413
From: <sip:Номер А@x.x.x.x>;tag=ed4be347a4
To: <sip:Y.Y.Y.Y>;tag=as74d28835
Call-ID: ed26724b-19da-e35f-8147-0002a402230e@x.x.x.x
CSeq: 13 ACK
Content-Length: 0
Max-Forwards: 70


16 <SIP 214> : Check Event Relation
17 <SIP 214> : ReleaseWithNothing
[83475.615] RTA(0/0/0) Rx RS_CLOSE_REQ callId=214 ssId=1 dir=reve
[83475.615] RTA(0/0/0) close Media socket
[83475.615] RTA(0/0/0) close RTCP socket
18 <Call 214> : Terminated from(fffffffe) this(Remote:InvalidNumber) before(NULL) forced(0)
19 <CEP 000000> : DisconnectCall at Busy
20 <CEP 000000> : StopSignal
[83475.615] RTA(0/0/0) Rx AP_SVC_REQ nSvcElem=1 rawDataLen=0
DTMF_STOP
[83475.615] VM(0/0/0) play mute
21 <CEP 000000> : Disconnect (0)
[83475.615] RTA(0/0/0) Rx CC_DISCONN_REQ CZ=0, peerId(0/0/0)
[83475.615] VM(0/0/0) play Reorder tone
22 <NetEP 214> : Call TO <sip:Y.Y.Y.Y> terminated reason(Remote:InvalidNumber)
[83477.630] VM(0/0/0) vmOnHook
[83477.680] VM(0/0/0) vmTmoOnHook
[83477.730] VM(0/0/0) vmTmoOnHook
[83477.780] VM(0/0/0) vmTmoOnHook
[83477.830] VM(0/0/0) vmTmoOnHook
[83477.880] VM(0/0/0) vmTmoOnHook
[83477.930] VM(0/0/0) vmTmoOnHook
[83477.980] VM(0/0/0) vmTmoOnHook
[83478.030] VM(0/0/0) vmTmoOnHook
[83478.080] VM(0/0/0) vmTmoOnHook
[83478.130] VM(0/0/0) vmTmoOnHook
[83478.180] VM(0/0/0) vmTmoOnHook
[83478.230] VM(0/0/0) vmTmoOnHook
[83478.280] VM(0/0/0) vmTmoOnHook
[83478.330] VM(0/0/0) vmTmoOnHook
[83478.330] VM(0/0/0) Rx OnHook
[83478.330] VM(0/0/0) vopp idle
[83478.330] VM(0/0/0) Tx DISCONN_CNF
23 <CEP 000000> : Disconnected(16) at Disconnecting

где:
Номер А - номер клиента
Y.Y.Y.Y - ip адрес софсвича
x.x.x.x - ip адрес аддпака

Номер какой набираете?!

Вот в этом и дело, что при поднятии трубки клиент слышит зуммер занято.
По различным трейсам можно увидеть, что аддпак посылает инвайт с пустым значением Номера Б(Поле To). Софтсвич получая пустой инвайт отвечает кодом разьединения 404.
Аддпак не дает набрать номер, почему то не ждет после откусывания перфикса дальнейшего набора

Как так не даёт?!
Клиент должен поднять трубку и набирать 4546XXXXX

Покажите весь конфиг.

Не дает, это очень похоже на Hot Line. При подянтии трубки формируется инвайт с пустым полем "To".

