Asterisk и Panasonic TDE

[Astronomik]

Пробовал настроить подключение астериска к панасонику по sip. Звонки ходят только с панасоника на астериск. В обратную не идут.
вот что выдает дебаг

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:192.168.1.13:62700 --->



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

<--- SIP read from UDP:192.168.1.10:21482 --->



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

<--- SIP read from UDP:192.168.1.10:21482 --->
INVITE sip:4401@192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:21482;branch=z9hG4bK-d8754z-962d75e3b1a58d20-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:4700@192.168.1.10:21482>
To: <sip:4401@192.168.1.1>
From: "Lexa"<sip:4700@192.168.1.1>;tag=612cfb5c
Call-ID: OTAzNzk3NTVkYTA0YmUzZjE5Y2IwZWMyMDI4M2JmZjU.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 406

v=0
o=- 12944217441319096 1 IN IP4 192.168.1.10
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.10
t=0 0
a=ice-ufrag:1978f5
a=ice-pwd:e7c1cdc598999218392b97df7b5bdbb1
m=audio 57956 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.10 57956 typ host
a=candidate:1 2 UDP 659134 192.168.1.10 57957 typ host

<------------->
--- (13 headers 14 lines) ---
Sending to 192.168.1.10 : 21482 (no NAT)
Using INVITE request as basis request - OTAzNzk3NTVkYTA0YmUzZjE5Y2IwZWMyMDI4M2JmZjU.
Found peer '4700' for '4700' from 192.168.1.10:21482

<--- Reliably Transmitting (no NAT) to 192.168.1.10:21482 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.10:21482;branch=z9hG4bK-d8754z-962d75e3b1a58d20-1---d8754z-;received=192.168.1.10;rport=21482
From: "Lexa"<sip:4700@192.168.1.1>;tag=612cfb5c
To: <sip:4401@192.168.1.1>;tag=as44a729b8
Call-ID: OTAzNzk3NTVkYTA0YmUzZjE5Y2IwZWMyMDI4M2JmZjU.
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4c1811f2"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'OTAzNzk3NTVkYTA0YmUzZjE5Y2IwZWMyMDI4M2JmZjU.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.10:21482 --->
ACK sip:4401@192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:21482;branch=z9hG4bK-d8754z-962d75e3b1a58d20-1---d8754z-;rport
Max-Forwards: 70
To: <sip:4401@192.168.1.1>;tag=as44a729b8
From: "Lexa"<sip:4700@192.168.1.1>;tag=612cfb5c
Call-ID: OTAzNzk3NTVkYTA0YmUzZjE5Y2IwZWMyMDI4M2JmZjU.
CSeq: 1 ACK
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.10:21482 --->
INVITE sip:4401@192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:21482;branch=z9hG4bK-d8754z-d9ba4f314d15c442-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:4700@192.168.1.10:21482>
To: <sip:4401@192.168.1.1>
From: "Lexa"<sip:4700@192.168.1.1>;tag=612cfb5c
Call-ID: OTAzNzk3NTVkYTA0YmUzZjE5Y2IwZWMyMDI4M2JmZjU.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.0 stamp 58832
Authorization: Digest username="4700",realm="asterisk",nonce="4c1811f2",uri="sip:4401@192.168.1.1",response="1a54ff8988179f4ba829f499e0cd5d59",algorithm=MD5
Content-Length: 406

v=0
o=- 12944217441319096 1 IN IP4 192.168.1.10
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.1.10
t=0 0
a=ice-ufrag:1978f5
a=ice-pwd:e7c1cdc598999218392b97df7b5bdbb1
m=audio 57956 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.10 57956 typ host
a=candidate:1 2 UDP 659134 192.168.1.10 57957 typ host

<------------->
--- (14 headers 14 lines) ---
Sending to 192.168.1.10 : 21482 (no NAT)
Using INVITE request as basis request - OTAzNzk3NTVkYTA0YmUzZjE5Y2IwZWMyMDI4M2JmZjU.
Found peer '4700' for '4700' from 192.168.1.10:21482
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.10:57956
Looking for 4401 in DLPN_test (domain 192.168.1.1)
list_route: hop: <sip:4700@192.168.1.10:21482>

