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SIP

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

Ответить
Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: SIP

Сообщение Vlad1983 »

так откуда они тогда беруться у вас?
с чего вобще взято предположение из стартового поста?

sip set debug on по всем
после него sip set debug peer 2002 бессмысленно

sip set debug off
sip set debug peer 2002
будете отбирать по конкретному пиру
ЛС: @rostel
Вернуться к началу
paaandaaa3
Сообщения: 13
Зарегистрирован: 23 ноя 2011, 16:54

Re: SIP

Сообщение paaandaaa3 »

в общем слушаю я тока один пир
как-то так


== Using SIP RTP CoS mark 5
-- Executing [2000@phones:1] Dial("SIP/4000-000000f7", "SIP/2000") in new stack
== Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.37:5060:
INVITE sip:2000@192.168.1.37:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;branch=z9hG4bK13923868
Max-Forwards: 70
From: "IP4000" <sip:4000@192.168.1.34>;tag=as007a326c
To: <sip:2000@192.168.1.37:5060>
Contact: <sip:4000@192.168.1.34:5060>
Call-ID: 120bfd8b5caab5526ba5f73504b1e2f0@192.168.1.34:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.7.1
Date: Wed, 23 Nov 2011 18:26:42 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 1898173449 1898173449 IN IP4 192.168.1.34
s=Asterisk PBX 1.8.7.1
c=IN IP4 192.168.1.34
t=0 0
m=audio 17098 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called SIP/2000

<--- SIP read from UDP:192.168.1.37:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.34:5060;branch=z9hG4bK13923868
From: "IP4000" <sip:4000@192.168.1.34>;tag=as007a326c
To: <sip:2000@192.168.1.37:5060>
Call-ID: 120bfd8b5caab5526ba5f73504b1e2f0@192.168.1.34:5060
CSeq: 102 INVITE
User-Agent: ATA 211 V1.0.0.46
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.37:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.34:5060;branch=z9hG4bK13923868
From: "IP4000" <sip:4000@192.168.1.34>;tag=as007a326c
To: <sip:2000@192.168.1.37:5060>;tag=d34d98adad09a309
Call-ID: 120bfd8b5caab5526ba5f73504b1e2f0@192.168.1.34:5060
CSeq: 102 INVITE
Contact: <sip:2000@192.168.1.37:5060>
User-Agent: ATA 211 V1.0.0.46
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
-- SIP/2000-000000f8 is ringing

<--- SIP read from UDP:192.168.1.37:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.34:5060;branch=z9hG4bK13923868
From: "IP4000" <sip:4000@192.168.1.34>;tag=as007a326c
To: <sip:2000@192.168.1.37:5060>;tag=d34d98adad09a309
Call-ID: 120bfd8b5caab5526ba5f73504b1e2f0@192.168.1.34:5060
CSeq: 102 INVITE
Contact: <sip:2000@192.168.1.37:5060>
User-Agent: ATA 211 V1.0.0.46
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length: 210

v=0
o=- 6666 137 IN IP4 192.168.1.37
s=SIP Call
c=IN IP4 192.168.1.37
t=0 0
m=audio 5004 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
<------------->
--- (11 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/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 192.168.1.37:5004
list_route: hop: <sip:2000@192.168.1.37:5060>
set_destination: Parsing <sip:2000@192.168.1.37:5060> for address/port to send to
set_destination: set destination to 192.168.1.37:5060
Transmitting (no NAT) to 192.168.1.37:5060:
ACK sip:2000@192.168.1.37:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;branch=z9hG4bK704fc2a6
Max-Forwards: 70
From: "IP4000" <sip:4000@192.168.1.34>;tag=as007a326c
To: <sip:2000@192.168.1.37:5060>;tag=d34d98adad09a309
Contact: <sip:4000@192.168.1.34:5060>
Call-ID: 120bfd8b5caab5526ba5f73504b1e2f0@192.168.1.34:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.7.1
Content-Length: 0


---
-- SIP/2000-000000f8 answered SIP/4000-000000f7
-- Remotely bridging SIP/4000-000000f7 and SIP/2000-000000f8
set_destination: Parsing <sip:2000@192.168.1.37:5060> for address/port to send to
set_destination: set destination to 192.168.1.37:5060
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.37:5060:
INVITE sip:2000@192.168.1.37:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;branch=z9hG4bK5c95a212
Max-Forwards: 70
From: "IP4000" <sip:4000@192.168.1.34>;tag=as007a326c
To: <sip:2000@192.168.1.37:5060>;tag=d34d98adad09a309
Contact: <sip:4000@192.168.1.34:5060>
Call-ID: 120bfd8b5caab5526ba5f73504b1e2f0@192.168.1.34:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 1898173449 1898173450 IN IP4 192.168.1.35
s=Asterisk PBX 1.8.7.1
c=IN IP4 192.168.1.35
t=0 0
m=audio 41000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.37:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.34:5060;branch=z9hG4bK5c95a212
From: "IP4000" <sip:4000@192.168.1.34>;tag=as007a326c
To: <sip:2000@192.168.1.37:5060>;tag=d34d98adad09a309
Call-ID: 120bfd8b5caab5526ba5f73504b1e2f0@192.168.1.34:5060
CSeq: 103 INVITE
User-Agent: ATA 211 V1.0.0.46
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.37:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.34:5060;branch=z9hG4bK5c95a212
From: "IP4000" <sip:4000@192.168.1.34>;tag=as007a326c
To: <sip:2000@192.168.1.37:5060>;tag=d34d98adad09a309
Call-ID: 120bfd8b5caab5526ba5f73504b1e2f0@192.168.1.34:5060
CSeq: 103 INVITE
Contact: <sip:2000@192.168.1.37:5060>
User-Agent: ATA 211 V1.0.0.46
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Type: application/sdp
Content-Length: 210

