Не видно, чтобы зарегистрировался, не видно ответа ОК, есть только запросы без ответа
Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #1)
Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #2)
Транк до Ростелекома
Re: Транк до Ростелекома
Есть ОК, в самом конце
И sip show registry
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: LOG CLI
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:39:42] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #28)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK43830c46;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as519de5ba
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 130 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:40:02] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #29)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK48deae65;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as298424d6
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 131 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:40:22] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #30)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK7c385914;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as786d8fb6
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 132 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:40:42] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #31)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK73f5dcfd;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as385151a9
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 133 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:41:02] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #32)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK242a0bd1;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as6a33171b
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 134 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:41:22] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #33)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK79294797;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as4affbd27
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 135 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:41:43] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #34)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK3a571817;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as75e8d358
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 136 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:42:03] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #35)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK523678b0;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as2f118efe
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 137 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:42:23] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #36)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK4e7f7ce5;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as5b966f80
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 138 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
<--- SIP read from TCP:77.51.250.219:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK4e7f7ce5;received=89.179.102.101;rport=36045
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as5b966f80
To: <sip:admin@286970.lp.centertelecom.ru>;tag=VW3hh8BST0kAH4P2GYQmTvyqfj8Bqgpk
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 138 REGISTER
WWW-Authenticate: Digest realm="CallManager", nonce="hvSAsZXei6", opaque="opaqueData", algorithm=MD5, qop="auth"
Max-Forwards: 70
User-Agent: Svetets CallManager R12622
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, PUBLISH, SUBSCRIBE, OPTIONS, INFO
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name 286970.lp.centertelecom.ru
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK47452549;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as4cbe0501
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 139 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Authorization: Digest username="admin", realm="CallManager", algorithm=MD5, uri="sip:286970.lp.centertelecom.ru", nonce="hvSAsZXei6", response="9b54cda4c17668d48e1471cc861b9b80", opaque="opaqueData", qop=auth, cnonce="18ad5ed8", nc=00000001
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
<--- SIP read from TCP:77.51.250.219:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK47452549;received=89.179.102.101;rport=36045
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as4cbe0501
To: <sip:admin@286970.lp.centertelecom.ru>;tag=UMlXYYKEkk3tZL6xFeFqY5y5tTtMKjTF
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 139 REGISTER
Contact: <sip:admin@89.179.102.101;transport=TCP>;expires=120
Max-Forwards: 70
User-Agent: Svetets CallManager R12622
Date: Thu, 29 Sep 2011 09:43:39 GMT
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, SUBSCRIBE, PUBLISH, OPTIONS, INFO
Contacts-Url: <http://admin.centertelecom.ru/contactlist2/>
Application-Proxy: <77.51.250.227:12345>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, PUBLISH, SUBSCRIBE, OPTIONS, INFO
Content-Length: 0
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:39:42] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #28)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK43830c46;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as519de5ba
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 130 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:40:02] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #29)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK48deae65;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as298424d6
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 131 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:40:22] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #30)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK7c385914;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as786d8fb6