<--- Transmitting (no NAT) to 192.168.1.10:21482 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:21482;branch=z9hG4bK-d8754z-d9ba4f314d15c442-1---d8754z-;received=192.168.1.10;rport=21482
From: "Lexa"<sip:4700@192.168.1.1>;tag=612cfb5c
To: <sip:4401@192.168.1.1>
Call-ID: OTAzNzk3NTVkYTA0YmUzZjE5Y2IwZWMyMDI4M2JmZjU.
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:4401@192.168.1.1>
Content-Length: 0


<------------>
Audio is at 192.168.4.1 port 13712
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.4.2:5060:
INVITE sip:4401@192.168.4.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.1:5060;branch=z9hG4bK09df928c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.4.1>;tag=as49b4aea0
To: <sip:4401@192.168.4.2>
Contact: <sip:asterisk@192.168.4.1>
Call-ID: 3310cbc954289ce637a92d0d4ab1eb35@192.168.4.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 10 Mar 2011 07:57:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 106643526 106643526 IN IP4 192.168.4.1
s=Asterisk PBX 1.6.2.9-2+squeeze1
c=IN IP4 192.168.4.1
t=0 0
m=audio 13712 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #1 (no NAT) to 192.168.4.2:5060:
INVITE sip:4401@192.168.4.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.1:5060;branch=z9hG4bK09df928c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.4.1>;tag=as49b4aea0
To: <sip:4401@192.168.4.2>
Contact: <sip:asterisk@192.168.4.1>
Call-ID: 3310cbc954289ce637a92d0d4ab1eb35@192.168.4.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 10 Mar 2011 07:57:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 106643526 106643526 IN IP4 192.168.4.1
s=Asterisk PBX 1.6.2.9-2+squeeze1
c=IN IP4 192.168.4.1
t=0 0
m=audio 13712 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #2 (no NAT) to 192.168.4.2:5060:
INVITE sip:4401@192.168.4.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.1:5060;branch=z9hG4bK09df928c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.4.1>;tag=as49b4aea0
To: <sip:4401@192.168.4.2>
Contact: <sip:asterisk@192.168.4.1>
Call-ID: 3310cbc954289ce637a92d0d4ab1eb35@192.168.4.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 10 Mar 2011 07:57:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 106643526 106643526 IN IP4 192.168.4.1
s=Asterisk PBX 1.6.2.9-2+squeeze1
c=IN IP4 192.168.4.1
t=0 0
m=audio 13712 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #3 (no NAT) to 192.168.4.2:5060:
INVITE sip:4401@192.168.4.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.1:5060;branch=z9hG4bK09df928c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.4.1>;tag=as49b4aea0
To: <sip:4401@192.168.4.2>
Contact: <sip:asterisk@192.168.4.1>
Call-ID: 3310cbc954289ce637a92d0d4ab1eb35@192.168.4.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 10 Mar 2011 07:57:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 106643526 106643526 IN IP4 192.168.4.1
s=Asterisk PBX 1.6.2.9-2+squeeze1
c=IN IP4 192.168.4.1
t=0 0
m=audio 13712 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Retransmitting #4 (no NAT) to 192.168.4.2:5060:
INVITE sip:4401@192.168.4.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.1:5060;branch=z9hG4bK09df928c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.4.1>;tag=as49b4aea0
To: <sip:4401@192.168.4.2>
Contact: <sip:asterisk@192.168.4.1>
Call-ID: 3310cbc954289ce637a92d0d4ab1eb35@192.168.4.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 10 Mar 2011 07:57:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 106643526 106643526 IN IP4 192.168.4.1
s=Asterisk PBX 1.6.2.9-2+squeeze1
c=IN IP4 192.168.4.1
t=0 0
m=audio 13712 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.13:62700 --->



