вот так после того как снимешь и положешь убку на телефоне который не прекрощает звенеть, когда на первом трубка уже лежит



<--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK26c5f487;rport
Require: timer
Session-Expires: 1800;refresher=uac
From: "Jane Smith" <sip:5678@10.5.5.1>;tag=as0c9545ba
To: <sip:555@10.5.5.21>;tag=001E58F79F6F_T1772613033
Call-ID: 0ce90c59763e8845564a9a9822ae1ff5@10.5.5.1
Contact: <sip:555@10.5.5.21:5060>
CSeq: 103 INVITE
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Supported: 100rel,timer,replaces
Content-Type: application/sdp
Content-Length: 155

v=0
o=- 2064075099 2064075099 IN IP4 10.5.5.21
s=DPH-150S 01.03
c=IN IP4 10.5.5.21
t=0 0
m=audio 41000 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv

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

<--- SIP read from UDP://10.5.5.21:5060 --->
BYE sip:5678@10.5.5.1 SIP/2.0
Via: SIP/2.0/UDP 10.5.5.21:5060;branch=z9hG4bK_001E58F79F6F_T2A18FBA6
From: <sip:555@10.5.5.21>;tag=001E58F79F6F_T1772613033
To: "Jane Smith" <sip:5678@10.5.5.1>;tag=as0c9545ba
Call-ID: 0ce90c59763e8845564a9a9822ae1ff5@10.5.5.1
CSeq: 1 BYE
User-Agent: DPH-150S 01.03
Contact: <sip:555@10.5.5.21:5060>
Max-Forwards: 70
Content-Length: 0


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

<--- Transmitting (no NAT) to 10.5.5.21:5060 --->
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 10.5.5.21:5060;branch=z9hG4bK_001E58F79F6F_T2A18FBA6;received=10.5.5.21
From: <sip:555@10.5.5.21>;tag=001E58F79F6F_T1772613033
To: "Jane Smith" <sip:5678@10.5.5.1>;tag=as0c9545ba
Call-ID: 0ce90c59763e8845564a9a9822ae1ff5@10.5.5.1
CSeq: 1 BYE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


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

при звонке ( в сокращенном варианте вот так)
Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Executing [555@office:1] Dial("SIP/444-08263590", "SIP/555,20,rt") in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Called 555
-- SIP/555-08267a50 is ringing

после того как кладу трубку
== Spawn extension (office, 555, 1) exited non-zero on 'SIP/444-08263590'

вот в дебаг моде, только часть начала не влезла


v=0
o=- 2066868945 2066868945 IN IP4 10.5.5.20
s=DPH-150S 01.03
c=IN IP4 10.5.5.20
t=0 0
m=audio 41000 RTP/AVP 0 8 18
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=sendrecv

<------------->
--- (15 headers 11 lines) ---
Sending to 10.5.5.20 : 5060 (no NAT)
Using INVITE request as basis request - CALL_ID19_001E58F79F9F_T1314018807@10.5.5.20
Found peer '444' for '444' from 10.5.5.20:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Peer audio RTP is at port 10.5.5.20:41000
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.5.5.20:41000
Looking for 555 in office (domain )
list_route: hop: <sip:444@10.5.5.20:5060>

<--- Transmitting (no NAT) to 10.5.5.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.5.5.20:5060;branch=z9hG4bK_001E58F79F9F_T6346731D;received=10.5.5.20
From: "LEO" <sip:444@>;tag=001E58F79F9F_T150593785
To: <sip:555@>
Call-ID: CALL_ID19_001E58F79F9F_T1314018807@10.5.5.20
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:555@10.5.5.1>
Content-Length: 0


<------------>
-- Executing [555@office:1] Dial("SIP/444-08267080", "SIP/555,20,rt") in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
Audio is at 10.5.5.1 port 13904
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 (no NAT) to 10.5.5.21:5060:
INVITE sip:555@10.5.5.21:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK6b7a44d1;rport
Max-Forwards: 70
From: "Jane Smith" <sip:5678@10.5.5.1>;tag=as57d5df00
To: <sip:555@10.5.5.21:5060>
Contact: <sip:5678@10.5.5.1>
Call-ID: 355ed17469142dbb536d28ba25fcddfa@10.5.5.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Sun, 02 Aug 2009 13:14:26 GMT
Session-Expires: 600
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300

v=0
o=root 335363361 335363361 IN IP4 10.5.5.1
s=Asterisk PBX 1.6.1.1
c=IN IP4 10.5.5.1
t=0 0
m=audio 13904 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

