Помощь с разбором логов

Durimar · Сообщение **Durimar** » 22 ноя 2011, 09:05

Никак не могу настроить входящую и исходящую через сип транк...

при входящей:
В ваершарке и в режиме дебага ничего не видно, ничего не летит...
Со стороны софтсвича в трейсах виден лишь одно сообщение рисевд..

при исходящей:
Со стороны софт-свича в трейсах ничего не видно.
При этом, что в ваершарке, что в астериске, при дебаге, вижу обмен с сервером регистрации.

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from UDP:10.40.4.203:5060 --->
INVITE sip:9356656@10.40.4.200 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac739
From: "102"<sip:102@10.40.4.200>;tag=dac72c-cb04280a-0026cf00
To: <sip:9356656@10.40.4.200>
Call-ID: 48dabb67-dac70a@10.40.4.203
CSeq: 100 INVITE
Contact: <sip:102@10.40.4.203:5060>
Max-Forwards: 70
User-Agent: Watertek-UA/2.1
Content-Type: application/SDP
Content-Length: 236

v=0
o=9356656 20000001 20000001 IN IP4 10.40.4.203
s=A call
c=IN IP4 10.40.4.203
t=0 0
m=audio 2040 RTP/AVP 8 18 4 0
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 12 lines) ---
Sending to 10.40.4.203:5060 (no NAT)
Using INVITE request as basis request - 48dabb67-dac70a@10.40.4.203
Found peer '102' for '102' from 10.40.4.203:5060

<--- Reliably Transmitting (no NAT) to 10.40.4.203:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac739;received=10.40.4.203
From: "102"<sip:102@10.40.4.200>;tag=dac72c-cb04280a-0026cf00
To: <sip:9356656@10.40.4.200>;tag=as6bf9bfab
Call-ID: 48dabb67-dac70a@10.40.4.203
CSeq: 100 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="realm", nonce="7ea6fb05"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '48dabb67-dac70a@10.40.4.203' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:10.40.4.203:5060 --->
ACK sip:9356656@10.40.4.200 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac739;received=10.40.4.203
From: "102"<sip:102@10.40.4.200>;tag=dac72c-cb04280a-0026cf00
To: <sip:9356656@10.40.4.200>;tag=as6bf9bfab
Call-ID: 48dabb67-dac70a@10.40.4.203
CSeq: 100 ACK
Contact: <sip:102@10.40.4.203:5060>
Max-Forwards: 70
User-Agent: Watertek-UA/2.1
Content-Length: 0

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

<--- SIP read from UDP:10.40.4.203:5060 --->
INVITE sip:9356656@10.40.4.200 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac73a
From: "102"<sip:102@10.40.4.200>;tag=dac72c-cb04280a-0026cf00
To: <sip:9356656@10.40.4.200>
Call-ID: 48dabb67-dac70a@10.40.4.203
CSeq: 101 INVITE
Contact: <sip:102@10.40.4.203:5060>
Authorization: Digest username="102", realm="realm", nonce="7ea6fb05", response="589623b43e341999057b023cf602adc1", uri="sip:9356656@10.40.4.200", algorithm=MD5
Max-Forwards: 70
User-Agent: Watertek-UA/2.1
Content-Type: application/SDP
Content-Length: 236

v=0
o=9356656 20000001 20000001 IN IP4 10.40.4.203
s=A call
c=IN IP4 10.40.4.203
t=0 0
m=audio 2040 RTP/AVP 8 18 4 0
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 12 lines) ---
Sending to 10.40.4.203:5060 (no NAT)
Using INVITE request as basis request - 48dabb67-dac70a@10.40.4.203
Found peer '102' for '102' from 10.40.4.203:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format PCMU for ID 0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|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 10.40.4.203:2040
Looking for 9356656 in default (domain 10.40.4.200)
list_route: hop: <sip:102@10.40.4.203:5060>

<--- Transmitting (no NAT) to 10.40.4.203:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac73a;received=10.40.4.203
From: "102"<sip:102@10.40.4.200>;tag=dac72c-cb04280a-0026cf00
To: <sip:9356656@10.40.4.200>
Call-ID: 48dabb67-dac70a@10.40.4.203
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9356656@10.40.4.200:5060>
Content-Length: 0

