создал внутренний номер 007 подключился к нему через IP телефон
создал SIP транк на UTEL
voip*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
nugngn.usi.ru:5060 N 201901 105 Registered Sat, 25 Dec 2010 11:05:16
1 SIP registrations.
Как понял что работает.

Исходящий маршрут, все оставил по дефолту. добавил маску XXXXXX и NXXXXX последовательность набора первым поставил мой sip транк.

В SIP транке тоже сделал маски набора NXXXXX XXXXXX

В общем по идее должно звонить на улицу по идее. а в итоге :

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [250823@from-internal:1] Macro("SIP/007-00000011", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/007-00000011", "AMPUSER=007") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/007-00000011", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/007-00000011", "1?Set(REALCALLERIDNUM=007)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/007-00000011", "AMPUSER=007") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/007-00000011", "AMPUSERCIDNAME=Aleksey Piletckiy") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/007-00000011", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/007-00000011", "AMPUSERCID=007") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/007-00000011", "CALLERID(all)="Aleksey Piletckiy" <007>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/007-00000011", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/007-00000011", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/007-00000011", "Using CallerID "Aleksey Piletckiy" <007>") in new stack
-- Executing [250823@from-internal:2] Set("SIP/007-00000011", "_NODEST=") in new stack
-- Executing [250823@from-internal:3] Macro("SIP/007-00000011", "record-enable,007,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/007-00000011", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] ExecIf("SIP/007-00000011", "0?MacroExit()") in new stack
-- Executing [s@macro-record-enable:5] GotoIf("SIP/007-00000011", "0?Group:OUT") in new stack
-- Goto (macro-record-enable,s,15)
-- Executing [s@macro-record-enable:15] GotoIf("SIP/007-00000011", "0?IN") in new stack
-- Executing [s@macro-record-enable:16] ExecIf("SIP/007-00000011", "1?MacroExit()") in new stack
-- Executing [250823@from-internal:4] Macro("SIP/007-00000011", "dialout-trunk,2,250823,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/007-00000011", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/007-00000011", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/007-00000011", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/007-00000011", "DIAL_NUMBER=250823") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/007-00000011", "DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/007-00000011", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/007-00000011", "0?nomax") in new stack
-- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/007-00000011", "0?chanfull") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/007-00000011", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/007-00000011", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/007-00000011", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/007-00000011", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/007-00000011", "0?Set(REALCALLERIDNUM=007)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/007-00000011", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/007-00000011", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/007-00000011", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/007-00000011", "TRUNKOUTCID=201901") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/007-00000011", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/007-00000011", "1?Set(CALLERID(all)=201901)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/007-00000011", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/007-00000011", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:15] ExecIf("SIP/007-00000011", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/007-00000011", "1?AGI(fixlocalprefix)") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
== fixlocalprefix: Dialpattern NXXXXX matched. 250823 -> 250823
-- <SIP/007-00000011>AGI Script fixlocalprefix completed, returning 0
-- Executing [s@macro-dialout-trunk:13] Set("SIP/007-00000011", "OUTNUM=250823") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/007-00000011", "custom=SIP/201901") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/007-00000011", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/007-00000011", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/007-00000011", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/007-00000011", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/007-00000011", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/007-00000011", "SIP/201901/250823,300,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 201901/250823
-- SIP/201901-00000012 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [s@macro-dialout-trunk:20] NoOp("SIP/007-00000011", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:21] Goto("SIP/007-00000011", "s-CONGESTION,1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/007-00000011", "RC=21") in new stack
-- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/007-00000011", "21,1") in new stack
-- Goto (macro-dialout-trunk,21,1)
-- Executing [21@macro-dialout-trunk:1] Goto("SIP/007-00000011", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/007-00000011", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,continue,3)
-- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/007-00000011", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:4] Set("SIP/007-00000011", "CALLERID(number)=007") in new stack
-- Executing [250823@from-internal:5] Macro("SIP/007-00000011", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/007-00000011", "") in new stack
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/007-00000011", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/007-00000011", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/007-00000011", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/007-00000011> Playing 'all-circuits-busy-now.gsm' (language 'en')
-- <SIP/007-00000011> Playing 'pls-try-call-later.gsm' (language 'en')
-- Executing [s@macro-outisbusy:5] Congestion("SIP/007-00000011", "20") in new stack
== Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'SIP/007-00000011' in macro 'outisbusy'
== Spawn extension (from-internal, 250823, 5) exited non-zero on 'SIP/007-00000011'
-- Executing [h@from-internal:1] Macro("SIP/007-00000011", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/007-00000011", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/007-00000011", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/007-00000011", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/007-00000011", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/007-00000011", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/007-00000011", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/007-00000011' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/007-00000011'