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 132 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:40:42] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #31)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK73f5dcfd;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as385151a9
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 133 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:41:02] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #32)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK242a0bd1;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as6a33171b
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 134 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:41:22] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #33)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK79294797;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as4affbd27
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 135 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:41:43] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #34)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK3a571817;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as75e8d358
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 136 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:42:03] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #35)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK523678b0;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as2f118efe
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 137 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
[Sep 29 13:42:23] NOTICE[23207]: chan_sip.c:12104 sip_reg_timeout: -- Registration for 'admin@286970.lp.centertelecom.ru' timed out, trying again (Attempt #36)
REGISTER 10 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK4e7f7ce5;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as5b966f80
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 138 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
Really destroying SIP dialog '0b9623732b54670d7f5e6f354694a38f@192.168.0.254' Method: REGISTER
<--- SIP read from TCP:77.51.250.219:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK4e7f7ce5;received=89.179.102.101;rport=36045
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as5b966f80
To: <sip:admin@286970.lp.centertelecom.ru>;tag=VW3hh8BST0kAH4P2GYQmTvyqfj8Bqgpk
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 138 REGISTER
WWW-Authenticate: Digest realm="CallManager", nonce="hvSAsZXei6", opaque="opaqueData", algorithm=MD5, qop="auth"
Max-Forwards: 70
User-Agent: Svetets CallManager R12622
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, PUBLISH, SUBSCRIBE, OPTIONS, INFO
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Responding to challenge, registration to domain/host name 286970.lp.centertelecom.ru
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 77.51.250.219:5060:
REGISTER sip:286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK47452549;rport
Max-Forwards: 70
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as4cbe0501
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 139 REGISTER
User-Agent: Asterisk PBX 1.6.2.20
Authorization: Digest username="admin", realm="CallManager", algorithm=MD5, uri="sip:286970.lp.centertelecom.ru", nonce="hvSAsZXei6", response="9b54cda4c17668d48e1471cc861b9b80", opaque="opaqueData", qop=auth, cnonce="18ad5ed8", nc=00000001
Expires: 120
Contact: <sip:admin@89.179.102.101;transport=TCP>
Content-Length: 0
---
<--- SIP read from TCP:77.51.250.219:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 89.179.102.101:5060;branch=z9hG4bK47452549;received=89.179.102.101;rport=36045
From: <sip:admin@286970.lp.centertelecom.ru>;tag=as4cbe0501
To: <sip:admin@286970.lp.centertelecom.ru>;tag=UMlXYYKEkk3tZL6xFeFqY5y5tTtMKjTF
Call-ID: 0b9623732b54670d7f5e6f354694a38f@192.168.0.254
CSeq: 139 REGISTER
Contact: <sip:admin@89.179.102.101;transport=TCP>;expires=120
Max-Forwards: 70
User-Agent: Svetets CallManager R12622
Date: Thu, 29 Sep 2011 09:43:39 GMT
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, SUBSCRIBE, PUBLISH, OPTIONS, INFO
Contacts-Url: <http://admin.centertelecom.ru/contactlist2/>
Application-Proxy: <77.51.250.227:12345>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, PUBLISH, SUBSCRIBE, OPTIONS, INFO
Content-Length: 0
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: registry
srv-cps*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
286970.lp.centertelecom.ru:506 N admin 105 Registered Thu, 29 Sep 2011 13:44:08
Host dnsmgr Username Refresh State Reg.Time
286970.lp.centertelecom.ru:506 N admin 105 Registered Thu, 29 Sep 2011 13:44:08
Re: Транк до Ростелекома
Есть подвижки 
Во время входящего звонка в CLI посыпались сообщения, но софтфон так и не звонит
В extensions.conf следующее:
Во время входящего звонка в CLI посыпались сообщения, но софтфон так и не звонитВ extensions.conf следующее:
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: extensions
[from-admin]
include=>office
exten=> s,1,Answer
exten=> s,2,Dial(SIP/111,30,Ttr)
exten=> s,3,Hangup
include=>office
exten=> s,1,Answer
exten=> s,2,Dial(SIP/111,30,Ttr)
exten=> s,3,Hangup
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER: CLI
<--- Reliably Transmitting (NAT) to 77.51.250.219:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 77.51.250.219:5060;branch=z9hG4bKuZ33ozbvQ8Oxn;received=77.51.250.219;rport=5060
From: "89036435544" <sip:89036435544@77.51.255.30:5062>;tag=MeSrxeD3
To: <sip:admin@286970.lp.centertelecom.ru>;tag=as21b1d4f4
Call-ID: M1LNqKFpT8Oy9Q9v1317290004@77.51.250.219
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Sep 29 13:52:07] NOTICE[1488]: chan_sip.c:20785 handle_request_invite: Call from 'admin' to extension 'admin' rejected because extension not found in context 'from-admin'.