---
-- Called 555

<--- Transmitting (no NAT) to 10.5.5.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.5.5.20:5060;branch=z9hG4bK_001E58F79F9F_T6346731D;received=10.5.5.20
From: "LEO" <sip:444@>;tag=001E58F79F9F_T150593785
To: <sip:555@>;tag=as303f1156
Call-ID: CALL_ID19_001E58F79F9F_T1314018807@10.5.5.20
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:555@10.5.5.1>
Content-Length: 0


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

<--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK6b7a44d1;rport
From: "Jane Smith" <sip:5678@10.5.5.1>;tag=as57d5df00
To: <sip:555@10.5.5.21>
Call-ID: 355ed17469142dbb536d28ba25fcddfa@10.5.5.1
Contact: <sip:555@10.5.5.21:5060>
CSeq: 102 INVITE
User-Agent: DPH-150S 01.03
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Content-Length: 0


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

<--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 422 Session Interval Too Small
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK6b7a44d1;rport
Min-SE: 1800
From: "Jane Smith" <sip:5678@10.5.5.1>;tag=as57d5df00
To: <sip:555@10.5.5.21>;tag=001E58F79F6F_T578849083
Call-ID: 355ed17469142dbb536d28ba25fcddfa@10.5.5.1
Contact: <sip:555@10.5.5.21:5060>
CSeq: 102 INVITE
User-Agent: DPH-150S 01.03
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 10.5.5.21:5060:
ACK sip:555@10.5.5.21:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK6b7a44d1;rport
Max-Forwards: 70
From: "Jane Smith" <sip:5678@10.5.5.1>;tag=as57d5df00
To: <sip:555@10.5.5.21:5060>;tag=001E58F79F6F_T578849083
Contact: <sip:5678@10.5.5.1>
Call-ID: 355ed17469142dbb536d28ba25fcddfa@10.5.5.1
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.1
Content-Length: 0


---
Audio is at 10.5.5.1 port 13904
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 (no NAT) to 10.5.5.21:5060:
INVITE sip:555@10.5.5.21:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK54d43173;rport
Max-Forwards: 70
From: "Jane Smith" <sip:5678@10.5.5.1>;tag=as57d5df00
To: <sip:555@10.5.5.21:5060>
Contact: <sip:5678@10.5.5.1>
Call-ID: 355ed17469142dbb536d28ba25fcddfa@10.5.5.1
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Sun, 02 Aug 2009 13:14:26 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300

v=0
o=root 335363361 335363362 IN IP4 10.5.5.1
s=Asterisk PBX 1.6.1.1
c=IN IP4 10.5.5.1
t=0 0
m=audio 13904 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

---

<--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK54d43173;rport
From: "Jane Smith" <sip:5678@10.5.5.1>;tag=as57d5df00
To: <sip:555@10.5.5.21>
Call-ID: 355ed17469142dbb536d28ba25fcddfa@10.5.5.1
Contact: <sip:555@10.5.5.21:5060>
CSeq: 103 INVITE
User-Agent: DPH-150S 01.03
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Content-Length: 0


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

<--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK54d43173;rport
From: "Jane Smith" <sip:5678@10.5.5.1>;tag=as57d5df00
To: <sip:555@10.5.5.21>;tag=001E58F79F6F_T855061335
Call-ID: 355ed17469142dbb536d28ba25fcddfa@10.5.5.1
Contact: <sip:555@10.5.5.21:5060>
CSeq: 103 INVITE
User-Agent: DPH-150S 01.03
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
-- SIP/555-082ef920 is ringing

<--- Transmitting (no NAT) to 10.5.5.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.5.5.20:5060;branch=z9hG4bK_001E58F79F9F_T6346731D;received=10.5.5.20
From: "LEO" <sip:444@>;tag=001E58F79F9F_T150593785
To: <sip:555@>;tag=as303f1156
Call-ID: CALL_ID19_001E58F79F9F_T1314018807@10.5.5.20
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:555@10.5.5.1>
Content-Length: 0