<------------>
-- Executing [9356656@default:1] Dial("SIP/102-00000014", "SIP/356656@10.1.0.194") in new stack
== Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.0.194:5060:
INVITE sip:356656@10.1.0.194 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK79c3b9c6
Max-Forwards: 70
From: "102" <sip:102@10.40.4.200>;tag=as38f25050
To: <sip:356656@10.1.0.194>
Contact: <sip:102@10.40.4.200:5060>
Call-ID: 743bcaf410b98c925e5abdb249ea00e1@10.40.4.200:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.7.1
Date: Tue, 22 Nov 2011 03:53:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 332505897 332505897 IN IP4 10.40.4.200
s=Asterisk PBX 1.8.7.1
c=IN IP4 10.40.4.200
t=0 0
m=audio 16612 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called SIP/356656@10.1.0.194

<--- SIP read from UDP:10.1.0.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK79c3b9c6
To: <sip:356656@10.1.0.194>
From: "102"<sip:102@10.40.4.200>;tag=as38f25050
Call-ID: 743bcaf410b98c925e5abdb249ea00e1@10.40.4.200:5060
CSeq: 102 INVITE
User-Agent: ZTE-SBC
Content-Length: 0

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

<--- SIP read from UDP:10.1.0.194:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK79c3b9c6
To: <sip:356656@10.1.0.194>;tag=09c50b8009783f55-0acd5bc1
From: "102"<sip:102@10.40.4.200>;tag=as38f25050
Call-ID: 743bcaf410b98c925e5abdb249ea00e1@10.40.4.200:5060
CSeq: 102 INVITE
User-Agent: ZTE-SBC
X-ZTE-Cause: "SBC-4409"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 10.1.0.194:5060:
ACK sip:356656@10.1.0.194 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK79c3b9c6
Max-Forwards: 70
From: "102" <sip:102@10.40.4.200>;tag=as38f25050
To: <sip:356656@10.1.0.194>;tag=09c50b8009783f55-0acd5bc1
Contact: <sip:102@10.40.4.200:5060>
Call-ID: 743bcaf410b98c925e5abdb249ea00e1@10.40.4.200:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.7.1
Content-Length: 0

---
-- SIP/10.1.0.194-00000015 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/102-00000014' status is 'CONGESTION'

<--- Reliably Transmitting (no NAT) to 10.40.4.203:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac73a;received=10.40.4.203
From: "102"<sip:102@10.40.4.200>;tag=dac72c-cb04280a-0026cf00
To: <sip:9356656@10.40.4.200>;tag=as7ffa44ba
Call-ID: 48dabb67-dac70a@10.40.4.203
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0

<------------>
Really destroying SIP dialog '743bcaf410b98c925e5abdb249ea00e1@10.40.4.200:5060' Method: INVITE

<--- SIP read from UDP:10.40.4.203:5060 --->
ACK sip:9356656@10.40.4.200 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac73a;received=10.40.4.203
From: "102"<sip:102@10.40.4.200>;tag=dac72c-cb04280a-0026cf00
To: <sip:9356656@10.40.4.200>;tag=as7ffa44ba
Call-ID: 48dabb67-dac70a@10.40.4.203
CSeq: 101 ACK
Contact: <sip:102@10.40.4.203:5060>
Max-Forwards: 70
User-Agent: Watertek-UA/2.1
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '48dabb67-dac70a@10.40.4.203' Method: ACK
Really destroying SIP dialog '7a45200307191120@10.1.0.194' Method: OPTIONS

Vlad1983 · Сообщение **Vlad1983** » 22 ноя 2011, 09:13

для исходящий в пир добавить fromuser=номер_который_за вами_в_нужном_формате
а для входящих надо бы проверить сначала зареген или нет

jugatsu · Сообщение **jugatsu** » 22 ноя 2011, 10:14

при входящей:
В ваершарке и в режиме дебага ничего не видно, ничего не летит...
Со стороны софтсвича в трейсах виден лишь одно сообщение рисевд..

Покажи sip show registry и соответственно строку в sip.conf, которая отвечает за регистрацию на прокси прова. Попробуй позвонить на выданный номер и поймать INVITE

ngrep -d eth0 -q -W byline host <softswitch провайдера> INVITE

а также дай вывод iptables-save

Durimar · Сообщение **Durimar** » 22 ноя 2011, 11:29

добавил на внутренние номера поля fromuser
та же самая ерунда...

