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Нет исходящих вывозов через asterisk

0

Добрый день.

Установил, настроил asterisk - подключается к rosnet'у и имеет входящие и исходящие звонки. К астериску подрубаются SIP телефоны и принимают/совершают вызовы.

Все работало нормально на исходящие и на входящие. Но после того как начал ковыряться с кодеками - перестали работать исходящие вызовы, хотя если софтфоном подрубится к роснету - все работает прекрасно.

Не подскажите - где может быть косяк?

Вот конфиги: sip.conf:

[general]

register => user:123456@6653055.phone.rosnet.ru/6653055

localnet = 192.168.1.0/255.255.255.0
externip = 87.244.7.206
externrefresh = 60
nat = no
canreinvite = no
context = 6653055-is

language=ru
defaultexpiry=600

allowguest = no
bindport = 5060

allowoverlap = no

udpbindaddr = 0.0.0.0
tcpenable = no
tcpbindaddr = 0.0.0.0

srvlookup = yes
subscribecontext = default

;allow = all
disallow = all
allow = alaw
allow = ulaw
allow = g729
allow = gsm

[sipnet]
secret = 123456
defaultuser = user
username = user
trunkname = sipnet
host = 6653055.phone.rosnet.ru
context = 6653055-is
insecure = invite
fromuser = 6653055
fromdomain = 6653055.phone.rosnet.ru
type = peer
disallow = all
allow = alaw
allow = ulaw
allow = g729
allow = gsm
nat = no
canreinvite = no
dtmfmode = info
callcounter = yes

users.conf:

[general]
fullname = New User
userbase = 6000
hasvoicemail = yes
vmsecret = 1234
hassip = yes
hasiax = yes
hasmanager = no
callwaiting = yes
threewaycalling = yes
callwaitingcallerid = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
callgroup = 1
pickupgroup = 1

[1002]
type=friend
nat = yes
canreinvite = no
hassip = yes
defaultuser = 1002
context=office-users
secret=sandello
cid_number = 1002
vmsecret=123
host=dynamic
dtmfmode=rfc2833
username=1002
disallow=all
allow=alaw
allow=ulaw
allow=g729
allow=gsm

extensions.conf:

[6653055-is]

    exten => user,1,Answer
    exten => user,2,Playback(privetstvie-work)
    exten => user,3,Dial(SIP/1002)
    exten => user,4,Hangup

    [office-users]
    exten => _XXXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipnet,30,r)

Звонки извне проходят замечательно, а вот исходящие не идут... Вот логи, при попытке исходящего звонка:

<--- SIP read from UDP:213.108.18.100:60950 --->
INVITE sip:79671983788@87.244.7.206 SIP/2.0

Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK807476705

From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098

To: <sip:79671983788@87.244.7.206>

Call-ID: 527852667@10.0.2.134

CSeq: 1 INVITE

Contact: <sip:1002@10.0.2.134:5062>

Content-Type: application/sdp

Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE

Max-Forwards: 70

User-Agent: Yealink SIP-T18 18.0.14.5

Supported: replaces

Allow-Events: talk,hold,conference,refer,check-sync

Content-Length: 290



v=0

o=- 20006 20006 IN IP4 10.0.2.134

s=SDP data

c=IN IP4 10.0.2.134

t=0 0

m=audio 11792 RTP/AVP 8 0 9 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:9 G722/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

a=sendrecv


<------------->
--- (14 headers 14 lines) ---

gateway*CLI> 
Sending to 10.0.2.134 : 5062 (no NAT)

gateway*CLI> 
Using INVITE request as basis request - 527852667@10.0.2.134
Found peer '1002' for '1002' from 213.108.18.100:60950

<--- Reliably Transmitting (NAT) to 213.108.18.100:60950 --->
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK807476705;received=213.108.18.100

From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098

To: <sip:79671983788@87.244.7.206>;tag=as1bda2e66

Call-ID: 527852667@10.0.2.134

CSeq: 1 INVITE

Server: Asterisk PBX 1.6.2.20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4509296c"

Content-Length: 0




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

gateway*CLI> 
Scheduling destruction of SIP dialog '527852667@10.0.2.134' in 32000 ms (Method: INVITE)

gateway*CLI> 

<--- SIP read from UDP:213.108.18.100:60950 --->
ACK sip:79671983788@87.244.7.206 SIP/2.0

Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK807476705

From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098

To: <sip:79671983788@87.244.7.206>;tag=as1bda2e66

Call-ID: 527852667@10.0.2.134

CSeq: 1 ACK

Content-Length: 0




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

gateway*CLI> 
--- (7 headers 0 lines) ---

gateway*CLI> 

<--- SIP read from UDP:213.108.18.100:60950 --->
INVITE sip:79671983788@87.244.7.206 SIP/2.0

Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK248076524

From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098

To: <sip:79671983788@87.244.7.206>

Call-ID: 527852667@10.0.2.134

CSeq: 2 INVITE

Contact: <sip:1002@10.0.2.134:5062>

Authorization: Digest username="1002", realm="asterisk", nonce="4509296c", uri="sip:79671983788@87.244.7.206", response="7a447fd38d7a9c25bab04cfe42ca0dc7", algorithm=MD5

Content-Type: application/sdp

Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE

Max-Forwards: 70

User-Agent: Yealink SIP-T18 18.0.14.5

Supported: replaces

Allow-Events: talk,hold,conference,refer,check-sync

Content-Length: 290



v=0

o=- 20006 20006 IN IP4 10.0.2.134

s=SDP data

c=IN IP4 10.0.2.134

t=0 0

m=audio 11792 RTP/AVP 8 0 9 18 101

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:9 G722/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=fmtp:101 0-15

a=rtpmap:101 telephone-event/8000

a=sendrecv


<------------->
--- (15 headers 14 lines) ---
Sending to 213.108.18.100 : 60950 (NAT)
Using INVITE request as basis request - 527852667@10.0.2.134
Found peer '1002' for '1002' from 213.108.18.100:60950

gateway*CLI> 
Found RTP audio format 8

gateway*CLI> 
Found RTP audio format 0
Found RTP audio format 9
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 G722 for ID 9
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x110c (ulaw|alaw|g729|g722)/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 10.0.2.134:11792
Looking for 79671983788 in office-users (domain 87.244.7.206)

gateway*CLI> 
list_route: hop: <sip:1002@10.0.2.134:5062>

gateway*CLI> 

<--- Transmitting (NAT) to 213.108.18.100:60950 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK248076524;received=213.108.18.100

From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098

To: <sip:79671983788@87.244.7.206>

Call-ID: 527852667@10.0.2.134

CSeq: 2 INVITE

Server: Asterisk PBX 1.6.2.20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Contact: <sip:79671983788@87.244.7.206>

Content-Length: 0




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

gateway*CLI> 
Audio is at 87.244.7.206 port 19186
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP

gateway*CLI> 
Reliably Transmitting (no NAT) to 195.90.137.30:5060:
INVITE sip:79671983788@6653055.phone.rosnet.ru SIP/2.0

Via: SIP/2.0/UDP 87.244.7.206:5060;branch=z9hG4bK794bb845;rport

Max-Forwards: 70

From: "New User" <sip:6653055@6653055.phone.rosnet.ru>;tag=as182b55f4

To: <sip:79671983788@6653055.phone.rosnet.ru>

Contact: <sip:6653055@87.244.7.206>

Call-ID: 4d79ff2d5828fccb2546befa44fdee29@6653055.phone.rosnet.ru

CSeq: 102 INVITE

User-Agent: Asterisk PBX 1.6.2.20

Date: Fri, 21 Oct 2011 11:13:12 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 228



v=0

o=root 2014370682 2014370682 IN IP4 87.244.7.206

s=Asterisk PBX 1.6.2.20

c=IN IP4 87.244.7.206

t=0 0

m=audio 19186 RTP/AVP 8 0 3

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=ptime:20

a=sendrecv


---

gateway*CLI> 

<--- Transmitting (NAT) to 213.108.18.100:60950 --->
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK248076524;received=213.108.18.100

From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098

To: <sip:79671983788@87.244.7.206>;tag=as603a5a56

Call-ID: 527852667@10.0.2.134

CSeq: 2 INVITE

Server: Asterisk PBX 1.6.2.20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Contact: <sip:79671983788@87.244.7.206>

Content-Length: 0


gateway*CLI> 



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

gateway*CLI> 

<--- SIP read from UDP:195.90.137.30:5060 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 87.244.7.206:5060;branch=z9hG4bK794bb845;received=87.244.7.206;rport=5060

From: "New User" <sip:6653055@6653055.phone.rosnet.ru>;tag=as182b55f4

To: <sip:79671983788@6653055.phone.rosnet.ru>;tag=8KUVKeRbYer8uhjEGUvKpmDJbsn8iAEq