sh ver
AP1100-FXS8 System software Revision 8.234

sh run
no bridge spanning-tree
!
proxy-arp
!
dhcp-list 1 type server
dhcp-list 1 address server 10.1.1.2 10.1.1.126 255.255.255.128
!
!
no ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 10.227.8.22 255.255.255.0
qos-control
!
interface ether1.0
no ip address
!
snmp name AP1100-FXS8
!
no arp reset
!
route 0.0.0.0 0.0.0.0 10.227.8.1
!
ip-tos rtp precedence 5
ip-tos rtp delay
ip-tos rtp throughput
ip-tos sig precedence 3
ip-tos sig throughput
!
! VoIP configuration.
!
! Voice service voip configuration.
!
voice service voip
fax protocol t38 redundancy 2
fax rate 9600
h323 call start fast
h323 call tunnel enable
timeout tidt 10
timeout tttl 10
busyout monitor gatekeeper
busyout monitor voip-interface
!
!
! Voice port configuration.
!
! FXS
voice-port 0/0
fax-early-detect
translate-incoming called-number 2
!
!
! FXS
voice-port 0/1
fax-early-detect
!
!
! FXS
voice-port 0/2
fax-early-detect
!
!
! FXS
voice-port 0/3
fax-early-detect
!
!
! FXS
voice-port 1/0
fax-early-detect
!
!
! FXS
voice-port 1/1
fax-early-detect
!
!
! FXS
voice-port 1/2
fax-early-detect
!
!
! FXS
voice-port 1/3
fax-early-detect
!
!
!
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 7343xxxxxxx
port 0/0
!
dial-peer voice 1 pots
destination-pattern 7343zzzzzzzz
port 0/1
!
dial-peer voice 2 pots
destination-pattern 7343cccccccc
port 0/2
!
dial-peer voice 3 pots
destination-pattern 7343vvvvvvv
port 0/3
!
dial-peer voice 4 pots
destination-pattern 7343bbbbbbb
port 1/0
!
dial-peer voice 5 pots
destination-pattern 7343nnnnnnn
port 1/1
!
dial-peer voice 6 pots
destination-pattern 7343mmmmmmm
port 1/2
!
dial-peer voice 7 pots
destination-pattern 7343hhhhhhh
port 1/3
!
!
!
! Voip peer configuration.
!
dial-peer voice 9 voip
destination-pattern 4546T
session target Y.Y.Y.Y 5060
session protocol sip
voice-class codec 0
dtmf-relay h245-alphanumeric
no vad
translate-outgoing called-number 3
fax protocol t38 redundancy 2
fax rate 9600
!
dial-peer voice 10 voip
destination-pattern [01]..F
session target T.T.T.T
voice-class codec 0
dtmf-relay h245-alphanumeric
no vad
translate-outgoing called-number 1
fax protocol t38 redundancy 2
fax rate 9600
!
dial-peer voice 11 voip
destination-pattern [2-3]......F
session target T.T.T.T
voice-class codec 0
dtmf-relay h245-alphanumeric
no vad
translate-outgoing called-number 1
preference 1
fax protocol t38 redundancy 2
fax rate 9600
!
dial-peer voice 12 voip
destination-pattern 8[2-9].........F
session target T.T.T.T
voice-class codec 0
dtmf-relay h245-alphanumeric
no vad
translate-outgoing called-number 1
preference 2
fax protocol t38 redundancy 2
fax rate 9600
!
dial-peer voice 13 voip
destination-pattern 810T
session target T.T.T.T
voice-class codec 0
dtmf-relay h245-alphanumeric
no vad
translate-outgoing called-number 1
preference 2
fax protocol t38 redundancy 2
fax rate 9600
!
!
!
!
!
!
gatekeeper
!
!
! Gateway configuration.
!
gateway
!
!
! Codec classes configuration.
!
voice class codec 0
codec preference 1 g729
codec preference 2 g711alaw
codec preference 3 g711ulaw
!
!
!
! Translation Rule configuration.
!
translation-rule 1
rule 1 [01]T %99
rule 2 [23]T 7343%99
rule 3 810T %04%99
rule 4 8T 7%02%99
!
translation-rule 2
rule 1 T 4546T
!
translation-rule 3
rule 1 4546T .T

Не понятно почему шлюз сразу соединяется... Пошейтесь до последней версии
http://www.addpac.su/arhiv.htm