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from UDP:10.40.4.203:5060 --->
INVITE sip:9356656@10.40.4.200 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac71b
From: "102"<sip:102@10.40.4.200>;tag=dac718-cb04280a-001284c8
To: <sip:9356656@10.40.4.200>
Call-ID: 48dab635-dac704@10.40.4.203
CSeq: 100 INVITE
Contact: <sip:102@10.40.4.203:5060>
Max-Forwards: 70
User-Agent: Watertek-UA/2.1
Content-Type: application/SDP
Content-Length: 236

v=0
o=9356656 20000001 20000001 IN IP4 10.40.4.203
s=A call
c=IN IP4 10.40.4.203
t=0 0
m=audio 2032 RTP/AVP 8 18 4 0
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (11 headers 12 lines) ---
Sending to 10.40.4.203:5060 (no NAT)
Using INVITE request as basis request - 48dab635-dac704@10.40.4.203
Found peer '102' for '102' from 10.40.4.203:5060

<--- Reliably Transmitting (no NAT) to 10.40.4.203:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac71b;received=10.40.4.203
From: "102"<sip:102@10.40.4.200>;tag=dac718-cb04280a-001284c8
To: <sip:9356656@10.40.4.200>;tag=as386bfbd1
Call-ID: 48dab635-dac704@10.40.4.203
CSeq: 100 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0f68d9b2"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '48dab635-dac704@10.40.4.203' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:10.40.4.203:5060 --->
ACK sip:9356656@10.40.4.200 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac71b;received=10.40.4.203
From: "102"<sip:102@10.40.4.200>;tag=dac718-cb04280a-001284c8
To: <sip:9356656@10.40.4.200>;tag=as386bfbd1
Call-ID: 48dab635-dac704@10.40.4.203
CSeq: 100 ACK
Contact: <sip:102@10.40.4.203:5060>
Max-Forwards: 70
User-Agent: Watertek-UA/2.1
Content-Length: 0

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

<--- SIP read from UDP:10.40.4.203:5060 --->
INVITE sip:9356656@10.40.4.200 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac71c
From: "102"<sip:102@10.40.4.200>;tag=dac718-cb04280a-001284c8
To: <sip:9356656@10.40.4.200>
Call-ID: 48dab635-dac704@10.40.4.203
CSeq: 101 INVITE
Contact: <sip:102@10.40.4.203:5060>
Authorization: Digest username="102", realm="asterisk", nonce="0f68d9b2", response="64669574b4c17317e8c098e696c19fea", uri="sip:9356656@10.40.4.200", algorithm=MD5
Max-Forwards: 70
User-Agent: Watertek-UA/2.1
Content-Type: application/SDP
Content-Length: 236

v=0
o=9356656 20000001 20000001 IN IP4 10.40.4.203
s=A call
c=IN IP4 10.40.4.203
t=0 0
m=audio 2032 RTP/AVP 8 18 4 0
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 12 lines) ---
Sending to 10.40.4.203:5060 (no NAT)
Using INVITE request as basis request - 48dab635-dac704@10.40.4.203
Found peer '102' for '102' from 10.40.4.203:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format PCMU for ID 0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10d (g723|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 10.40.4.203:2032
Looking for 9356656 in default (domain 10.40.4.200)
list_route: hop: <sip:102@10.40.4.203:5060>

<--- Transmitting (no NAT) to 10.40.4.203:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac71c;received=10.40.4.203
From: "102"<sip:102@10.40.4.200>;tag=dac718-cb04280a-001284c8
To: <sip:9356656@10.40.4.200>
Call-ID: 48dab635-dac704@10.40.4.203
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:9356656@10.40.4.200:5060>
Content-Length: 0

<------------>
-- Executing [9356656@default:1] Dial("SIP/102-0000000e", "SIP/356656@10.1.0.194") in new stack
== Using SIP RTP CoS mark 5
Really destroying SIP dialog '48dab17c-dac6f9@10.40.4.203' Method: OPTIONS
Really destroying SIP dialog '425e200307191120@10.1.0.194' Method: OPTIONS
[Nov 22 12:10:25] NOTICE[16103]: chan_sip.c:12596 sip_reregister: -- Re-registration for 318105@10.1.0.194
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.1.0.194:5060:
REGISTER sip:10.1.0.194 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK5755be22
Max-Forwards: 70
From: <sip:318105@10.1.0.194>;tag=as4f4fcfd4
To: <sip:318105@10.1.0.194>
Call-ID: 34a672f967d9889777912768438a2d2c@127.0.0.2
CSeq: 110 REGISTER
User-Agent: Asterisk PBX 1.8.7.1
Authorization: Digest username="318105", realm="zte", algorithm=MD5, uri="sip:10.1.0.194", nonce="e33752cbeb60fc692277d1a8c1d7e973", response="232e3bb2f0e48836082be237a5206a4b"
Expires: 120
Contact: <sip:s@10.40.4.200:5060>
Content-Length: 0