Call-ID: 4d79ff2d5828fccb2546befa44fdee29@6653055.phone.rosnet.ru

CSeq: 102 INVITE

Contact: <sip:79671983788@195.90.137.30:5060>

Max-Forwards: 70

User-Agent: Svetets CallManager R12642

Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, PUBLISH, SUBSCRIBE, OPTIONS, INFO

Content-Length: 0




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

gateway*CLI> 

<--- SIP read from UDP:195.90.137.30:5060 --->
SIP/2.0 603 Decline

Via: SIP/2.0/UDP 87.244.7.206:5060;branch=z9hG4bK794bb845;received=87.244.7.206;rport=5060

From: "New User" <sip:6653055@6653055.phone.rosnet.ru>;tag=as182b55f4

To: <sip:79671983788@6653055.phone.rosnet.ru>;tag=9cGY5h7SMFgutpbfCIKLFWT1RZWiKDbU

Call-ID: 4d79ff2d5828fccb2546befa44fdee29@6653055.phone.rosnet.ru

CSeq: 102 INVITE

Contact: <sip:79671983788@195.90.137.30:5060>

Max-Forwards: 70

User-Agent: Svetets CallManager R12642

Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, MESSAGE, PUBLISH, SUBSCRIBE, OPTIONS, INFO

Content-Length: 0




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

gateway*CLI> 
Transmitting (no NAT) to 195.90.137.30:5060:
ACK sip:79671983788@6653055.phone.rosnet.ru SIP/2.0

Via: SIP/2.0/UDP 87.244.7.206:5060;branch=z9hG4bK794bb845;rport

Max-Forwards: 70

From: "New User" <sip:6653055@6653055.phone.rosnet.ru>;tag=as182b55f4

To: <sip:79671983788@6653055.phone.rosnet.ru>;tag=9cGY5h7SMFgutpbfCIKLFWT1RZWiKDbU

Contact: <sip:6653055@87.244.7.206>

Call-ID: 4d79ff2d5828fccb2546befa44fdee29@6653055.phone.rosnet.ru

CSeq: 102 ACK

User-Agent: Asterisk PBX 1.6.2.20

Content-Length: 0



gateway*CLI> 


gateway*CLI> 

---

gateway*CLI> 

<--- Reliably Transmitting (NAT) to 213.108.18.100:60950 --->
SIP/2.0 486 Busy Here

Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK248076524;received=213.108.18.100

From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098

To: <sip:79671983788@87.244.7.206>;tag=as603a5a56

Call-ID: 527852667@10.0.2.134

CSeq: 2 INVITE

Server: Asterisk PBX 1.6.2.20

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

X-Asterisk-HangupCause: Call Rejected

X-Asterisk-HangupCauseCode: 21

Content-Length: 0




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

gateway*CLI> 

<--- SIP read from UDP:213.108.18.100:60950 --->
ACK sip:79671983788@87.244.7.206 SIP/2.0

Via: SIP/2.0/UDP 10.0.2.134:5062;branch=z9hG4bK248076524

From: "Operator 1002" <sip:1002@87.244.7.206>;tag=1497460098

To: <sip:79671983788@87.244.7.206>;tag=as603a5a56

Call-ID: 527852667@10.0.2.134

CSeq: 2 ACK

Content-Length: 0




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

gateway*CLI> 
--- (7 headers 0 lines) ---
Really destroying SIP dialog '527852667@10.0.2.134' Method: ACK

Заранее спасибо.

удалить закрыть спам изменить тег редактировать

спросил 2011-10-21 15:16:14 +0400

fr33m2n Gravatar fr33m2n
1 2 2

Comments

чето мне кажеться никто эту простыню читать не будет....

meral ( 2011-10-23 00:10:32 +0400 )редактировать

1 Ответ

0

Где конфиг rosnet'у ? или он у вас через сип нет уходит?

Воспользуйтесь такой командой при совершении исходящего звонка

tcpdump -v -s0 udp and port 5060 -w /root/dump_debug.cup И расшифруйте его Wireshark`om

Очень не понятный дебаг который выслали, но суда по всему ваш сервер не может договорится с роснетом о кодеках. (спасибо кеэП)

ссылка удалить спам редактировать

ответил 2011-10-23 02:23:14 +0400

gmurik Gravatar gmurik
15 11 2 9
http://gmurik.ru/

Comments

циско по кодекам выдает not aceptable here. а там decline. это просто надо провайдера спросить.

meral ( 2011-10-23 06:04:27 +0400 )редактировать

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Задан: 2011-10-21 15:16:14 +0400

Просмотрен: 1,262 раз

Обновлен: Oct 23 '11

Проект компании "АТС Дизайн"
Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.