v=0
o=- 6666 138 IN IP4 192.168.1.37
s=SIP Call
c=IN IP4 192.168.1.37
t=0 0
m=audio 5004 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11,16
<------------->
--- (11 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/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 192.168.1.37:5004
set_destination: Parsing <sip:2000@192.168.1.37:5060> for address/port to send to
set_destination: set destination to 192.168.1.37:5060
Transmitting (no NAT) to 192.168.1.37:5060:
ACK sip:2000@192.168.1.37:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.34:5060;branch=z9hG4bK27dfbf43
Max-Forwards: 70
From: "IP4000" <sip:4000@192.168.1.34>;tag=as007a326c
To: <sip:2000@192.168.1.37:5060>;tag=d34d98adad09a309
Contact: <sip:4000@192.168.1.34:5060>
Call-ID: 120bfd8b5caab5526ba5f73504b1e2f0@192.168.1.34:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.7.1
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.37:5060 --->


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

<--- SIP read from UDP:192.168.1.37:5060 --->


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

<--- SIP read from UDP:192.168.1.37:5060 --->


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

<--- SIP read from UDP:192.168.1.37:5060 --->


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

<--- SIP read from UDP:192.168.1.37:5060 --->


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

<--- SIP read from UDP:192.168.1.37:5060 --->
Вернуться к началу
paaandaaa3
Сообщения: 13
Зарегистрирован: 23 ноя 2011, 16:54

Re: SIP

Сообщение paaandaaa3 »

А вот такая штука потом появляется

<--- SIP read from UDP:192.168.1.37:5060 --->
REGISTER sip:192.168.1.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bKd6c393e3
From: "2000" <sip:2000@192.168.1.34>;tag=cde09310c20e7a10
To: "2000" <sip:2000@192.168.1.34>
Call-ID: 8625cdd59adbd1bb@192.168.1.37
CSeq: 505 REGISTER
Contact: <sip:2000@192.168.1.37:5060>
Max-Forwards: 70
Supported: path
User-Agent: ATA 211 V1.0.0.46
Expires: 900
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.1.37:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.37:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bKd6c393e3;received=192.168.1.37
From: "2000" <sip:2000@192.168.1.34>;tag=cde09310c20e7a10
To: "2000" <sip:2000@192.168.1.34>;tag=as34349746
Call-ID: 8625cdd59adbd1bb@192.168.1.37
CSeq: 505 REGISTER
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="3d899b4f"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8625cdd59adbd1bb@192.168.1.37' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.37:5060 --->
REGISTER sip:192.168.1.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bKba5cf06e
From: "2000" <sip:2000@192.168.1.34>;tag=cde09310c20e7a10
To: "2000" <sip:2000@192.168.1.34>
Call-ID: 8625cdd59adbd1bb@192.168.1.37
CSeq: 506 REGISTER
Contact: <sip:2000@192.168.1.37:5060>
Authorization: Digest username="2000", realm="asterisk", nonce="3d899b4f", uri="sip:192.168.1.34:5060", response="3b8b6c51d477789d2cf83f9a75e42d97", algorithm=MD5
Max-Forwards: 70
Supported: path
User-Agent: ATA 211 V1.0.0.46
Expires: 900
Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.1.37:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.37:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bKba5cf06e;received=192.168.1.37
From: "2000" <sip:2000@192.168.1.34>;tag=cde09310c20e7a10
To: "2000" <sip:2000@192.168.1.34>;tag=as34349746
Call-ID: 8625cdd59adbd1bb@192.168.1.37
CSeq: 506 REGISTER
Server: Asterisk PBX 1.8.7.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 900
Contact: <sip:2000@192.168.1.37:5060>;expires=900
Date: Wed, 23 Nov 2011 18:37:21 GMT
Content-Length: 0
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paaandaaa3
Сообщения: 13
Зарегистрирован: 23 ноя 2011, 16:54

Re: SIP

Сообщение paaandaaa3 »

ddkprog писал(а):теги что ли используйте да?
то есть?) вроде нет..
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Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: SIP

Сообщение Vlad1983 »

да, накой было выкладывать нет там UPDATE один черт

попробовать добавить в пиры
directmedia=update,nonat
nat=yes быть не должно
ЛС: @rostel
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paaandaaa3
Сообщения: 13
Зарегистрирован: 23 ноя 2011, 16:54

Re: SIP

Сообщение paaandaaa3 »

ответ по существу..
Вернуться к началу
Vlad1983
Сообщения: 4251
Зарегистрирован: 09 авг 2011, 11:51

Re: SIP

Сообщение Vlad1983 »

А вот такая штука потом появляется
это вообще в топку
ЛС: @rostel
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paaandaaa3
Сообщения: 13
Зарегистрирован: 23 ноя 2011, 16:54

Re: SIP

Сообщение paaandaaa3 »

Vlad1983 писал(а):да, накой было выкладывать нет там UPDATE один черт

попробовать добавить в пиры
directmedia=update,nonat
nat=yes быть не должно
ЕСТЬ! СПАСИБО!
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