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

<--- SIP read from UDP:192.168.1.10:21482 --->



<------------->
Retransmitting #5 (no NAT) to 192.168.4.2:5060:
INVITE sip:4401@192.168.4.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.1:5060;branch=z9hG4bK09df928c;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.4.1>;tag=as49b4aea0
To: <sip:4401@192.168.4.2>
Contact: <sip:asterisk@192.168.4.1>
Call-ID: 3310cbc954289ce637a92d0d4ab1eb35@192.168.4.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 10 Mar 2011 07:57:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 106643526 106643526 IN IP4 192.168.4.1
s=Asterisk PBX 1.6.2.9-2+squeeze1
c=IN IP4 192.168.4.1
t=0 0
m=audio 13712 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
Nastja*CLI> sip set debug off
SIP Debugging Disabled
Nastja*CLI> sip set debug on
SIP Debugging enabled

<--- SIP read from UDP:192.168.4.2:35060 --->
INVITE sip:4700@192.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:35060;branch=z9hG4bK00005633;rport
Max-Forwards: 70
To: sip:4700@192.168.4.1
From: "ip test" <sip:tde@192.168.4.1>;tag=13915
Call-ID: 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
CSeq: 1 INVITE
Contact: sip:tde@192.168.4.2:35060
Supported: timer,100rel
Session-Expires: 180
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,UPDATE
Content-Type: application/sdp
User-Agent: Panasonic-MPR07-VSIPGW/V3.0007
Content-Length: 264

v=0
o=- 1 1 IN IP4 192.168.4.3
s=-
c=IN IP4 192.168.4.3
t=0 0
m=audio 12154 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=rtcp:12155

<------------->
--- (14 headers 14 lines) ---
Sending to 192.168.4.2 : 35060 (no NAT)
Using INVITE request as basis request - 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
Found peer 'tde' for 'tde' from 192.168.4.2:35060

<--- Reliably Transmitting (no NAT) to 192.168.4.2:35060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.4.2:35060;branch=z9hG4bK00005633;received=192.168.4.2;rport=35060
From: "ip test" <sip:tde@192.168.4.1>;tag=13915
To: sip:4700@192.168.4.1;tag=as0adef143
Call-ID: 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37908468"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.4.2:35060 --->
ACK sip:4700@192.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:35060;branch=z9hG4bK00005633;rport
Max-Forwards: 70
To: sip:4700@192.168.4.1;tag=as0adef143
From: "ip test" <sip:tde@192.168.4.1>;tag=13915
Call-ID: 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
CSeq: 1 ACK
Content-Length: 0


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

<--- SIP read from UDP:192.168.4.2:35060 --->
INVITE sip:4700@192.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:35060;branch=z9hG4bK000040f4;rport
Max-Forwards: 70
To: sip:4700@192.168.4.1
From: "ip test" <sip:tde@192.168.4.1>;tag=13915
Call-ID: 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
CSeq: 2 INVITE
Contact: sip:tde@192.168.4.2:35060
Supported: timer,100rel
Authorization: Digest realm="asterisk", nonce="37908468", algorithm=MD5, uri="sip:4700@192.168.4.1", username="tde", response="e98aa9b44b81049dd5b43519eeb92ff5"
Session-Expires: 180
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,UPDATE
Content-Type: application/sdp
User-Agent: Panasonic-MPR07-VSIPGW/V3.0007
Content-Length: 264

v=0
o=- 1 1 IN IP4 192.168.4.3
s=-
c=IN IP4 192.168.4.3
t=0 0
m=audio 12154 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=rtcp:12155

<------------->
--- (15 headers 14 lines) ---
Sending to 192.168.4.2 : 35060 (no NAT)
Using INVITE request as basis request - 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
Found peer 'tde' for 'tde' from 192.168.4.2:35060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.4.3:12154
Looking for 4700 in DLPN_test (domain 192.168.4.1)
list_route: hop: <sip:tde@192.168.4.2:35060>

<--- Transmitting (no NAT) to 192.168.4.2:35060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.2:35060;branch=z9hG4bK000040f4;received=192.168.4.2;rport=35060
From: "ip test" <sip:tde@192.168.4.1>;tag=13915
To: sip:4700@192.168.4.1
Call-ID: 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 180;refresher=uas
Contact: <sip:4700@192.168.4.1>
Content-Length: 0


