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Обрывается исходящий звонок на Addpac IP200 Обрывается исходящий звонок на 20-30сек, а входящие звонки идут

Нужна помощь.

Есть железка Addpac IP200 работает как SIP телефон, подключен к SIP серверу в качестве которого выступает SIP плата на АТС Меридиан.

Так вот, проблема в том, что при исходящем звонке с Addpac IP200 на любой SIP номер через 20-30 секунд разговора рвется соединение. Входящие звонки на Addpac IP200 проходят нормально. Если вместо Addpac IP200 поставить копм с софтовым клиентом Xlite то все звонки проходят без проблем и входящие и исходящие.

 

Из дебага видно что сессия рвется по ошибке 504.

Вот конфиг Addpac IP200(версия :ip200_g2_v8_41_028)

hostname IP200

clock timezone EET 2

!

username хххх password ххххх administrator

!

!

interface Loopback0

ip address 127.0.0.1 255.0.0.0

!

interface FastEthernet0/0

ip address dhcp

speed auto

!

interface FastEthernet0/1

no ip address

speed auto

!

! ip route 0.0.0.0 0.0.0.0 x.y.64.1 via dhcp

!

!

!

snmp name IP200_G2

snmp enable-cputrap

!

ip udp blackhole

http server

!

!

! IP PHONE OSD configuration.

!

osd

language english

network signaling sip

network sscp disable

network lan-dhcp none

phone lcd-type graphic

phone ring-type 1

phone volume ring 2

phone volume input 7

phone volume output 5

phone volume micbooster disable

phone auto-hook-on disable

phone display-name Odessa

phone voice-codec 3

phone dnd-mode silence

!

! SSCP configuration.!

!

!

! SSCP Static CM List

sscp

!

! SSCP Dynamic CM List

sscp

!

!

sscp

call-manager broadcast port 8855

logger disable

logger level info

!

!

!

!

! VoIP configuration.

!

!

! Voice service voip configuration.

!

voice service voip

fax protocol t38 redundancy 0

fax rate disable

h323 call start fast

h323 call tunnel enable

announcement language english

!

!

! Voice port configuration.

!

! SPEECH

voice-port 0/0

!

!

! FXO

voice-port 0/1

!

!

!

!

! Pots peer configuration.

!

dial-peer voice 0 pots

destination-pattern 6010

port 0/0

!

!

!

! Voip peer configuration.

!

dial-peer voice 1001 voip

destination-pattern T

session target sip-server

session protocol sip

voice-class codec 0

dtmf-relay rtp-2833

huntstop

!

dial-peer voice 1002 voip

destination-pattern T

session target ras

voice-class codec 0

dtmf-relay h245-alphanumeric

preference 1

huntstop

!

!

!

!

!

!

! Gateway configuration.

!

gateway

h323-id voip.x.y.64.210

!

!

! Codec classes configuration.

!

voice class codec 0

codec preference 1 g729

codec preference 2 g7231r63

codec preference 3 g711alaw

codec preference 4 g711ulaw

!

!

!

! SIP UA configuration.

!

sip-ua

user-register

sip-username 6010

sip-password xxxxxxxx

sip-server x.y.69.1

srv enable

rport enable

media-channel early

register e164

!

!

! MGCP configuration.

!

mgcp

codec g729

vad

!

!

! Tones

!

!

!

!

line console

!

line vty

!

!

! APOS File System

!

! mount mem 512 /tmp

!

!

sms

quata 30

!

Буду рад любой помощи в этом вопросе.

 

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Стандартная лажа Addpac ---- уберите VAD.

Кроме VAD еще может, что-то мешать? Я уже пробовал отключать VAD и результата это не дало.

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Кроме VAD еще может, что-то мешать? Я уже пробовал отключать VAD и результата это не дало.

Так бывает, когда АСК некорректно долетает. Покажите sip flow

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Очень похоже на то, что Addpac или не шлет ACK на SIP OK сервера, или шлет в некорректном виде ... надо бы дамп обмена глянуть.

