Столкнулся с такой ситуацией, при ответе агента не происходит соединение.
Asterisk 1.4.29

queues.conf
[general]
persistentmembers=yes
autofill=yes
monitor-type=MixMonitor

[myQueue]
strategy=rrmemory
timeout=15
wrapuptime=15
autofill=no
autopause=no
joinempty=yes
leavewhenempty=no
reportholdtime=no
maxlen=0
musicclass=default
member=Agent/1001

agents.conf
[general]
persistentagents=yes
[agents]
agent=1001,1234,user

extension.conf
exten=7001,1,agentcallbacklogin()

exten=57,1,Answer()
exten=57,n,Ringing
exten=57,n,Wait(2)
exten=57,n,Queue(oren)
exten=57,n,Hangup

Вот что говорит sip set debug
<--- SIP read from 192.168.3.117:21424 --->



<------------->
Audio is at 192.168.3.254 port 18768
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x200 (speex) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.3.117:21424:
INVITE sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb SIP/2.0
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK2e48aacf;rport
From: "403" <sip:403@192.168.3.254>;tag=as472e90c6
To: <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb>
Contact: <sip:403@192.168.3.254>
Call-ID: 701e364769da003321ffd67f3f3c4151@192.168.3.254
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 29 Apr 2010 10:58:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 361

v=0
o=root 1771 1771 IN IP4 192.168.3.254
s=session
c=IN IP4 192.168.3.254
t=0 0
m=audio 18768 RTP/AVP 3 0 8 97 110 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
tagate*CLI>
<--- SIP read from 192.168.3.117:21424 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK2e48aacf;rport=5060
Contact: <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb>
To: <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb>;tag=b85f7f57
From: "403"<sip:403@192.168.3.254>;tag=as472e90c6
Call-ID: 701e364769da003321ffd67f3f3c4151@192.168.3.254
CSeq: 102 INVITE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
tagate*CLI>
<--- SIP read from 192.168.3.117:21424 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK2e48aacf;rport=5060
Contact: <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb>
To: <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb>;tag=b85f7f57
From: "403"<sip:403@192.168.3.254>;tag=as472e90c6
Call-ID: 701e364769da003321ffd67f3f3c4151@192.168.3.254
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 243

v=0
o=- 4 2 IN IP4 192.168.3.117
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.3.117
t=0 0
m=audio 1046 RTP/AVP 3 0 8 97 110 101
a=fmtp:101 0-15
a=rtpmap:97 iLBC/8000
a=rtpmap:110 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (11 headers 11 lines) ---
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 101
Found audio description format iLBC for ID 97
Found audio description format SPEEX for ID 110
Found audio description format telephone-event for ID 101
Capabilities: us - 0x60e (gsm|ulaw|alaw|speex|ilbc), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - (gsm|ulaw|alaw|ilbc|speex)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.3.117:1046
list_route: hop: <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb>
set_destination: Parsing <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb> for address/port to send to
set_destination: set destination to 192.168.3.117, port 21424
Transmitting (NAT) to 192.168.3.117:21424:
ACK sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb SIP/2.0
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK10a1533d;rport
From: "403" <sip:403@192.168.3.254>;tag=as472e90c6
To: <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb>;tag=b85f7f57
Contact: <sip:403@192.168.3.254>
Call-ID: 701e364769da003321ffd67f3f3c4151@192.168.3.254
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '701e364769da003321ffd67f3f3c4151@192.168.3.254' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb> for address/port to send to
set_destination: set destination to 192.168.3.117, port 21424
Reliably Transmitting (NAT) to 192.168.3.117:21424:
BYE sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb SIP/2.0
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK30277766;rport
From: "403" <sip:403@192.168.3.254>;tag=as472e90c6
To: <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb>;tag=b85f7f57
Call-ID: 701e364769da003321ffd67f3f3c4151@192.168.3.254
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---
tagate*CLI>
<--- SIP read from 192.168.3.117:21424 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK30277766;rport=5060
Contact: <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb>
To: <sip:6000@192.168.3.117:21424;rinstance=bbba878c73ad09eb>;tag=b85f7f57
From: "403"<sip:403@192.168.3.254>;tag=as472e90c6
Call-ID: 701e364769da003321ffd67f3f3c4151@192.168.3.254
CSeq: 103 BYE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '701e364769da003321ffd67f3f3c4151@192.168.3.254' Method: INVITE
[Apr 29 16:59:08] NOTICE[1771]: chan_sip.c:8083 sip_reg_timeout: -- Registration for '6000@dynamic' timed out, trying again (Attempt #2)
tagate*CLI> sip set debug off

если пир сказал BYE , значит так тому и быть. И зачем нат используется , если сервер в той же сети? Да и вообще, топологию в студию!