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

1) Делайте дебаг для себя, я ж не просил его сюда вываливать?
2) У Вас что-то непонятное с пирами, звоните с 444 на 555 а в заголовках везде присутствует какой-то
From: "Jane Smith" <sip:5678@10.5.5.1> - что это?
Соотевтственно, когда ложится трубка - ответ BYE идёт не на sip:444@10.5.5.20 а на
неведомый 5678:
From: <sip:555@10.5.5.21>;tag=001E58F79F6F_T1772613033
To: "Jane Smith" <sip:5678@10.5.5.1>;tag=as0c9545ba
на что Астериск резонно сообщает, что
SIP/2.0 481 Call leg/transaction does not exist

При правильных настройках ИП телефона вызов INVITE должен быть не от
From: "Jane Smith" <sip:5678@10.5.5.1>
а от
From: "LEO" <sip:444@10.5.5.20>
тогда и отбиваться будет нормпльно.

Вы наверно накказывали на DPH-150s в настройках всё, что под руку попалось? Типа outbound proxy, register proxy?

Да соггласен забыл закомментировать , caller id поправил

-- Executing [555@office:1] Dial("SIP/444-08263790", "SIP/555,20,rt") in new stack
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
Audio is at 10.5.5.1 port 19490
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 (no NAT) to 10.5.5.21:5060:
INVITE sip:555@10.5.5.21:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK16195ed8;rport
Max-Forwards: 70
From: "444" <sip:444@10.5.5.1>;tag=as7b979fec
To: <sip:555@10.5.5.21:5060>
Contact: <sip:444@10.5.5.1>
Call-ID: 10a30f2518130a493b5bd89f416ca3c2@10.5.5.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Sun, 02 Aug 2009 13:43:31 GMT
Session-Expires: 600
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300

v=0
o=root 458427624 458427624 IN IP4 10.5.5.1
s=Asterisk PBX 1.6.1.1
c=IN IP4 10.5.5.1
t=0 0
m=audio 19490 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

---
-- Called 555

<--- Transmitting (no NAT) to 10.5.5.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.5.5.20:5060;branch=z9hG4bK_001E58F79F9F_T16CFE5AA;received=10.5.5.20
From: "LEO" <sip:444@>;tag=001E58F79F9F_T605439666
To: <sip:555@>;tag=as43d25e39
Call-ID: CALL_ID3_001E58F79F9F_T1912806211@10.5.5.20
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:555@10.5.5.1>
Content-Length: 0


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

<--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK16195ed8;rport
From: "444" <sip:444@10.5.5.1>;tag=as7b979fec
To: <sip:555@10.5.5.21>
Call-ID: 10a30f2518130a493b5bd89f416ca3c2@10.5.5.1
Contact: <sip:555@10.5.5.21:5060>
CSeq: 102 INVITE
User-Agent: DPH-150S 01.03
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Content-Length: 0


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

<--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 422 Session Interval Too Small
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK16195ed8;rport
Min-SE: 1800
From: "444" <sip:444@10.5.5.1>;tag=as7b979fec
To: <sip:555@10.5.5.21>;tag=001E58F79F6F_T1771820413
Call-ID: 10a30f2518130a493b5bd89f416ca3c2@10.5.5.1
Contact: <sip:555@10.5.5.21:5060>
CSeq: 102 INVITE
User-Agent: DPH-150S 01.03
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 10.5.5.21:5060:
ACK sip:555@10.5.5.21:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK16195ed8;rport
Max-Forwards: 70
From: "444" <sip:444@10.5.5.1>;tag=as7b979fec
To: <sip:555@10.5.5.21:5060>;tag=001E58F79F6F_T1771820413
Contact: <sip:444@10.5.5.1>
Call-ID: 10a30f2518130a493b5bd89f416ca3c2@10.5.5.1
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.1
Content-Length: 0


