> Дамп снимай на стороне астера и сюда выкладывай.дамп с астера
13:07:24.837585 IP (tos 0x68, ttl 250, id 13425, offset 0, flags [none], proto UDP (17), length 891)
1.2.3.35.5060 > 1.2.3.1.3305: SIP, length: 863
INVITE sip:8011234567@asterisk.voip:3305 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.35:5060;branch=z9hG4bK-d352aff9
From: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
To: <sip:8011234567@asterisk.voip>
Remote-Party-ID: 202 <sip:202@asterisk.voip:3305>;screen=yes;party=calling
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 202 <sip:202@1.2.3.35:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 271
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 9026627 9026627 IN IP4 1.2.3.35
s=-
c=IN IP4 1.2.3.35
t=0 0
m=audio 16480 RTP/AVP 8 0 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
13:07:24.837996 IP (tos 0x0, ttl 64, id 27322, offset 0, flags [none], proto UDP (17), length 522, bad cksum 0 (->602)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 494
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 1.2.3.35:5060;branch=z9hG4bK-d352aff9;received=1.2.3.35
From: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
To: <sip:8011234567@asterisk.voip>;tag=as33f2bba5
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 101 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3b35f668"
Content-Length: 0
13:07:24.847886 IP (tos 0x68, ttl 250, id 13436, offset 0, flags [none], proto UDP (17), length 412)
1.2.3.35.5060 > 1.2.3.1.3305: SIP, length: 384
ACK sip:8011234567@asterisk.voip:3305 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.35:5060;branch=z9hG4bK-d352aff9
From: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
To: <sip:8011234567@asterisk.voip>;tag=as33f2bba5
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 101 ACK
Max-Forwards: 70
Contact: 202 <sip:202@1.2.3.35:5060>
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
13:07:24.851679 IP (tos 0x68, ttl 250, id 13442, offset 0, flags [none], proto UDP (17), length 1061)
1.2.3.35.5060 > 1.2.3.1.3305: SIP, length: 1033
INVITE sip:8011234567@asterisk.voip:3305 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.35:5060;branch=z9hG4bK-4b25d23d
From: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
To: <sip:8011234567@asterisk.voip>
Remote-Party-ID: 202 <sip:202@asterisk.voip:3305>;screen=yes;party=calling
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="202",realm="asterisk",nonce="3b35f668",uri="sip:8011234567@asterisk.voip:3305",algorithm=MD5,response="f5e853e121c61d3c3c1609f12984dcb6"
Contact: 202 <sip:202@1.2.3.35:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 271
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 9026627 9026627 IN IP4 1.2.3.35
s=-
c=IN IP4 1.2.3.35
t=0 0
m=audio 16480 RTP/AVP 8 0 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
13:07:24.852907 IP (tos 0x0, ttl 64, id 27359, offset 0, flags [none], proto UDP (17), length 465, bad cksum 0 (->616)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 437
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.2.3.35:5060;branch=z9hG4bK-4b25d23d;received=1.2.3.35
From: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
To: <sip:8011234567@asterisk.voip>
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8011234567@1.2.3.1:3305>
Content-Length: 0
13:07:25.397240 IP (tos 0x0, ttl 64, id 27440, offset 0, flags [none], proto UDP (17), length 481, bad cksum 0 (->5b5)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 453
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 1.2.3.35:5060;branch=z9hG4bK-4b25d23d;received=1.2.3.35
From: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
To: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8011234567@1.2.3.1:3305>
Content-Length: 0
13:07:25.397514 IP (tos 0x0, ttl 64, id 27443, offset 0, flags [none], proto UDP (17), length 764, bad cksum 0 (->497)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 736
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 1.2.3.35:5060;branch=z9hG4bK-4b25d23d;received=1.2.3.35
From: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
To: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8011234567@1.2.3.1:3305>
Content-Type: application/sdp
Content-Length: 241
v=0
o=qqq 1721809561 1721809561 IN IP4 1.2.3.1
s=Asterisk PBX
c=IN IP4 1.2.3.1
t=0 0
m=audio 12484 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
13:07:27.452921 IP (tos 0x0, ttl 64, id 27929, offset 0, flags [none], proto UDP (17), length 750, bad cksum 0 (->2bf)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 722
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.2.3.35:5060;branch=z9hG4bK-4b25d23d;received=1.2.3.35
From: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
To: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:8011234567@1.2.3.1:3305>
Content-Type: application/sdp
Content-Length: 241
v=0
o=qqq 1721809561 1721809562 IN IP4 1.2.3.1
s=Asterisk PBX
c=IN IP4 1.2.3.1
t=0 0
m=audio 12484 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
13:07:27.470156 IP (tos 0x68, ttl 250, id 13586, offset 0, flags [none], proto UDP (17), length 576)
1.2.3.35.5060 > 1.2.3.1.3305: SIP, length: 548
ACK sip:8011234567@1.2.3.1:3305 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.35:5060;branch=z9hG4bK-5db67ab0
From: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
To: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="202",realm="asterisk",nonce="3b35f668",uri="sip:8011234567@asterisk.voip:3305",algorithm=MD5,response="f5e853e121c61d3c3c1609f12984dcb6"
Contact: 202 <sip:202@1.2.3.35:5060>
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
13:07:38.241351 IP (tos 0x0, ttl 64, id 30153, offset 0, flags [none], proto UDP (17), length 784, bad cksum 0 (->f9ec)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 756