Scheduling destruction of SIP dialog 'M1LNqKFpT8Oy9Q9v1317290004@77.51.250.219' in 32000 ms (Method: INVITE)
<--- SIP read from TCP:77.51.250.219:5060 --->
ACK sip:admin@286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 77.51.250.219:5060;branch=z9hG4bKuZ33ozbvQ8Oxn;rport
From: "89036435544" <sip:89036435544@77.51.255.30:5062>;tag=MeSrxeD3
To: <sip:admin@286970.lp.centertelecom.ru>;tag=as21b1d4f4
Call-ID: M1LNqKFpT8Oy9Q9v1317290004@77.51.250.219
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'M1LNqKFpT8Oy9Q9v1317290004@77.51.250.219' Method: INVITE
Really destroying SIP dialog '31ba53453c8cacca5746ea4a55123c6a@192.168.0.254' Method: REGISTER
srv-cps*CLI>
srv-cps*CLI>
srv-cps*CLI>
srv-cps*CLI>
<--- SIP read from TCP:77.51.250.219:5060 --->
INVITE sip:admin@286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 77.51.250.219:5060;branch=z9hG4bK5JuQuVgmfOG5Z;rport
From: "89036435544" <sip:89036435544@77.51.255.27:5062>;tag=QGp2bDjq
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: Qbwk98aKtV1VV01u1317290023@77.51.250.219
CSeq: 1 INVITE
Contact: <sip:89036435544@77.51.250.219:5060;transport=TCP>
Max-Forwards: 70
User-Agent: Svetets CallManager R12622
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, PUBLISH, SUBSCRIBE, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 236
v=0
o=user 89905511101 08859302 IN IP4 77.51.250.225
s=Svetets
c=IN IP4 77.51.250.225
t=0 0
m=audio 12938 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
== Using SIP RTP CoS mark 5
Sending to 77.51.250.219 : 5060 (no NAT)
Using INVITE request as basis request - Qbwk98aKtV1VV01u1317290023@77.51.250.219
Found peer 'admin' for '89036435544' from 77.51.250.219:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 77.51.250.225:12938
Looking for admin in from-admin (domain 286970.lp.centertelecom.ru)
<--- Reliably Transmitting (NAT) to 77.51.250.219:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 77.51.250.219:5060;branch=z9hG4bK5JuQuVgmfOG5Z;received=77.51.250.219;rport=5060
From: "89036435544" <sip:89036435544@77.51.255.27:5062>;tag=QGp2bDjq
To: <sip:admin@286970.lp.centertelecom.ru>;tag=as26403ba0
Call-ID: Qbwk98aKtV1VV01u1317290023@77.51.250.219
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Sep 29 13:52:26] NOTICE[1488]: chan_sip.c:20785 handle_request_invite: Call from 'admin' to extension 'admin' rejected because extension not found in context 'from-admin'.
Scheduling destruction of SIP dialog 'Qbwk98aKtV1VV01u1317290023@77.51.250.219' in 32000 ms (Method: INVITE)
<--- SIP read from TCP:77.51.250.219:5060 --->
ACK sip:admin@286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 77.51.250.219:5060;branch=z9hG4bK5JuQuVgmfOG5Z;rport
From: "89036435544" <sip:89036435544@77.51.255.27:5062>;tag=QGp2bDjq
To: <sip:admin@286970.lp.centertelecom.ru>;tag=as26403ba0
Call-ID: Qbwk98aKtV1VV01u1317290023@77.51.250.219
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'Qbwk98aKtV1VV01u1317290023@77.51.250.219' Method: INVITE
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 77.51.250.219:5060;branch=z9hG4bKuZ33ozbvQ8Oxn;received=77.51.250.219;rport=5060
From: "89036435544" <sip:89036435544@77.51.255.30:5062>;tag=MeSrxeD3
To: <sip:admin@286970.lp.centertelecom.ru>;tag=as21b1d4f4
Call-ID: M1LNqKFpT8Oy9Q9v1317290004@77.51.250.219
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Sep 29 13:52:07] NOTICE[1488]: chan_sip.c:20785 handle_request_invite: Call from 'admin' to extension 'admin' rejected because extension not found in context 'from-admin'.
Scheduling destruction of SIP dialog 'M1LNqKFpT8Oy9Q9v1317290004@77.51.250.219' in 32000 ms (Method: INVITE)
<--- SIP read from TCP:77.51.250.219:5060 --->
ACK sip:admin@286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 77.51.250.219:5060;branch=z9hG4bKuZ33ozbvQ8Oxn;rport
From: "89036435544" <sip:89036435544@77.51.255.30:5062>;tag=MeSrxeD3
To: <sip:admin@286970.lp.centertelecom.ru>;tag=as21b1d4f4
Call-ID: M1LNqKFpT8Oy9Q9v1317290004@77.51.250.219
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'M1LNqKFpT8Oy9Q9v1317290004@77.51.250.219' Method: INVITE
Really destroying SIP dialog '31ba53453c8cacca5746ea4a55123c6a@192.168.0.254' Method: REGISTER
srv-cps*CLI>
srv-cps*CLI>
srv-cps*CLI>
srv-cps*CLI>
<--- SIP read from TCP:77.51.250.219:5060 --->
INVITE sip:admin@286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 77.51.250.219:5060;branch=z9hG4bK5JuQuVgmfOG5Z;rport
From: "89036435544" <sip:89036435544@77.51.255.27:5062>;tag=QGp2bDjq
To: <sip:admin@286970.lp.centertelecom.ru>
Call-ID: Qbwk98aKtV1VV01u1317290023@77.51.250.219
CSeq: 1 INVITE
Contact: <sip:89036435544@77.51.250.219:5060;transport=TCP>
Max-Forwards: 70
User-Agent: Svetets CallManager R12622
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, PUBLISH, SUBSCRIBE, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 236
v=0
o=user 89905511101 08859302 IN IP4 77.51.250.225
s=Svetets
c=IN IP4 77.51.250.225
t=0 0
m=audio 12938 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
== Using SIP RTP CoS mark 5
Sending to 77.51.250.219 : 5060 (no NAT)
Using INVITE request as basis request - Qbwk98aKtV1VV01u1317290023@77.51.250.219
Found peer 'admin' for '89036435544' from 77.51.250.219:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc7fffff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 77.51.250.225:12938
Looking for admin in from-admin (domain 286970.lp.centertelecom.ru)
<--- Reliably Transmitting (NAT) to 77.51.250.219:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/TCP 77.51.250.219:5060;branch=z9hG4bK5JuQuVgmfOG5Z;received=77.51.250.219;rport=5060
From: "89036435544" <sip:89036435544@77.51.255.27:5062>;tag=QGp2bDjq
To: <sip:admin@286970.lp.centertelecom.ru>;tag=as26403ba0
Call-ID: Qbwk98aKtV1VV01u1317290023@77.51.250.219
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
[Sep 29 13:52:26] NOTICE[1488]: chan_sip.c:20785 handle_request_invite: Call from 'admin' to extension 'admin' rejected because extension not found in context 'from-admin'.