---
Audio is at 10.5.5.1 port 19490
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 (no NAT) to 10.5.5.21:5060:
INVITE sip:555@10.5.5.21:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK5eecd775;rport
Max-Forwards: 70
From: "444" <sip:444@10.5.5.1>;tag=as7b979fec
To: <sip:555@10.5.5.21:5060>
Contact: <sip:444@10.5.5.1>
Call-ID: 10a30f2518130a493b5bd89f416ca3c2@10.5.5.1
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.1.1
Date: Sun, 02 Aug 2009 13:43:31 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 300

v=0
o=root 458427624 458427625 IN IP4 10.5.5.1
s=Asterisk PBX 1.6.1.1
c=IN IP4 10.5.5.1
t=0 0
m=audio 19490 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

---

<--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK5eecd775;rport
From: "444" <sip:444@10.5.5.1>;tag=as7b979fec
To: <sip:555@10.5.5.21>
Call-ID: 10a30f2518130a493b5bd89f416ca3c2@10.5.5.1
Contact: <sip:555@10.5.5.21:5060>
CSeq: 103 INVITE
User-Agent: DPH-150S 01.03
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Content-Length: 0


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

<--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK5eecd775;rport
From: "444" <sip:444@10.5.5.1>;tag=as7b979fec
To: <sip:555@10.5.5.21>;tag=001E58F79F6F_T340828856
Call-ID: 10a30f2518130a493b5bd89f416ca3c2@10.5.5.1
Contact: <sip:555@10.5.5.21:5060>
CSeq: 103 INVITE
User-Agent: DPH-150S 01.03
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
-- SIP/555-08267ea0 is ringing

<--- Transmitting (no NAT) to 10.5.5.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.5.5.20:5060;branch=z9hG4bK_001E58F79F9F_T16CFE5AA;received=10.5.5.20
From: "LEO" <sip:444@>;tag=001E58F79F9F_T605439666
To: <sip:555@>;tag=as43d25e39
Call-ID: CALL_ID3_001E58F79F9F_T1912806211@10.5.5.20
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:555@10.5.5.1>
Content-Length: 0


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

а вот это после того как кладу трубку на телефоне с которого звоню

--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK2d457eee;rport
Require: timer
Session-Expires: 1800;refresher=uac
From: "444" <sip:444@10.5.5.1>;tag=as1081e8e6
To: <sip:555@10.5.5.21>;tag=001E58F79F6F_T133753141
Call-ID: 3cc3e3ce649a790246af93a30da6df71@10.5.5.1
Contact: <sip:555@10.5.5.21:5060>
CSeq: 103 INVITE
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,REFER,DO,UPDATE,OPTIONS,SUBSCRIBE,PRACK,INFO
Supported: 100rel,timer,replaces
Content-Type: application/sdp
Content-Length: 155

v=0
o=- 2061721372 2061721372 IN IP4 10.5.5.21
s=DPH-150S 01.03
c=IN IP4 10.5.5.21
t=0 0
m=audio 41000 RTP/AVP 0
a=rtpmap:0 PCMU/8000/1
a=sendrecv

<------------->
--- (13 headers 8 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 10.5.5.21:41000
Found audio description format PCMU for ID 0
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 10.5.5.21:41000
list_route: hop: <sip:555@10.5.5.21:5060>
set_destination: Parsing <sip:555@10.5.5.21:5060> for address/port to send to
set_destination: set destination to 10.5.5.21, port 5060
Transmitting (no NAT) to 10.5.5.21:5060:
ACK sip:555@10.5.5.21:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK508a0b22;rport
Max-Forwards: 70
From: "444" <sip:444@10.5.5.1>;tag=as1081e8e6
To: <sip:555@10.5.5.21:5060>;tag=001E58F79F6F_T133753141
Contact: <sip:444@10.5.5.1>
Call-ID: 3cc3e3ce649a790246af93a30da6df71@10.5.5.1
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.1.1
Content-Length: 0