---

<--- SIP read from UDP:10.1.0.194:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK5755be22
To: <sip:318105@10.1.0.194>
From: <sip:318105@10.1.0.194>;tag=as4f4fcfd4
Call-ID: 34a672f967d9889777912768438a2d2c@127.0.0.2
CSeq: 110 REGISTER
User-Agent: ZTE-SoftSwitch
WWW-Authenticate: Digest realm="zte", nonce="689246574200a94890b838e1825e805c", ZTE-ID=aa0f1c42d3671d701e07f69839426ce6
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Responding to challenge, registration to domain/host name 10.1.0.194
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.1.0.194:5060:
REGISTER sip:10.1.0.194 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK78c2fab5
Max-Forwards: 70
From: <sip:318105@10.1.0.194>;tag=as76f39881
To: <sip:318105@10.1.0.194>
Call-ID: 34a672f967d9889777912768438a2d2c@127.0.0.2
CSeq: 111 REGISTER
User-Agent: Asterisk PBX 1.8.7.1
Authorization: Digest username="318105", realm="zte", algorithm=MD5, uri="sip:10.1.0.194", nonce="689246574200a94890b838e1825e805c", response="4d64d4a7afdbb7903572585a7fc22d4a"
Expires: 120
Contact: <sip:s@10.40.4.200:5060>
Content-Length: 0

---

<--- SIP read from UDP:10.1.0.194:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK78c2fab5
To: <sip:318105@10.1.0.194>
From: <sip:318105@10.1.0.194>;tag=as76f39881
Call-ID: 34a672f967d9889777912768438a2d2c@127.0.0.2
CSeq: 111 REGISTER
Contact: <sip:s@10.40.4.200:5060>;expires=120
User-Agent: ZTE-SoftSwitch
Date: Tue, 22 Nov 2011 13:10:24 GMT
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Scheduling destruction of SIP dialog '34a672f967d9889777912768438a2d2c@127.0.0.2' in 32000 ms (Method: REGISTER)
[Nov 22 12:10:25] NOTICE[16103]: chan_sip.c:20148 handle_response_register: Outbound Registration: Expiry for 10.1.0.194 is 120 sec (Scheduling reregistration in 105 s)
Audio is at 5060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.1.0.194:5060:
INVITE sip:356656@10.1.0.194 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK6c70b038
Max-Forwards: 70
From: "102" <sip:102@10.40.4.200>;tag=as2a4eeab9
To: <sip:356656@10.1.0.194>
Contact: <sip:102@10.40.4.200:5060>
Call-ID: 5c7fb5cf2c6b3718009d74ef157e834c@10.40.4.200:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.7.1
Date: Tue, 22 Nov 2011 06:10:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 283

v=0
o=root 409574363 409574363 IN IP4 10.40.4.200
s=Asterisk PBX 1.8.7.1
c=IN IP4 10.40.4.200
t=0 0
m=audio 16634 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called SIP/356656@10.1.0.194

<--- SIP read from UDP:10.1.0.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK6c70b038
To: <sip:356656@10.1.0.194>
From: "102"<sip:102@10.40.4.200>;tag=as2a4eeab9
Call-ID: 5c7fb5cf2c6b3718009d74ef157e834c@10.40.4.200:5060
CSeq: 102 INVITE
User-Agent: ZTE-SBC
Content-Length: 0

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

<--- SIP read from UDP:10.1.0.194:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK6c70b038
To: <sip:356656@10.1.0.194>;tag=6699219f2aed08df-2f227540
From: "102"<sip:102@10.40.4.200>;tag=as2a4eeab9
Call-ID: 5c7fb5cf2c6b3718009d74ef157e834c@10.40.4.200:5060
CSeq: 102 INVITE
User-Agent: ZTE-SBC
X-ZTE-Cause: "SBC-4409"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 10.1.0.194:5060:
ACK sip:356656@10.1.0.194 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.200:5060;branch=z9hG4bK6c70b038
Max-Forwards: 70
From: "102" <sip:102@10.40.4.200>;tag=as2a4eeab9
To: <sip:356656@10.1.0.194>;tag=6699219f2aed08df-2f227540
Contact: <sip:102@10.40.4.200:5060>
Call-ID: 5c7fb5cf2c6b3718009d74ef157e834c@10.40.4.200:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.7.1
Content-Length: 0

---
-- SIP/10.1.0.194-0000000f is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/102-0000000e' status is 'CONGESTION'

<--- Reliably Transmitting (no NAT) to 10.40.4.203:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac71c;received=10.40.4.203
From: "102"<sip:102@10.40.4.200>;tag=dac718-cb04280a-001284c8
To: <sip:9356656@10.40.4.200>;tag=as5ee19e34
Call-ID: 48dab635-dac704@10.40.4.203
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0

<------------>
Really destroying SIP dialog '5c7fb5cf2c6b3718009d74ef157e834c@10.40.4.200:5060' Method: INVITE