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Наконец появилась возможность снять дебаг

Вот, что он отвечает на debug voip sip

 

 

IP200#

Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060

INVITE sip:6012@10.100.69.1 SIP/2.0

Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a425

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 25 INVITE

Supported: replaces, timer, 100rel, early-session

Min-SE: 1800

Date: Fri, 01 Apr 2011 13:12:02 GMT

Session-Expires: 1800

User-Agent: AddPac SIP Gateway

Contact: <sip:6010@10.115.64.98>

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO

Allow-Events: talk, hold, conference

Content-Type: application/sdp

Content-Length: 305

Max-Forwards: 70

 

v=0

o=6010 1301663522 1301663522 IN IP4 10.115.64.98

s=AddPac Gateway SDP

c=IN IP4 10.115.64.98

t=1301663522 0

m=audio 23002 RTP/AVP 18 4 8 0 101

a=ptime:20

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 100 Trying

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 25 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a425

Content-Length: 0

Organization: 10.100.69.1

Server: Avaya SIP Enablement Services

 

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 407 Proxy Authentication Required

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120036123744

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 25 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a425

Content-Length: 0

Proxy-Authenticate: Digest realm="10.100.69.1",domain="10.100.69.1",nonce="MTI3MDEyMDAzNjpTREZTZXJ2ZXJTZWNyZXRLZXk6MjA2Nzg0NTQ0Ng==",algorithm=MD5

Server: Avaya SIP Enablement Services

Organization: 10.100.69.1

 

 

 

Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060

ACK sip:6012@10.100.69.1 SIP/2.0

Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a425

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120036123744

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 25 ACK

Content-Length: 0

Max-Forwards: 70

 

 

 

Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060

INVITE sip:6012@10.100.69.1 SIP/2.0

Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a426

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 26 INVITE

Supported: replaces, timer, 100rel, early-session

Min-SE: 1800

Date: Fri, 01 Apr 2011 13:12:02 GMT

Session-Expires: 1800

User-Agent: AddPac SIP Gateway

Contact: <sip:6010@10.115.64.98>

Accept: application/sdp

Proxy-Authorization: Digest username="6010", realm="10.100.69.1", nonce="MTI3MDEyMDAzNjpTREZTZXJ2ZXJTZWNyZXRLZXk6MjA2Nzg0NTQ0Ng==", ri="sip:6012@10.100.69.1", response="e95608718af7e87c5d012a852a23e5d6", algorithm=MD5

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO

Allow-Events: talk, hold, conference

Content-Type: application/sdp

Content-Length: 315

Max-Forwards: 70

 

v=0

o=6010 1301663522 1301663522 IN IP4 10.115.64.98

s=AddPac Gateway SDP

c=IN IP4 10.115.64.98

t=1301663522 0

m=audio 23002 RTP/AVP 18 4 8 0 101

a=ptime:20

a=rtpmap:18 G729/8000/1

a=rtpmap:4 G723/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:0 PCMU/8000/1

a=rtpmap:101 telephone-event/8000/1

a=fmtp:101 0-15

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 100 Trying

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 26 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a426

Content-Length: 0

Organization: 10.100.69.1

Server: Avaya SIP Enablement Services

 

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 422 Session Interval Too Small

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120037123747

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 26 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;psrrposn=1;received=10.115.64.98;branch=z9hG4bK224d7919a426

Server: Avaya CM/R015x.02.1.016.4

Min-SE: 3600

Content-Length: 0

 

 

 

Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060

ACK sip:6012@10.100.69.1 SIP/2.0

Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a426

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120037123747

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 26 ACK

Proxy-Authorization: Digest username="6010", realm="10.100.69.1", nonce="MTI3MDEyMDAzNjpTREZTZXJ2ZXJTZWNyZXRLZXk6MjA2Nzg0NTQ0Ng==", ri="sip:6012@10.100.69.1", response="7991b5da1f3102c5f03a9a71c41aab3e", algorithm=MD5

Content-Length: 0

Max-Forwards: 70

 

 

 

 

Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060

INVITE sip:6012@10.100.69.1 SIP/2.0

Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a427

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 INVITE

Supported: replaces, timer, 100rel, early-session

Min-SE: 3600

Date: Fri, 01 Apr 2011 13:12:02 GMT

Session-Expires: 3600

User-Agent: AddPac SIP Gateway

Contact: <sip:6010@10.115.64.98>

Accept: application/sdp

Proxy-Authorization: Digest username="6010", realm="10.100.69.1", nonce="MTI3MDEyMDAzNjpTREZTZXJ2ZXJTZWNyZXRLZXk6MjA2Nzg0NTQ0Ng==", uri="sip:6012@10.100.69.1", response="e95608718af7e87c5d012a852a23e5d6", algorithm=MD5

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, PRACK, REFER, NOTIFY, INFO