<------------>
Audio is at 192.168.1.1 port 19590
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.10
INVITE sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK78bfc1e2;rport
Max-Forwards: 70
From: "New User" <sip:tde@192.168.1.1>;tag=as36e0e832
To: <sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154>
Contact: <sip:tde@192.168.1.1>
Call-ID: 79f24d2a6a97c6bb3757e6ef249a1cd6@192.168.1.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 10 Mar 2011 08:00:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 266

v=0
o=root 838342918 838342918 IN IP4 192.168.1.1
s=Asterisk PBX 1.6.2.9-2+squeeze1
c=IN IP4 192.168.1.1
t=0 0
m=audio 19590 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Mar 10 10:00:07] WARNING[1563]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)

<--- SIP read from UDP:192.168.1.10:21482 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK78bfc1e2;rport=5060
Contact: <sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154>
To: <sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154>;tag=1f6f1b4b
From: "New User"<sip:tde@192.168.1.1>;tag=as36e0e832
Call-ID: 79f24d2a6a97c6bb3757e6ef249a1cd6@192.168.1.1
CSeq: 102 INVITE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


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

<--- Transmitting (no NAT) to 192.168.4.2:35060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.4.2:35060;branch=z9hG4bK000040f4;received=192.168.4.2;rport=35060
From: "ip test" <sip:tde@192.168.4.1>;tag=13915
To: sip:4700@192.168.4.1;tag=as13c83f5c
Call-ID: 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 180;refresher=uas
Contact: <sip:4700@192.168.4.1>
Content-Length: 0


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

<--- SIP read from UDP:192.168.4.2:35060 --->
CANCEL sip:4700@192.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:35060;branch=z9hG4bK000040f4;rport
Max-Forwards: 70
To: sip:4700@192.168.4.1
From: "ip test" <sip:tde@192.168.4.1>;tag=13915
Call-ID: 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
CSeq: 2 CANCEL
Authorization: Digest realm="asterisk", nonce="37908468", algorithm=MD5, uri="sip:4700@192.168.4.1", username="tde", response="3dbceb16e623f90f519704094ccd016d"
User-Agent: Panasonic-MPR07-VSIPGW/V3.0007
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.4.2 : 35060 (no NAT)

<--- Reliably Transmitting (no NAT) to 192.168.4.2:35060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.4.2:35060;branch=z9hG4bK000040f4;received=192.168.4.2;rport=35060
From: "ip test" <sip:tde@192.168.4.1>;tag=13915
To: sip:4700@192.168.4.1;tag=as13c83f5c
Call-ID: 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (no NAT) to 192.168.4.2:35060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.2:35060;branch=z9hG4bK000040f4;received=192.168.4.2;rport=35060
From: "ip test" <sip:tde@192.168.4.1>;tag=13915
To: sip:4700@192.168.4.1;tag=as13c83f5c
Call-ID: 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
CSeq: 2 CANCEL
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '79f24d2a6a97c6bb3757e6ef249a1cd6@192.168.1.1' in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.1.10
CANCEL sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK78bfc1e2;rport
Max-Forwards: 70
From: "New User" <sip:tde@192.168.1.1>;tag=as36e0e832
To: <sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154>
Call-ID: 79f24d2a6a97c6bb3757e6ef249a1cd6@192.168.1.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