Нат используется для того, что я иногда присоединяюсь из-за ната.
Вот топология

смотрите кодеки

Про кодеки я тоже думал, разрешил все. Ситуация не изменилась... Если настроить вызов напрямую к user'у, то все нормально соединяется...

разрешать для начала надо не все, а опеределенные. А то тут у вас и ilbc и gsm смешались. Поставьте и для транка, и для пира
disallow=all
allow=ulaw
и дебажьте

поставил, ситуация та же

вот дебаг

tagate*CLI> sip set debug peer 6000
SIP Debugging Enabled for IP: 192.168.3.117:12850
tagate*CLI>
<--- SIP read from 192.168.3.117:12850 --->



<------------->
[May 4 14:53:39] WARNING[78647]: ast_expr2.fl:415 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input:
= 1
^
[May 4 14:53:39] WARNING[78647]: ast_expr2.fl:419 ast_yyerror: If you have questions, please refer to doc/channelvariables.txt in the asterisk source.
Audio is at 192.168.3.254 port 14082
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.3.117:12850:
INVITE sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b SIP/2.0
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK1b185064;rport
From: "403" <sip:403@192.168.3.254>;tag=as218296fa
To: <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b>
Contact: <sip:403@192.168.3.254>
Call-ID: 223a3b942b55a0bf32d954c62ceba681@192.168.3.254
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 04 May 2010 08:53:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 78647 78647 IN IP4 192.168.3.254
s=session
c=IN IP4 192.168.3.254
t=0 0
m=audio 14082 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[May 4 14:53:39] WARNING[78647]: app_dial.c:1296 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
tagate*CLI>
<--- SIP read from 192.168.3.117:12850 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK1b185064;rport=5060
Contact: <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b>
To: <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b>;tag=9e776326
From: "403"<sip:403@192.168.3.254>;tag=as218296fa
Call-ID: 223a3b942b55a0bf32d954c62ceba681@192.168.3.254
CSeq: 102 INVITE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
tagate*CLI>
<--- SIP read from 192.168.3.117:12850 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK1b185064;rport=5060
Contact: <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b>
To: <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b>;tag=9e776326
From: "3532519403"<sip:403@192.168.3.254>;tag=as218296fa
Call-ID: 223a3b942b55a0bf32d954c62ceba681@192.168.3.254
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 189

v=0
o=- 7 2 IN IP4 192.168.3.117
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.3.117
t=0 0
m=audio 40294 RTP/AVP 0 3 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - (ulaw|gsm|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.3.117:40294
list_route: hop: <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b>
set_destination: Parsing <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b> for address/port to send to
set_destination: set destination to 192.168.3.117, port 12850
Transmitting (NAT) to 192.168.3.117:12850:
ACK sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b SIP/2.0
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK1b939307;rport
From: "403" <sip:403@192.168.3.254>;tag=as218296fa
To: <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b>;tag=9e776326
Contact: <sip:3532519403@192.168.3.254>
Call-ID: 223a3b942b55a0bf32d954c62ceba681@192.168.3.254
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Scheduling destruction of SIP dialog '223a3b942b55a0bf32d954c62ceba681@192.168.3.254' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b> for address/port to send to
set_destination: set destination to 192.168.3.117, port 12850
Reliably Transmitting (NAT) to 192.168.3.117:12850:
BYE sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b SIP/2.0
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK568417e6;rport
From: "403" <sip:403@192.168.3.254>;tag=as218296fa
To: <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b>;tag=9e776326
Call-ID: 223a3b942b55a0bf32d954c62ceba681@192.168.3.254
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
ontent-Length: 0


---
tagate*CLI>
<--- SIP read from 192.168.3.117:12850 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.254:5060;branch=z9hG4bK568417e6;rport=5060
Contact: <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b>
To: <sip:6000@192.168.3.117:12850;rinstance=11fe2baf1397e58b>;tag=9e776326
From: "403"<sip:403@192.168.3.254>;tag=as218296fa
Call-ID: 223a3b942b55a0bf32d954c62ceba681@192.168.3.254
CSeq: 103 BYE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '223a3b942b55a0bf32d954c62ceba681@192.168.3.254' Method: INVITE

v=0
o=- 7 2 IN IP4 192.168.3.117
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.3.117
t=0 0
m=audio 40294 RTP/AVP 0 3 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

Где здесь кодек?