при входящем звонке пишет
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5

Что Это, кодек какой то что ли ?

опции для PEER:
host=nugngn.usi.ru
username=201901
secret=ХХХХХХ
type=peer

в SIP транке макси делать не надо, это для других целей
уверены, что провайдер пропустит звонки XXXXXX?
вообще - сниффер или sip debug поможет выяснить, почему пров не хочет принимать вызов

ХХХХХХ - я сделал чтоб наверняка работало все цифры убегали.
В общем при звонке с улице получаю вот что:

CSeq: 5859 REGISTER
Contact: <sip:007@192.168.100.68:5060>
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (no NAT)

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK3034867581766030529;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5859 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK3034867581766030529;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>;tag=as5b0036ae
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5859 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="43851e45"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.100.68:5060 --->
REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2028716029890824009;rport
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5860 REGISTER
Contact: <sip:007@192.168.100.68:5060>
Authorization: Digest username="007", realm="asterisk", nonce="43851e45", uri="sip:192.168.100.44:5060", response="f00f064f84229e843d47d7a0768678de", algorithm=MD5
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2028716029890824009;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5860 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
Reliably Transmitting (NAT) to 192.168.100.68:5060:
OPTIONS sip:007@192.168.100.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK38131ae1;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.100.44>;tag=as5e73e59a
To: <sip:007@192.168.100.68:5060>
Contact: <sip:Unknown@192.168.100.44>
Call-ID: 1ba0fa5023c0b8a37043b2794ee4e1f2@192.168.100.44
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 27 Dec 2010 05:35:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2028716029890824009;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>;tag=as5b0036ae
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5860 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:007@192.168.100.68:5060>;expires=60
Date: Mon, 27 Dec 2010 05:35:38 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.100.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK38131ae1;rport
From: "Unknown" <sip:Unknown@192.168.100.44>;tag=as5e73e59a
To: <sip:007@192.168.100.68:5060>
Call-ID: 1ba0fa5023c0b8a37043b2794ee4e1f2@192.168.100.44
CSeq: 102 OPTIONS
Contact: <sip:192.168.100.68:5060>
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '1ba0fa5023c0b8a37043b2794ee4e1f2@192.168.100.44' Method: OPTIONS
Really destroying SIP dialog '59776e52d291f11b20e85e74777a17eddf018a@62.148.237.152' Method: ACK
Really destroying SIP dialog '321181390410115-2516193436649@192.168.100.68' Method: REGISTER

<--- SIP read from UDP:192.168.100.68:5060 --->
REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK30780121931304228956;rport
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5861 REGISTER
Contact: <sip:007@192.168.100.68:5060>
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (no NAT)

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK30780121931304228956;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5861 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