Scheduling destruction of SIP dialog 'Qbwk98aKtV1VV01u1317290023@77.51.250.219' in 32000 ms (Method: INVITE)
<--- SIP read from TCP:77.51.250.219:5060 --->
ACK sip:admin@286970.lp.centertelecom.ru SIP/2.0
Via: SIP/2.0/TCP 77.51.250.219:5060;branch=z9hG4bK5JuQuVgmfOG5Z;rport
From: "89036435544" <sip:89036435544@77.51.255.27:5062>;tag=QGp2bDjq
To: <sip:admin@286970.lp.centertelecom.ru>;tag=as26403ba0
Call-ID: Qbwk98aKtV1VV01u1317290023@77.51.250.219
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'Qbwk98aKtV1VV01u1317290023@77.51.250.219' Method: INVITE
Re: Транк до Ростелекома
[Sep 29 14:06:56] NOTICE[7044]: chan_sip.c:20785 handle_request_invite: Call from 'admin' to extension 'admin' rejected because extension not found in context 'admin'.
Scheduling destruction of SIP dialog 'ROG3Dc9QZrizWyoA1317290892@77.51.250.219' in 32000 ms (Method: INVITE)
Т.е. ругается на то, что отсутствует экстеншн, хотя в extensions.conf он есть (((( почему так ?
Scheduling destruction of SIP dialog 'ROG3Dc9QZrizWyoA1317290892@77.51.250.219' in 32000 ms (Method: INVITE)
Т.е. ругается на то, что отсутствует экстеншн, хотя в extensions.conf он есть (((( почему так ?
Re: Транк до Ростелекома
У вас контекст [from-admin], а звонки приезжают в контекст [admin]
Re: Транк до Ростелекома
exten=> admin,1,Answer
exten=> admin,2,Dial(SIP/111,30,Ttr)
exten=> admin,3,Hangup
либо такую строку регистрации register => tcp://admin:pass@286970.lp.centertelecom.ru, т.е. без опции [/extension] тогда на s падает
exten=> admin,2,Dial(SIP/111,30,Ttr)
exten=> admin,3,Hangup
либо такую строку регистрации register => tcp://admin:pass@286970.lp.centertelecom.ru, т.е. без опции [/extension] тогда на s падает
ЛС: @rostel
Re: Транк до Ростелекома
Решает проблему, перевод АТС-ки в облако. 100% проблема в роутере. У меня такая проблема возникала раз 10, когда клиент в упор хотел АТСку в офисе, и сажал весь трафик на обычный бытовой роутер. Форвандингки по портам до ****** - одного места))) Все равно валятся соединения через час - пару часов, иногда часов на 6 хватало.
Re: Транк до Ростелекома
globalmp, эта дискуссия состоялась почти 5 лет назад.
Re: Транк до Ростелекома
ded, да и уверенность товарища забавная.
6 серверов работают через бытовые роутеры и каналы (не поддерживаю, но деваться некуда, заказчики так решили).
Нагрузка кое-где реально большая, одновременно 6-10 каналов открыто в рабочее время постоянно (холодный обзвон без продыха).
И ростелеком - в одной конторе. Все работает.
6 серверов работают через бытовые роутеры и каналы (не поддерживаю, но деваться некуда, заказчики так решили).
Нагрузка кое-где реально большая, одновременно 6-10 каналов открыто в рабочее время постоянно (холодный обзвон без продыха).
И ростелеком - в одной конторе. Все работает.
облачные и локальные сервера asterisk/freepbx/a2billing/crm с полной техподдержкой. skype: whoim2, sipuri: whoim@asterisk.ru