Allow-Events: talk, hold, conference

Content-Type: application/sdp

Content-Length: 315

Max-Forwards: 70

 

v=0

o=6010 1301663522 1301663522 IN IP4 10.115.64.98

s=AddPac Gateway SDP

c=IN IP4 10.115.64.98

t=1301663522 0

m=audio 23002 RTP/AVP 18 4 8 0 101

a=ptime:20

a=rtpmap:18 G729/8000/1

a=rtpmap:4 G723/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:0 PCMU/8000/1

a=rtpmap:101 telephone-event/8000/1

a=fmtp:101 0-15

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 100 Trying

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427

Content-Length: 0

Organization: 10.100.69.1

Server: Avaya SIP Enablement Services

 

 

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 180 Ringing

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427

Record-Route: <sip:10.100.69.1:6001;lr;transport=tls>

Record-Route: <sip:10.100.69.1:5060;lr>

Server: Avaya CM/R015x.02.1.016.4

Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls>

P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone>

Supported: timer,replaces,join,histinfo,100rel

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

RSeq: 1

Require: 100rel

Content-Type: application/sdp

Content-Length: 184

 

v=0

o=- 1 2 IN IP4 10.100.69.1

s=-

c=IN IP4 10.100.69.2

b=AS:64

t=0 0

m=audio 2050 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

 

 

Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060

PRACK sip:6012@10.100.69.1:6001;transport=tls;user=phone SIP/2.0

Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a428

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 28 PRACK

Route: <sip:10.100.69.1:5060;lr>,<sip:10.100.69.1:6001;lr;transport=tls>

Date: Fri, 01 Apr 2011 13:12:03 GMT

User-Agent: AddPac SIP Gateway

RAck: 1 27 INVITE

Contact: <sip:6010@10.115.64.98>

Content-Length: 0

Max-Forwards: 70

 

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 407 Proxy Authentication Required

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 28 PRACK

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a428

Content-Length: 0

Proxy-Authenticate: Digest realm="10.100.69.1",domain="10.100.69.1",nonce="MTI3MDEyMDAzNzpTREZTZXJ2ZXJTZWNyZXRLZXk6Mzk3MTA1NjA0",algorithm=MD5

Server: Avaya SIP Enablement Services

Organization: 10.100.69.1

 

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 180 Ringing

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427

Record-Route: <sip:10.100.69.1:6001;lr;transport=tls>

Record-Route: <sip:10.100.69.1:5060;lr>

Server: Avaya CM/R015x.02.1.016.4

Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls>

P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone>

Supported: timer,replaces,join,histinfo,100rel

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

RSeq: 1

Require: 100rel

Content-Type: application/sdp

Content-Length: 184

 

v=0

o=- 1 2 IN IP4 10.100.69.1

s=-

c=IN IP4 10.100.69.2

b=AS:64

t=0 0

m=audio 2050 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

 

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 180 Ringing

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427

Record-Route: <sip:10.100.69.1:6001;lr;transport=tls>

Record-Route: <sip:10.100.69.1:5060;lr>

Server: Avaya CM/R015x.02.1.016.4

Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls>

P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone>

Supported: timer,replaces,join,histinfo,100rel

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

RSeq: 1

Require: 100rel

Content-Type: application/sdp

Content-Length: 184

 

v=0

o=- 1 2 IN IP4 10.100.69.1

s=-

c=IN IP4 10.100.69.2

b=AS:64

t=0 0

m=audio 2050 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 180 Ringing

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427

Record-Route: <sip:10.100.69.1:6001;lr;transport=tls>

Record-Route: <sip:10.100.69.1:5060;lr>

Server: Avaya CM/R015x.02.1.016.4

Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls>

P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone>

Supported: timer,replaces,join,histinfo,100rel

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

RSeq: 1

Require: 100rel

Content-Type: application/sdp

Content-Length: 184

 

v=0

o=- 1 2 IN IP4 10.100.69.1

s=-

c=IN IP4 10.100.69.2

b=AS:64

t=0 0

m=audio 2050 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 180 Ringing

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427

Record-Route: <sip:10.100.69.1:6001;lr;transport=tls>

Record-Route: <sip:10.100.69.1:5060;lr>

Server: Avaya CM/R015x.02.1.016.4

Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls>

P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone>

Supported: timer,replaces,join,histinfo,100rel

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

RSeq: 1

Require: 100rel

Content-Type: application/sdp

Content-Length: 184

v=0

o=- 1 2 IN IP4 10.100.69.1

s=-

c=IN IP4 10.100.69.2

b=AS:64

t=0 0

m=audio 2050 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 180 Ringing

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427

Record-Route: <sip:10.100.69.1:6001;lr;transport=tls>

Record-Route: <sip:10.100.69.1:5060;lr>

Server: Avaya CM/R015x.02.1.016.4

Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls>

P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone>

Supported: timer,replaces,join,histinfo,100rel

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

RSeq: 1

Require: 100rel

Content-Type: application/sdp

Content-Length: 184

 

v=0

o=- 1 2 IN IP4 10.100.69.1

s=-

c=IN IP4 10.100.69.2

b=AS:64

t=0 0

m=audio 2050 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 180 Ringing

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=8082a359cf52df110534b84149600