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

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK30780121931304228956;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>;tag=as42d175cb
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5861 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3be3adb3"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.100.68:5060 --->
REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2959992612496529687;rport
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5862 REGISTER
Contact: <sip:007@192.168.100.68:5060>
Authorization: Digest username="007", realm="asterisk", nonce="3be3adb3", uri="sip:192.168.100.44:5060", response="7e57f8163fe43922cd90e04b82aa8194", algorithm=MD5
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2959992612496529687;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5862 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
Reliably Transmitting (NAT) to 192.168.100.68:5060:
OPTIONS sip:007@192.168.100.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK3c8be1d6;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.100.44>;tag=as575d4fae
To: <sip:007@192.168.100.68:5060>
Contact: <sip:Unknown@192.168.100.44>
Call-ID: 0d1223673bb0aa7d78e8a7737c23797e@192.168.100.44
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 27 Dec 2010 05:36:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2959992612496529687;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>;tag=as42d175cb
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5862 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:007@192.168.100.68:5060>;expires=60
Date: Mon, 27 Dec 2010 05:36:36 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.100.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK3c8be1d6;rport
From: "Unknown" <sip:Unknown@192.168.100.44>;tag=as575d4fae
To: <sip:007@192.168.100.68:5060>
Call-ID: 0d1223673bb0aa7d78e8a7737c23797e@192.168.100.44
CSeq: 102 OPTIONS
Contact: <sip:192.168.100.68:5060>
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '0d1223673bb0aa7d78e8a7737c23797e@192.168.100.44' Method: OPTIONS
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 62.148.237.152:5060:
REGISTER sip:nugngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK751a985a;rport
Max-Forwards: 70
From: <sip:201901@nugngn.usi.ru>;tag=as6db62f0b
To: <sip:201901@nugngn.usi.ru>
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 106 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Authorization: Digest username="201901", realm="Realm", algorithm=MD5, uri="sip:nugngn.usi.ru", nonce="MTI5MzQyMDc2NTg3MjAxNmMyN2Y1NGI2OWQ4N2E2NjBiMGUxMmI5Njc4OTQw", response="8a54c50dedc33416ba798d6d32e2601e", qop=auth, cnonce="178dc8f6", nc=00000004
Expires: 120
Contact: <sip:s@192.168.100.44>
Content-Length: 0

---

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 100 Trying
From: <sip:201901@nugngn.usi.ru>;tag=as6db62f0b
To: <sip:201901@nugngn.usi.ru>
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 106 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=51130;branch=z9hG4bK751a985a
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 407 Proxy Authentication Required
From: <sip:201901@nugngn.usi.ru>;tag=as6db62f0b
To: <sip:201901@nugngn.usi.ru>;tag=863400534
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 106 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=51130;branch=z9hG4bK751a985a
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
proxy-authenticate: Digest realm="Realm",nonce="MTI5MzQyMTA3NzYyNjIwMzk4NzM1Mjg3ZGIzNjRkNzI2Yjc5YWE4YzRkNjAy",stale=true,algorithm=MD5,qop="auth,auth-int"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Responding to challenge, registration to domain/host name nugngn.usi.ru
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 62.148.237.152:5060:
REGISTER sip:nugngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK15c63bf4;rport
Max-Forwards: 70
From: <sip:201901@nugngn.usi.ru>;tag=as7fc9a353
To: <sip:201901@nugngn.usi.ru>
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 107 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Proxy-Authorization: Digest username="201901", realm="Realm", algorithm=MD5, uri="sip:nugngn.usi.ru", nonce="MTI5MzQyMTA3NzYyNjIwMzk4NzM1Mjg3ZGIzNjRkNzI2Yjc5YWE4YzRkNjAy", response="27f6f4f1bffb20c6478ff65d313e050b", qop=auth, cnonce="709a908f", nc=00000001
Expires: 120
Contact: <sip:s@192.168.100.44>
Content-Length: 0
---

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 100 Trying
From: <sip:201901@nugngn.usi.ru>;tag=as7fc9a353
To: <sip:201901@nugngn.usi.ru>
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 107 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=51130;branch=z9hG4bK15c63bf4
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 200 Registration Successful
From: "201901 201901"<sip:201901@nugngn.usi.ru>;tag=as7fc9a353
To: <sip:201901@nugngn.usi.ru>;tag=266258229
Call-ID: 356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1
CSeq: 107 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=51130;branch=z9hG4bK15c63bf4
contact: <sip:s@192.168.100.44:50002>;expires=4,<sip:s@192.168.100.44:5060>;expires=116
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '321181390410115-2516193436649@192.168.100.68' Method: REGISTER
Really destroying SIP dialog '356b74c77f08615e1fec7ad77bb3b0b1@127.0.0.1' Method: REGISTER

<--- SIP read from UDP:192.168.100.68:5060 --->
REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK1999646311544822368;rport
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5863 REGISTER
Contact: <sip:007@192.168.100.68:5060>
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (no NAT)