<--- SIP read from UDP:10.40.4.203:5060 --->
ACK sip:9356656@10.40.4.200 SIP/2.0
Via: SIP/2.0/UDP 10.40.4.203:5060;branch=z9hG4bK-dac71c;received=10.40.4.203
From: "102"<sip:102@10.40.4.200>;tag=dac718-cb04280a-001284c8
To: <sip:9356656@10.40.4.200>;tag=as5ee19e34
Call-ID: 48dab635-dac704@10.40.4.203
CSeq: 101 ACK
Contact: <sip:102@10.40.4.203:5060>
Max-Forwards: 70
User-Agent: Watertek-UA/2.1
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '48dab635-dac704@10.40.4.203' Method: ACK

Кроме того межу сообщениям с заголовком trying(на внтренний с айпи 203) и сообщением invite на проксисервер регистрации есть задержка в ~40секунд

Может накосячил в конфигах??
sip.conf:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[general]
context=default

disallow=all
allow=alaw
allow=ulaw

udpbindaddr=10.40.4.200/5060

register => 318105:XXXXX@10.1.0.194

[318105]
type=friend
secret=54321
username=318105
;accountcode=3467
host=10.1.0.194
fromdomain=10.40.4.200
fromuser=318105
context=318105inc
insecure=no

[101]
type=friend
host=dynamic
username=101
fromuser=101
context=default
secret=123

[102]
type=friend
host=dynamic
username=102
fromuser=102
context=default
secret=123

extension.conf:

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

[general]
static = no
writeprotect = no
clearglobalvars = no

[318105inc]
exten => 318105,1,Dial(SIP/102)

;[outgoing_calls]
;exten => _X.,1,Dial(SIP/318105/${EXTEN})

[default]
exten => 101,1,Dial(SIP/101)
exten => 102,1,Dial(SIP/102)
exten => _9XXXXXX,1,Dial(SIP/${EXTEN:1}@10.1.0.194)

linux-xair*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
10.1.0.194:5060 N 318105 105 Registered Tue, 22 Nov 2011 12:03:24

zzuz · Сообщение **zzuz** » 22 ноя 2011, 11:35

Код: Выделить всё

udpbindaddr=10.40.4.200/5060
insecure=no

это откуда такой синтаксис?

думаю стоило бы в библиотеку для начала прогуляться , не?

Durimar · Сообщение **Durimar** » 22 ноя 2011, 12:19

Синтаксис и insecure поправил, как и ожидалось без изменений...

В библиотеку?? А я откуда? Все сразу не разберешь, во всем разом не разберешься..

40секундная задержка была изза косяков на сети... В остальном без изменений.

jugatsu · Сообщение **jugatsu** » 22 ноя 2011, 12:20

Examples:
;
;register => 1234:password@mysipprovider.com
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@sip_proxy/1234
;
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
; connect to local extension 1234 in extensions.conf, default context,
; unless you configure a [sip_proxy] section below, and configure a
; context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate inbound and outbound sections for SIP providers
; (instead of type=friend) if you have calls in both directions
;
;register => 3456@mydomain:5082::@mysipprovider.com
;
; Note that in this example, the optional authuser and secret portions have
; been left blank because we have specified a port in the user section

Vlad1983 · Сообщение **Vlad1983** » 22 ноя 2011, 12:32

что это за кухнЯ
exten => _9XXXXXX,1,Dial(SIP/${EXTEN:1}@10.1.0.194)

при таком синтаксисе естественно никакие настройки в пирах не работают
при этом чуть выше используется нормальный
заменить на exten => _9XXXXXX,1,Dial(SIP/318105/${EXTEN:1})

и не надо это все в default совать сразу кандидат на слив через себя левого трафа

Durimar · Сообщение **Durimar** » 22 ноя 2011, 14:51

Vlad1983
Сделал как ты сказал - заработало... Неплохо было бы мне еще с этим разобратся... Но на готовом примере с помощью библиотеки, проб и ошибок надеюсь разбрусь....

jugatsu
Разумеется читал, прописывал при регистрации екстеншен, но регится транк не хотел, ругался...
Только догнал что необходимо было дописать имя авторизации...

Все работает. Всем спасибо

jugatsu · Сообщение **jugatsu** » 22 ноя 2011, 15:06

покажи строку register

Помощь с разбором логов

Помощь с разбором логов

Re: Помощь с разбором логов

Re: Помощь с разбором логов

Re: Помощь с разбором логов

Re: Помощь с разбором логов

Re: Помощь с разбором логов

Re: Помощь с разбором логов

Re: Помощь с разбором логов

Re: Помощь с разбором логов

Re: Помощь с разбором логов