INVITE sip:202@1.2.3.35:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.1:3305;branch=z9hG4bK0a055c3a
Max-Forwards: 70
From: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
To: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
Contact: <sip:8011234567@1.2.3.1:3305>
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 251
v=0
o=qqq 1721809561 1721809563 IN IP4 1.2.3.1
s=Asterisk PBX
c=IN IP4 1.2.3.1
t=0 0
m=image 4931 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:2400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:397
a=T38FaxUdpEC:t38UDPRedundancy
13:07:38.269585 IP (tos 0x68, ttl 250, id 14161, offset 0, flags [none], proto UDP (17), length 652)
1.2.3.35.5060 > 1.2.3.1.3305: SIP, length: 624
SIP/2.0 200 OK
To: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
From: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 102 INVITE
Via: SIP/2.0/UDP 1.2.3.1:3305;branch=z9hG4bK0a055c3a
Contact: 202 <sip:202@1.2.3.35:5060>
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 259
Content-Type: application/sdp
v=0
o=- 9027968 9027968 IN IP4 1.2.3.35
s=-
c=IN IP4 1.2.3.35
t=0 0
m=image 16480 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy
13:07:38.270168 IP (tos 0x0, ttl 64, id 30219, offset 0, flags [none], proto UDP (17), length 383, bad cksum 0 (->fb3b)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 355
ACK sip:202@1.2.3.35:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.1:3305;branch=z9hG4bK067dcb35
Max-Forwards: 70
From: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
To: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
Contact: <sip:8011234567@1.2.3.1:3305>
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
13:08:01.467417 IP (tos 0x0, ttl 64, id 32830, offset 0, flags [none], proto UDP (17), length 534, bad cksum 0 (->f071)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 506
OPTIONS sip:202@1.2.3.35:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.1:3305;branch=z9hG4bK02711726
Max-Forwards: 70
From: "Unknown" <sip:Unknown@1.2.3.1:3305>;tag=as4aa00396
To: <sip:202@1.2.3.35:5060>
Contact: <sip:Unknown@1.2.3.1:3305>
Call-ID: 36cb4cd07d86a4375c6d38524518262c@1.2.3.1:3305
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Tue, 11 Jun 2013 09:08:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
13:08:01.479942 IP (tos 0x68, ttl 250, id 15369, offset 0, flags [none], proto UDP (17), length 438)
1.2.3.35.5060 > 1.2.3.1.3305: SIP, length: 410
SIP/2.0 200 OK
To: <sip:202@1.2.3.35:5060>;tag=f218ae177dff0430i0
From: "Unknown" <sip:Unknown@1.2.3.1:3305>;tag=as4aa00396
Call-ID: 36cb4cd07d86a4375c6d38524518262c@1.2.3.1:3305
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 1.2.3.1:3305;branch=z9hG4bK02711726
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
13:08:04.942831 IP (tos 0x0, ttl 64, id 33226, offset 0, flags [none], proto UDP (17), length 774, bad cksum 0 (->edf5)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 746
INVITE sip:202@1.2.3.35:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.1:3305;branch=z9hG4bK10245c09
Max-Forwards: 70
From: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
To: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
Contact: <sip:8011234567@1.2.3.1:3305>
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241
v=0
o=qqq 1721809561 1721809564 IN IP4 1.2.3.1
s=Asterisk PBX
c=IN IP4 1.2.3.1
t=0 0
m=audio 12484 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
13:08:04.961484 IP (tos 0x68, ttl 250, id 15566, offset 0, flags [none], proto UDP (17), length 640)
1.2.3.35.5060 > 1.2.3.1.3305: SIP, length: 612
SIP/2.0 200 OK
To: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
From: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 103 INVITE
Via: SIP/2.0/UDP 1.2.3.1:3305;branch=z9hG4bK10245c09
Contact: 202 <sip:202@1.2.3.35:5060>
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 247
Content-Type: application/sdp
v=0
o=- 9030639 9030639 IN IP4 1.2.3.35
s=-
c=IN IP4 1.2.3.35
t=0 0
m=audio 16480 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
13:08:04.962060 IP (tos 0x0, ttl 64, id 33265, offset 0, flags [none], proto UDP (17), length 383, bad cksum 0 (->ef55)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 355
ACK sip:202@1.2.3.35:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.1:3305;branch=z9hG4bK7956b801
Max-Forwards: 70
From: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
To: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
Contact: <sip:8011234567@1.2.3.1:3305>
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
13:08:06.535401 IP (tos 0x0, ttl 64, id 33473, offset 0, flags [none], proto UDP (17), length 581, bad cksum 0 (->edbf)!)
1.2.3.1.3305 > 1.2.3.35.5060: SIP, length: 553
BYE sip:202@1.2.3.35:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.1:3305;branch=z9hG4bK58a5a794
Max-Forwards: 70
From: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
To: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 104 BYE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="202", realm="asterisk", algorithm=MD5, uri="sip:asterisk.voip", nonce="3b35f668", response="8bd5f129e907e7445e57bdd7681c7f1c"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
13:08:06.546135 IP (tos 0x68, ttl 250, id 15582, offset 0, flags [none], proto UDP (17), length 319)
1.2.3.35.5060 > 1.2.3.1.3305: SIP, length: 291
SIP/2.0 200 OK
To: 202 <sip:202@asterisk.voip>;tag=eba5124dbff9b8d9o0
From: <sip:8011234567@asterisk.voip>;tag=as1552c6e9
Call-ID: a6c2165d-609a0409@1.2.3.35
CSeq: 104 BYE
Via: SIP/2.0/UDP 1.2.3.1:3305;branch=z9hG4bK58a5a794
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0