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;received=10.115.64.98;branch=z9hG4bK224d7919a427

Record-Route: <sip:10.100.69.1:6001;lr;transport=tls>

Record-Route: <sip:10.100.69.1:5060;lr>

Server: Avaya CM/R015x.02.1.016.4

Contact: "Polycom" <sip:6012@10.100.69.1:6001;user=phone;transport=tls>

P-Asserted-Identity: "Polycom" <sip:6012@10.100.69.1:6001;user=phone>

Supported: timer,replaces,join,histinfo,100rel

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

RSeq: 1

Require: 100rel

Content-Type: application/sdp

Content-Length: 184

 

v=0

o=- 1 2 IN IP4 10.100.69.1

s=-

c=IN IP4 10.100.69.2

b=AS:64

t=0 0

m=audio 2050 RTP/AVP 18 101

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

 

 

 

Received SIP PDU from ( 10.100.69.1:5060 )

SIP/2.0 504 Server Time-out

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120037123753

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 INVITE

Via: SIP/2.0/UDP 10.115.64.98:5060;psrrposn=1;received=10.115.64.98;branch=z9hG4bK224d7919a427

Server: Avaya CM/R015x.02.1.016.4

Content-Length: 0

 

Sending SIP PDU to ( 10.100.69.1:5060 ) from 5060

ACK sip:6012@10.100.69.1 SIP/2.0

Via: SIP/2.0/UDP 10.115.64.98:5060;branch=z9hG4bK224d7919a427

From: <sip:6010@10.100.69.1>;tag=224d7919a4

To: <sip:6012@10.100.69.1>;tag=471CE0B3F3F312BA810FD40EEAC3794C1270120037123753

Call-ID: 22cf954d-2c91-7989-8019-0002a4035c40@10.115.64.98

CSeq: 27 ACK

 

Proxy-Authorization: Digest username="6010", realm="10.100.69.1", nonce="MTI3MDEyMDAzNjpTREZTZXJ2ZXJTZWNyZXRLZXk6MjA2Nzg0NTQ0Ng==", uri="sip:6012@10.100.69.1", response="7991b5da1f3102c5f03a9a71c41aab3e", algorithm=MD5

Content-Length: 0

Max-Forwards: 70

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Во первых отключите на аддпаке поддержку 100rel чтобы она prack-ами не бросалась.

 

Во вторых у вас сессия вообще не стартует и весь голос идет (если идет) в предответе. Таймаут выходит (там по дефолту не то 20 не то 30 секунд), сессия рвется.

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Во вторых у вас сессия вообще не стартует и весь голос идет (если идет) в предответе. Таймаут выходит (там по дефолту не то 20 не то 30 секунд), сессия рвется.

Вы, бы не могли подсказать возможные пути решения данной проблемы?

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У вас 2 проблемы:

 

SIP/2.0 422 Session Interval Too Small и PRACK'и. Звонок у вас вообще не состоялся.

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да чёто не видно что трубку то подняли и пошёл разговор.

Разговор там точно был, но не длительный.

 

Т.е. на сколько я понял, то нужно отключить 100rel и увеличить время сессии Min-SE до 3600?

Дело в том, что у меня ограниченный доступ к оборудованию и хотелось бы определиться с багами.

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да чёто не видно что трубку то подняли и пошёл разговор.

Разговор там точно был, но не длительный.

 

Т.е. на сколько я понял, то нужно отключить 100rel и увеличить время сессии Min-SE до 3600?

Дело в том, что у меня ограниченный доступ к оборудованию и хотелось бы определиться с багами.

 

Да. После этого надо обязательно увидеть 200 OK & ACK в ответ на INVITE. Без этого доступ к оборудованию придется получать еще раз.

 

.

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Спасибо за советы. Сегодня поеду проверять.

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Отключил 100rel, увеличил Min-SE до 3600 и все заработало!

Всем спасибо за помощь.

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