---
set_destination: Parsing <sip:555@10.5.5.21:5060> for address/port to send to
set_destination: set destination to 10.5.5.21, port 5060
Reliably Transmitting (no NAT) to 10.5.5.21:5060:
BYE sip:555@10.5.5.21:5060 SIP/2.0
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK70a62771;rport
Max-Forwards: 70
From: "444" <sip:444@10.5.5.1>;tag=as1081e8e6
To: <sip:555@10.5.5.21:5060>;tag=001E58F79F6F_T133753141
Call-ID: 3cc3e3ce649a790246af93a30da6df71@10.5.5.1
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.1.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Scheduling destruction of SIP dialog '3cc3e3ce649a790246af93a30da6df71@10.5.5.1' in 32000 ms (Method: INVITE)
LINK*CLI>
<--- SIP read from UDP://10.5.5.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.5.5.1:5060;branch=z9hG4bK70a62771;rport
From: "444" <sip:444@10.5.5.1>;tag=as1081e8e6
To: <sip:555@10.5.5.21>;tag=001E58F79F6F_T133753141
Call-ID: 3cc3e3ce649a790246af93a30da6df71@10.5.5.1
CSeq: 104 BYE
User-Agent: DPH-150S 01.03
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '3cc3e3ce649a790246af93a30da6df71@10.5.5.1' Method: INVITE
LINK*CLI>
Disconnected from Asterisk server
LINK:~ # /opt/asterisk/sbin/asterisk -r
Asterisk 1.6.1.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.1.1 currently running on LINK (pid = 17636)
LINK*CLI>
Disconnected from Asterisk server
LINK:~ # /opt/asterisk/sbin/asterisk -r
Asterisk 1.6.1.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.6.1.1 currently running on LINK (pid = 17636)
LINK*CLI> sip set debug on
SIP Debugging re-enabled
LINK*CLI>
<--- SIP read from UDP://10.5.5.20:5060 --->
CANCEL sip:555@ SIP/2.0
Via: SIP/2.0/UDP 10.5.5.20:5060;branch=z9hG4bK_001E58F79F9F_T5C40A607
From: "LEO" <sip:444@>;tag=001E58F79F9F_T670356501
To: <sip:555@>
Call-ID: CALL_ID5_001E58F79F9F_T1665704629@10.5.5.20
CSeq: 2 CANCEL
User-Agent: DPH-150S 01.03
Contact: <sip:444@10.5.5.20:5060>
Max-Forwards: 70
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 10.5.5.20 : 5060 (no NAT)

<--- Reliably Transmitting (no NAT) to 10.5.5.20:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.5.5.20:5060;branch=z9hG4bK_001E58F79F9F_T5C40A607;received=10.5.5.20
From: "LEO" <sip:444@>;tag=001E58F79F9F_T670356501
To: <sip:555@>;tag=as414780bc
Call-ID: CALL_ID5_001E58F79F9F_T1665704629@10.5.5.20
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (no NAT) to 10.5.5.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.5.5.20:5060;branch=z9hG4bK_001E58F79F9F_T5C40A607;received=10.5.5.20
From: "LEO" <sip:444@>;tag=001E58F79F9F_T670356501
To: <sip:555@>;tag=as414780bc
Call-ID: CALL_ID5_001E58F79F9F_T1665704629@10.5.5.20
CSeq: 2 CANCEL
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '376f43e02cd9fa4f1455deab7aa356cd@10.5.5.1' in 32000 ms (Method: INVITE)
LINK*CLI>
<--- SIP read from UDP://10.5.5.20:5060 --->
ACK sip:555@ SIP/2.0
Via: SIP/2.0/UDP 10.5.5.20:5060;branch=z9hG4bK_001E58F79F9F_T5C40A607
From: "LEO" <sip:444@>;tag=001E58F79F9F_T670356501
To: <sip:555@>;tag=as414780bc
Call-ID: CALL_ID5_001E58F79F9F_T1665704629@10.5.5.20
CSeq: 2 ACK
User-Agent: DPH-150S 01.03
Contact: <sip:444@10.5.5.20:5060>
Max-Forwards: 70
Content-Length: 0


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

прошу прощения что выкладываю, просто пока не могу найти причину .....

Еще кое-что, если вместо вышеописанного аппарата я звоню на софтфон, то звонки завершаются нормально .... значит дел все-таки в настройках самого аппарата?

Incoming No Answer Timer : sec. [0 - 600] 0: Disable

в настройках аппарата.....это так кому интересно , вобщем заработало, но решение крайне не красивое!.....

а прошивка на аппаратах последняя? 1.06? попробуйте аппарат в ноль сбросить и заново настроить... есть 40 таких телефонов и работают...тфу...тфу...тфу (через левое плечо ;-D)