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK1999646311544822368;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5863 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

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

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK1999646311544822368;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>;tag=as2c004b17
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5863 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="73839620"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.100.68:5060 --->
REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK4286254401256713458;rport
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5864 REGISTER
Contact: <sip:007@192.168.100.68:5060>
Authorization: Digest username="007", realm="asterisk", nonce="73839620", uri="sip:192.168.100.44:5060", response="806a1fc3ebaeb031db0453b9faeff5c5", algorithm=MD5
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK4286254401256713458;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5864 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
Reliably Transmitting (NAT) to 192.168.100.68:5060:
OPTIONS sip:007@192.168.100.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK5e4c21e3;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.100.44>;tag=as64e5b805
To: <sip:007@192.168.100.68:5060>
Contact: <sip:Unknown@192.168.100.44>
Call-ID: 4d69381a4d3064d949b88dc4147cf772@192.168.100.44
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 27 Dec 2010 05:37:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

---

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK4286254401256713458;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>;tag=as2c004b17
Call-ID: 321181390410115-2516193436649@192.168.100.68
CSeq: 5864 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:007@192.168.100.68:5060>;expires=60
Date: Mon, 27 Dec 2010 05:37:33 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '321181390410115-2516193436649@192.168.100.68' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.100.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK5e4c21e3;rport
From: "Unknown" <sip:Unknown@192.168.100.44>;tag=as64e5b805
To: <sip:007@192.168.100.68:5060>
Call-ID: 4d69381a4d3064d949b88dc4147cf772@192.168.100.44
CSeq: 102 OPTIONS
Contact: <sip:192.168.100.68:5060>
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '4d69381a4d3064d949b88dc4147cf772@192.168.100.44' Method: OPTIONS


Если изнутри:
<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK19248284741378116562;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=2409814001
To: "250823" <sip:250823@192.168.100.44:5060>;tag=as354dfe1d
Call-ID: 299762408519676-28830234116418@192.168.100.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:250823@192.168.100.44>
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1056850601 1056850601 IN IP4 192.168.100.44
s=Asterisk PBX 1.6.2.13
c=IN IP4 192.168.100.44
t=0 0
m=audio 17154 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
-- Executing [s@macro-outisbusy:2] GotoIf("SIP/007-0000001d", "0?emergency,1") in new stack
-- Executing [s@macro-outisbusy:3] GotoIf("SIP/007-0000001d", "0?intracompany,1") in new stack
-- Executing [s@macro-outisbusy:4] Playback("SIP/007-0000001d", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
-- <SIP/007-0000001d> Playing 'all-circuits-busy-now.gsm' (language 'en')
Really destroying SIP dialog '32feaa0462b38e4f720d9c7303589d33@192.168.100.44' Method: INVITE
-- <SIP/007-0000001d> Playing 'pls-try-call-later.gsm' (language 'en')

<--- SIP read from UDP:192.168.100.68:5060 --->
CANCEL sip:250823@192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK19248284741378116562;rport
From: 007 <sip:007@192.168.100.44:5060>;tag=2409814001
To: "250823" <sip:250823@192.168.100.44:5060>
Call-ID: 299762408519676-28830234116418@192.168.100.68
CSeq: 2 CANCEL
Max-Forwards: 70
User-Agent: Voip Phone 1.0
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (NAT)

<--- Reliably Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK19248284741378116562;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=2409814001
To: "250823" <sip:250823@192.168.100.44:5060>;tag=as354dfe1d
Call-ID: 299762408519676-28830234116418@192.168.100.68
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

