звонок на CC (астериск) приходит. Разговор идет все отлично, но если оператор CC ставит на удержание, то соединение рвется через 15 сек. администратор СС говорит, что bye шлет наша АТС.
<--- SIP read from UDP:213.79.126.139:35060 --->
INVITE sip:401@46.48.122.20 SIP/2.0
Via: SIP/2.0/UDP 213.79.126.139:35060;branch=z9hG4bK00007458;rport
Max-Forwards: 70
To: sip:401@46.48.122.20
From: sip:interhomein@46.48.122.20;tag=13741
Call-ID: 00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139
CSeq: 1 INVITE
Contact: sip:interhomein@213.79.126.139:35060
Supported: timer,100rel
Session-Expires: 180
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,NOTIFY,UPDATE
Content-Type: application/sdp
User-Agent: Panasonic-MPR07-V5.0203/VSIPGW-V2.3002
Content-Length: 270
v=0
o=- 1 1 IN IP4 213.79.126.140
s=-
c=IN IP4 213.79.126.140
t=0 0
m=audio 12062 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:12063
<------------->
--- (14 headers 14 lines) ---
Sending to 213.79.126.139:35060 (NAT)
Using INVITE request as basis request - 00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139
Found peer 'interhomein' for 'interhomein' from 213.79.126.139:35060
<--- Reliably Transmitting (NAT) to 213.79.126.139:35060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 213.79.126.139:35060;branch=z9hG4bK00007458;received=213.79.126.139;rport=35060
From: sip:interhomein@46.48.122.20;tag=13741
To: sip:401@46.48.122.20;tag=as198a83a3
Call-ID: 00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.8.1-1digium1~squeeze
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d418d36"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:213.79.126.139:35060 --->
ACK sip:401@46.48.122.20 SIP/2.0
Via: SIP/2.0/UDP 213.79.126.139:35060;branch=z9hG4bK00007458;rport
Max-Forwards: 70
To: sip:401@46.48.122.20;tag=as198a83a3
From: sip:interhomein@46.48.122.20;tag=13741
Call-ID: 00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:213.79.126.139:35060 --->
INVITE sip:401@46.48.122.20 SIP/2.0
Via: SIP/2.0/UDP 213.79.126.139:35060;branch=z9hG4bK000077b1;rport
Max-Forwards: 70
To: sip:401@46.48.122.20
From: sip:interhomein@46.48.122.20;tag=13741
Call-ID: 00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139
CSeq: 2 INVITE
Contact: sip:interhomein@213.79.126.139:35060
Supported: timer,100rel
Authorization: Digest realm="asterisk", nonce="5d418d36", algorithm=MD5, uri="sip:401@46.48.122.20", username="interhomein", response="491ad560df7537df95e353b80e4cb042"
Session-Expires: 180
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,NOTIFY,UPDATE
Content-Type: application/sdp
User-Agent: Panasonic-MPR07-V5.0203/VSIPGW-V2.3002
Content-Length: 270
v=0
o=- 1 1 IN IP4 213.79.126.140
s=-
c=IN IP4 213.79.126.140
t=0 0
m=audio 12062 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:12063
<------------->
--- (15 headers 14 lines) ---
Sending to 213.79.126.139:35060 (NAT)
Using INVITE request as basis request - 00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139
Found peer 'interhomein' for 'interhomein' from 213.79.126.139:35060
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 - 0x10c (ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 213.79.126.140:12062
Looking for 401 in interhomein (domain 46.48.122.20)
list_route: hop: <sip:interhomein@213.79.126.139:35060>
<--- Transmitting (NAT) to 213.79.126.139:35060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.79.126.139:35060;branch=z9hG4bK000077b1;received=213.79.126.139;rport=35060
From: sip:interhomein@46.48.122.20;tag=13741
To: sip:401@46.48.122.20
Call-ID: 00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.8.1-1digium1~squeeze
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 180;refresher=uas
Contact: <sip:401@46.48.122.20:5060>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 213.79.126.139:35060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 213.79.126.139:35060;branch=z9hG4bK000077b1;received=213.79.126.139;rport=35060
From: sip:interhomein@46.48.122.20;tag=13741
To: sip:401@46.48.122.20;tag=as71cb6999
Call-ID: 00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.8.1-1digium1~squeeze
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 180;refresher=uas
Contact: <sip:401@46.48.122.20:5060>
Content-Length: 0
<------------>
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 213.79.126.139:35060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.79.126.139:35060;branch=z9hG4bK000077b1;received=213.79.126.139;rport=35060
From: sip:interhomein@46.48.122.20;tag=13741
To: sip:401@46.48.122.20;tag=as71cb6999
Call-ID: 00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.8.1-1digium1~squeeze
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 180;refresher=uas
Contact: <sip:401@46.48.122.20:5060>
Content-Type: application/sdp
Content-Length: 322
v=0
o=root 513162180 513162180 IN IP4 46.48.122.20
s=Asterisk PBX 1.8.8.1-1digium1~squeeze
c=IN IP4 46.48.122.20
t=0 0
m=audio 31758 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from UDP:213.79.126.139:35060 --->
ACK sip:401@46.48.122.20:5060 SIP/2.0
Via: SIP/2.0/UDP 213.79.126.139:35060;branch=z9hG4bK00005d55;rport
Max-Forwards: 70
To: sip:401@46.48.122.20;tag=as71cb6999
From: sip:interhomein@46.48.122.20;tag=13741
Call-ID: 00004cd5-1219b65e33f31000842f0080f0bd096a@213.79.126.139
CSeq: 2 ACK
Authorization: Digest realm="asterisk", nonce="5d418d36", algorithm=MD5, uri="sip:401@46.48.122.20:5060", username="interhomein", response="18ea952d644fe01c0c5c65b21e30f13b"
ontent-Length: 0
<--- SIP read from UDP:213.79.126.139:35060 --->
BYE sip:401@46.48.122.20:5060 SIP/2.0
Via: SIP/2.0/UDP 213.79.126.139:35060;branch=z9hG4bK00005566;rport
Max-Forwards: 70
To: sip:401@46.48.122.20;tag=as36880473
From: sip:interhomein@46.48.122.20;tag=18945
Call-ID: 00007c29-1219b65e7fd61000842d0080f0bd096a@213.79.126.139
CSeq: 3 BYE
Authorization: Digest realm="asterisk", nonce="5d5607bf", algorithm=MD5, uri="sip:401@46.48.122.20:5060", username="interhomein", response="49acc00cd6cf35123b6cce63a6915728"
llow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,NOTIFY,UPDATE
User-Agent: Panasonic-MPR07-V5.0203/VSIPGW-V2.3002
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 213.79.126.139:35060 (NAT)
Scheduling destruction of SIP dialog '00007c29-1219b65e7fd61000842d0080f0bd096a@213.79.126.139' in 32000 ms (Method: BYE)
<--- Transmitting (NAT) to 213.79.126.139:35060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 213.79.126.139:35060;branch=z9hG4bK00005566;received=213.79.126.139;rport=35060
From: sip:interhomein@46.48.122.20;tag=18945
To: sip:401@46.48.122.20;tag=as36880473
Call-ID: 00007c29-1219b65e7fd61000842d0080f0bd096a@213.79.126.139
CSeq: 3 BYE
Server: Asterisk PBX 1.8.8.1-1digium1~squeeze
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
помогите разобраться с логом. в чем может быть проблема?