---
Scheduling destruction of SIP dialog '79f24d2a6a97c6bb3757e6ef249a1cd6@192.168.1.1' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.4.2:35060 --->
ACK sip:4700@192.168.4.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.4.2:35060;branch=z9hG4bK000040f4;rport
Max-Forwards: 70
To: sip:4700@192.168.4.1;tag=as13c83f5c
From: "ip test" <sip:tde@192.168.4.1>;tag=13915
Call-ID: 00006de6-1f2a63de00d2100097cb0080f0b80fe8@192.168.4.2
CSeq: 2 ACK
Authorization: Digest realm="asterisk", nonce="37908468", algorithm=MD5, uri="sip:4700@192.168.4.1", username="tde", response="20a13c70e080374fcf77ece4e99f9b94"
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.10:21482 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK78bfc1e2;rport=5060
Contact: <sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154>
To: <sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154>;tag=1f6f1b4b
From: "New User"<sip:tde@192.168.1.1>;tag=as36e0e832
Call-ID: 79f24d2a6a97c6bb3757e6ef249a1cd6@192.168.1.1
CSeq: 102 CANCEL
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.10:21482 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK78bfc1e2;rport=5060
To: <sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154>;tag=1f6f1b4b
From: "New User"<sip:tde@192.168.1.1>;tag=as36e0e832
Call-ID: 79f24d2a6a97c6bb3757e6ef249a1cd6@192.168.1.1
CSeq: 102 INVITE
User-Agent: X-Lite 4 release 4.0 stamp 58832
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 192.168.1.10
ACK sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK78bfc1e2;rport
Max-Forwards: 70
From: "New User" <sip:tde@192.168.1.1>;tag=as36e0e832
To: <sip:4700@192.168.1.10:21482;rinstance=9ac68965a18c9154>;tag=1f6f1b4b
Contact: <sip:tde@192.168.1.1>
Call-ID: 79f24d2a6a97c6bb3757e6ef249a1cd6@192.168.1.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0


---
Really destroying SIP dialog '79f24d2a6a97c6bb3757e6ef249a1cd6@192.168.1.1' Method: INVITE
Nastja*CLI> sip set debug off
SIP Debugging Disabled

Насколько я понял астериск отдает панасонику не номер на кого набрать а какую-то хрень. кто поскажет как это исправить чтобы звонки шли в обоих направлениях?

[ded]

Астериск посылает не какую то хрень, а именно номер, который набрать (и кто ему звонит)
To: <sip:4401@192.168.1.1>
From: "Lexa"<sip:4700@192.168.1.1>

А вот хрень у Вас, уважаемый Лёха с ИП адресацией - в приведённом дебаге фигурируют пакеты 192.168.1.1, 192.168.1.10, 192.168.4.1, 192.168.4.2 - на которые при INVITE ретрансмитит (перепосылает) много раз без ответа.
Другая хрень - неудачно придуманый диал-план: 4700 номер на Астериске и 4401 - на Панасонике. Разделили бы хоть по первой цифре, что ли...
Справитесь?

[vlego]

номер конечно на панасе может существовать любой. ..если он конечно обслуживается диалпланом и нет такого же на астериске.
вот только в толк не возьму - как Вы организовали связь панас<->астериск ?

к этому вопросу это конечно не относится но все же, у Вас в логе - присутствует интересная запись ---
[Mar 10 10:00:07] WARNING[1563]: app_dial.c:1747 dial_exec_full: Unable to create channel of type 'IAX2' (cause 20 - Unknown)

[Astronomik]

А вот хрень у Вас, уважаемый Лёха с ИП адресацией - в приведённом дебаге фигурируют пакеты 192.168.1.1, 192.168.1.10, 192.168.4.1, 192.168.4.2 - на которые при INVITE ретрансмитит (перепосылает) много раз без ответа.
Другая хрень - неудачно придуманый диал-план: 4700 номер на Астериске и 4401 - на Панасонике. Разделили бы хоть по первой цифре, что ли...
Справитесь?

На сервере 4 интерфейса. Панасоник в своей подсети (192.168.4.0/24). Астерикс на маршутизаторе(подсеть 192.168.10/24). Ната нету. В чем проблема с адресацией? В том куске что вы вырезали идет обращение от телефона к астериску. Далее номер посылается на панасоник
From: "asterisk" <sip:asterisk@192.168.4.1>;tag=as49b4aea0
To: <sip:4401@192.168.4.2>
Contact: <sip:asterisk@192.168.4.1>
Подскажите в чем может быть затык?
Неудачно придуманный диалплан не поправишь. Узловая АТС понимает только номера вида 4ХХХ. План набора на панасоник:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[CallingRule_panas4]
exten => _44XX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _40XX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _41XX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _42XX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _43XX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _45XX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _48XX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _49XX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _17XX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _15XX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _2XXXXXX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _3XXXXXX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _5XXXXXX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)
exten => _8wXXXXXXXX,1,Macro(trunkdial-panasonic,${panasonic}/${EXTEN:0},,panasonic,)

vlego
По SIP-транку. В цифровых панасониках реализовали таки нормальный SIP.
откуда запись - не знаю. Астериск из репозитория Debian squeeze. Конфиги дописывал.
Внизу что писал для транка на панасоник:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[panasonic]
transfer = yes
type = friend
host = 192.168.4.2
username = tde
secret = tde
trunkname = panasonic ; GUI metadata
context = DID_panasonic
dtmfmode = rfc2833
insecure = no
disallow = all
allow = alaw:20
callcounter = yes