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

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK19248284741378116562;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=2409814001
To: "250823" <sip:250823@192.168.100.44:5060>;tag=as354dfe1d
Call-ID: 299762408519676-28830234116418@192.168.100.68
CSeq: 2 CANCEL
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/007-0000001d' in macro 'outisbusy'
== Spawn extension (from-internal, 250823, 5) exited non-zero on 'SIP/007-0000001d'
-- Executing [h@from-internal:1] Macro("SIP/007-0000001d", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/007-0000001d", "1?noautomon") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] NoOp("SIP/007-0000001d", "TOUCH_MONITOR_OUTPUT=") in new stack
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/007-0000001d", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/007-0000001d", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,10)
-- Executing [s@macro-hangupcall:10] GotoIf("SIP/007-0000001d", "1?theend") in new stack
-- Goto (macro-hangupcall,s,12)
-- Executing [s@macro-hangupcall:12] Hangup("SIP/007-0000001d", "") in new stack
== Spawn extension (macro-hangupcall, s, 12) exited non-zero on 'SIP/007-0000001d' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/007-0000001d'

<--- SIP read from UDP:192.168.100.68:5060 --->
ACK sip:250823@192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK19248284741378116562;rport
From: 007 <sip:007@192.168.100.44:5060>;tag=2409814001
To: "250823" <sip:250823@192.168.100.44:5060>;tag=as354dfe1d
Call-ID: 299762408519676-28830234116418@192.168.100.68
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0

Ну Это не весь кусок, писанины много, что в терминалке стока строк не отображает.

никто не может объяснить в чем косяк ?

в общем сделал вот так, где то в нете нашел такое.
host=nugngn.usi.ru
username=201901
secret=++++++
insecure=invite
dtmfmode=rfc2833
type=friend
fromdomain=nugngn.usi.ru
outboundproxy=62.148.237.152

Вуаля.
Провайдер UTEL

И почему то звонит то не звонит, так же и входящие

Вот что пишет на входящий.
Really destroying SIP dialog '72d6d4ab35827331750e6752577ccf87@192.168.100.44' Method: OPTIONS
Really destroying SIP dialog '67672746522226-229372301815489@192.168.100.68' Method: REGISTER

<--- SIP read from UDP:192.168.100.68:5060 --->
REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK706127485295087026;rport
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1969 REGISTER
Contact: <sip:007@192.168.100.68:5060>
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (no NAT)

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK706127485295087026;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1969 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK706127485295087026;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>;tag=as11634ef1
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1969 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07969da8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '67672746522226-229372301815489@192.168.100.68' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.100.68:5060 --->
REGISTER sip:192.168.100.44:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2215923361990026385;rport
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1970 REGISTER
Contact: <sip:007@192.168.100.68:5060>
Authorization: Digest username="007", realm="asterisk", nonce="07969da8", uri="sip:192.168.100.44:5060", response="95d97a35e526118592c4936d35823126", algorithm=MD5
Max-Forwards: 70
Expires: 60
Supported: path
User-Agent: Voip Phone 1.0
Content-Length: 0


<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.100.68 : 5060 (NAT)

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2215923361990026385;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1970 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Reliably Transmitting (NAT) to 192.168.100.68:5060:
OPTIONS sip:007@192.168.100.68:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK385d8d46;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.100.44>;tag=as7f140993
To: <sip:007@192.168.100.68:5060>
Contact: <sip:Unknown@192.168.100.44>
Call-ID: 011d5a6b785d3e2264052b681bf1ba47@192.168.100.44
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Wed, 29 Dec 2010 05:22:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to 192.168.100.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.68:5060;branch=z9hG4bK2215923361990026385;received=192.168.100.68;rport=5060
From: 007 <sip:007@192.168.100.44:5060>;tag=974921936
To: 007 <sip:007@192.168.100.44:5060>;tag=as11634ef1
Call-ID: 67672746522226-229372301815489@192.168.100.68
CSeq: 1970 REGISTER
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 60
Contact: <sip:007@192.168.100.68:5060>;expires=60
Date: Wed, 29 Dec 2010 05:22:33 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '67672746522226-229372301815489@192.168.100.68' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.100.68:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK385d8d46;rport
From: "Unknown" <sip:Unknown@192.168.100.44>;tag=as7f140993
To: <sip:007@192.168.100.68:5060>
Call-ID: 011d5a6b785d3e2264052b681bf1ba47@192.168.100.44
CSeq: 102 OPTIONS
Contact: <sip:192.168.100.68:5060>
Supported: 100rel, replaces, timer
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Accept: application/sdp, message/sipfrag, application/dtmf-relay
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '011d5a6b785d3e2264052b681bf1ba47@192.168.100.44' Method: OPTIONS