[tde]
secret = tde
transer = yes
host = dynamic
context = DLPN_test
type = friend
;username = tde
callcounter = yes

[vlego]

диалплан менять и не надо.

что у вас показывает команда
sip show registry ?

и потом, Вы ручками все реализовали ?, а не через freePBX - судя по Вашим сообщенииям.

[Astronomik]

sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.

Ничего)
Руками. Изначально пробовал ставить какие-то веб-интерфейсы, потом оказалось что мне с ними разобраться сложнее чем с консолью и конфигами.

[vlego]

попробуйте так
CLI> sip show peers

должно показывать прим-но след-е

Name/username Host Dyn Nat ACL Port Status
panasonic ip address 5060 OK (4 ms)

совершить вызов на панас можно либо с экстеншена того же контекста где регился
панас либо из
extensions_custom.conf
-> [from-internal-custom]
....
exten => ВАШ МЕСТН.НОМЕР ПАНАС,1,Dial(SIP/panasonic,,${EXTEN))

с любого номера сущ-го на астериске.

однако лучше все "строить" через FreePBX.
Руками только тогда, когда есть донабор например.
Зачем избретать велосипед ?

Через FreePBX завел транк
Создал исх-й маршрут
и т.д. и т.п.
гораздо приятней.

[ded]

На пире панасоника укажите canreinvite=no

Не обязательно его регистрировать. Регистрация нужна только для динамических хостов, а там всё статично.
[from-internal-custom]
....
exten => ВАШ МЕСТН.НОМЕР ПАНАС,1,Dial(SIP/panasonic/${EXTEN)) - вот так правильней.

Ну и вследствие именно этого

На сервере 4 интерфейса. Панасоник в своей подсети (192.168.4.0/24). Астерикс на маршутизаторе(подсеть 192.168.10/24).

будут проблемы.
Долго объяснять. Вкратце: bindaddr=0.0.0.0 слушает то на всех 4-х интерфейсах, но отправляет только с одного.

Насколько я понял астериск отдает панасонику не номер на кого набрать а какую-то хрень.

Поясните свою мысль всё таки, где Вы не увидели номер который астериск отдает панасонику?

[Astronomik]

sip show peers
Name/username Host Dyn Nat ACL Port Status
4700/4700 192.168.1.10 D 21482 Unmonitored
4701/4701 192.168.1.13 D 62700 Unmonitored
4702/4702 (Unspecified) D N 0 Unmonitored
panasonic/tde 192.168.4.2 5060 Unmonitored
tde/tde 192.168.4.2 D 35060 Unmonitored
trunk_1/testprov 87.250.250.3 5060 Unmonitored
6 sip peers [Monitored: 0 online, 0 offline Unmonitored: 5 online, 1 offline]

Тут все есть и все активно даже вроде.

секции такой как и файла нету

l | grep ext
-rw-r----- 1 asterisk asterisk 2462 Сен 7 2010 extconfig.conf
-rw-r----- 1 asterisk asterisk 12801 Сен 7 2010 extensions.ael
-rw-r----- 1 asterisk asterisk 39189 Фев 9 13:43 extensions.conf
-rw-r----- 1 asterisk asterisk 5408 Сен 7 2010 extensions.lua
-rw-r----- 1 asterisk asterisk 7481 Сен 7 2010 extensions_minivm.conf

все настройки лежат в extconfig.conf, но в принципе правила написаны по указанной вами схеме...

[vlego]

ded, проблемы с интерфейсами (4шт) у него похоже будут.
можно конечно сетки в пареметре localnet указать, но это не всегда спасает.
я бы поставил в пире панаса
host=dynamic

bindaddr= можно указать в пире, вроде как не запрещено.