<--- SIP read from UDP:62.148.237.152:5060 --->
CANCEL sip:s@192.168.100.44:5060;maddr=188.19.10.72 SIP/2.0
From: <sip:3463250823@inc-out-hnt.usi.ru:5060;user=phone>;tag=-45026-41d8732-4a2142f5-41d8732
To: <sip:3463201901@inc-out-hnt.usi.ru:5060;user=phone>
Call-ID: 6caf7203d227708c20e85e3c7b6aa7786353a3@62.148.237.152
CSeq: 1 CANCEL
Via: SIP/2.0/UDP 62.148.237.152:5060;branch=z9hG4bK-23dbc6-8c127d8d-60dd155f
Max-Forwards: 70
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

<--- Transmitting (no NAT) to 62.148.237.152:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 62.148.237.152:5060;branch=z9hG4bK-23dbc6-8c127d8d-60dd155f;received=62.148.237.152
From: <sip:3463250823@inc-out-hnt.usi.ru:5060;user=phone>;tag=-45026-41d8732-4a2142f5-41d8732
To: <sip:3463201901@inc-out-hnt.usi.ru:5060;user=phone>;tag=as4c82ab73
Call-ID: 6caf7203d227708c20e85e3c7b6aa7786353a3@62.148.237.152
CSeq: 1 CANCEL
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 62.148.237.152:5060:
REGISTER sip:nugngn.usi.ru SIP/2.0
Via: SIP/2.0/UDP 192.168.100.44:5060;branch=z9hG4bK672cbbab;rport
Max-Forwards: 70
From: <sip:201901@nugngn.usi.ru>;tag=as00228927
To: <sip:201901@nugngn.usi.ru>
Call-ID: 32c26aad029c7b9b15e2cc1148899fae@127.0.0.1
CSeq: 109 REGISTER
User-Agent: Asterisk PBX 1.6.2.13
Authorization: Digest username="201901", realm="Realm", algorithm=MD5, uri="sip:nugngn.usi.ru", nonce="MTI5MzU5Mjg4MzQxN2Q2NWY4YTFhNWFhZDJjN2E2ZDUzMjY2MTk1OTg3YzE4", response="8a040867f9d2114fbbf519f0a3c8d507", qop=auth, cnonce="76666a32", nc=00000003
Expires: 120
Contact: <sip:s@192.168.100.44>
Content-Length: 0


---

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 100 Trying
From: <sip:201901@nugngn.usi.ru>;tag=as00228927
To: <sip:201901@nugngn.usi.ru>
Call-ID: 32c26aad029c7b9b15e2cc1148899fae@127.0.0.1
CSeq: 109 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=55334;branch=z9hG4bK672cbbab
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:62.148.237.152:5060 --->
SIP/2.0 200 Registration Successful
From: "201901 201901"<sip:201901@nugngn.usi.ru>;tag=as00228927
To: <sip:201901@nugngn.usi.ru>;tag=1010258005
Call-ID: 32c26aad029c7b9b15e2cc1148899fae@127.0.0.1
CSeq: 109 REGISTER
Via: SIP/2.0/UDP 192.168.100.44:5060;rport=55334;branch=z9hG4bK672cbbab
contact: <sip:s@19192.168.100.44:5060>;expires=111
supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Scheduling destruction of SIP dialog '32c26aad029c7b9b15e2cc1148899fae@127.0.0.1' in 32000 ms (Method: REGISTER)

Не нужно выкладывать сюда километры логов,они никому не интересны
Тем более что логи эти к рассматриваемым событиям не относятся. В первую очередь включите debug в logger.conf для консоли:
console => notice,warning,error,debug, verbose
И система сама скажет, что у вас не так.

А вообще изучите SIP, попытайтесь сделать выводы самостоятельно.

Да мне кажется что сеть от UTEL глючит. т.к. напрямую подключаю телефон к акаунту, телефон звонит через раз, и не пашут ютеловские службы помощи.