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Односторонняя слышимость между Cisco и Avaya Односторонняя слышимость между IP-телефонами Cisco и Avaya

Столкнулся с такой проблемой.

Есть два IP-телефона Cisco 7940 и Avaya 4602SW+IP. Данные телефонные аппараты регистрируются в свою очередь на Asreske.

Проблема в следующем:

При звонке с IP-телефона Cisco 7940 на IP-телефон Avaya 4602SW+IP, абонент сидящий за телефоном Avaya не слышит абонента сидящего за телефоном Cisco, при этом абонента телефона Cisco слышит абонента телефона Avaya.

При звонке с телефонного аппарата Avaya на Cisco голос идет нормально в две стороны.

При звонке на Avaya с другого IP-телефона Cisci 7911 голос идет нормально в обе стороны.

Телефоны работают по протоколу SIP.

Ниже настройки телефонных аппаратом.

---------------------------------------

Avaya 4602SWS+IP

Прошивка s02d02p2_2_3.bin

 

SET HTTPSRVR "10.1.19.4"

SET HTTPDIR "/avaya/"

SET HTTPPORT 80

#SET AUTH 0

SET SIG 2

#SET VLANTEST 896

SET SYSLANG Russian

SET REGISTERWAIT "3600"

SET SIPDOMAIN "x.x.x.x"

SET SIPPROXYSRVR "x.x.x.x"

SET SIPREGISTRAR "x.x.x.x"

SET SIPPORT "5060"

SET SNTPSRVR "x.x.x.x"

SET GMTOFFSET "6:00"

SET DATESEPARATOR "-"

SET DATETIMEFORMAT "3"

SET DIALWAIT "3"

SET DIALPLAN "981xxxxxxxxxxxxxxx|98[2-9]xxxxxxxxxx|9[1-79]xxxxx|9[0]xx|[1-8]xx"

 

<TEMPLATE MATCH="98.........." TIMEOUT="0"/>

 

<TEMPLATE MATCH="9810*" TIMEOUT="3"/>

 

<TEMPLATE MATCH="9000*" TIMEOUT="3"/>

 

<TEMPLATE MATCH="9......" TIMEOUT="0"/>

 

<TEMPLATE MATCH="0..." TIMEOUT="0"/>

 

<TEMPLATE MATCH="1000" TIMEOUT="0"/>

 

<TEMPLATE MATCH="..." TIMEOUT="0"/>

 

<TEMPLATE MATCH="*" TIMEOUT="3"/>

---------------------------------------

Cisco 7940

image_version: P0S3-8-12-00

proxy1_address: "x.x.x.x"

proxy1_port: "5060"

proxy_register: 1

outbound_proxy: "x.x.x.x"

outbound_proxy_port: "5060"

voip_control_port: "5060"

start_media_port: "10000"

end_media_port: "20000"

sntp_server: "x.x.x.x"; SNTP Server IP Address

sntp_mode : unicast

time_zone: WAST

dst_auto_adjust: 0

date_format : D/M/Y

telnet_level: 2

dial_template: dialplan

call_stats: 1

directory_url: "http://x.x.x.x./directory/directory.xml"; URL for external Directory location

services_url: "http://x.x.x.x/directory/service.xml"

#logo_url: "http://x.x.x.x/enter.bmp"; URL for branding logo to be used on phone display

logo_url: ""

language: RU

---------------------------------------

Ну и следовательно вопрос. Кто с таким сталкивался и как проблема решилась.

Так же, кому интересно, могу выслать дам звонка с односторонней слышимостью.

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start_media_port: "10000"

end_media_port: "20000"

Имхо собака тут порылась. Кошак промахивается портом при передаче rtp на аваю. Попробуйте вообще неуказывать эти параметры или расширить диаппазон.

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start_media_port: "10000"

end_media_port: "20000"

Имхо собака тут порылась. Кошак промахивается портом при передаче rtp на аваю. Попробуйте вообще неуказывать эти параметры или расширить диаппазон.

это, кажись, дефолтные настройки Авайи.

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Из "дефолтных", у аваи, еще встречалось 56000 - 59200 и 4000 - 10000.

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start_media_port: "10000"

end_media_port: "20000"

Имхо собака тут порылась. Кошак промахивается портом при передаче rtp на аваю. Попробуйте вообще неуказывать эти параметры или расширить диаппазон.

Выставил на AVAYA RTP порты ручками

SET RTP_PORT_LOW 10000 - выставляешь начало

SET RTP_PORT_RANGE 10000 - выставляешь диапазон

Ситуация не изменилась.

 

PS:Какие значения по дифолту у AVAYA я не знаю(((

PS:Вот пример настройки портов

## UDP Minimum Port Value

## Specifies the lower limit of the UDP port range

## to be used by RTP/RTCP or SRTP/SRTCP connections.

## (1024 -65503).

## Note : This setting is applicable for 1603 SIP phones also.

## SET RTP_PORT_LOW 5004

##

## UDP Port Range

## Specifies the range or number of UDP ports

## available for RTP/RTCP or SRTP/SRTCP connections.

## This value is added to RTP_PORT_LOW to determine

## the upper limit of the UDP port range (32-64511).

## Note : This setting is applicable for 1603 SIP phones also.

## SET RTP_PORT_RANGE 40

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YagmothAgenda

Давайте трассировку проблемного звонка.

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YagmothAgenda

Давайте трассировку проблемного звонка.

Могу дать дамп проблемного звонка.

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А приоритет используемых кодеков одинаков? Не может быть тут проблема? НАТ есть между устройствами?

Edited by shader

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А приоритет используемых кодеков одинаков? Не может быть тут проблема? НАТ есть между станциями?

НАТа нету. Ребята находятся в одной сети. Есть дебаг звонка, есть дамп звонка. В дампе RTP пакеты ходят, но голоса нету.

ЗЫ: Ниже выкладываю дебаг звонка.

<--- SIP read from UDP:10.1.19.218:52662 --->

INVITE sip:507@10.1.19.4 SIP/2.0

Via: SIP/2.0/UDP 10.1.19.218:5060;branch=z9hG4bK26938e6f

From: "508" <sip:508@10.1.19.4>;tag=000b5f05273e17f97b117787-3a309321

To: <sip:507@10.1.19.4>

Call-ID: 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

Max-Forwards: 70

Date: Mon, 17 Feb 2014 07:57:04 GMT

CSeq: 101 INVITE

User-Agent: Cisco-CP7940G/8.0

Contact: <sip:508@10.1.19.218:5060;transport=udp>

Expires: 180

Accept: application/sdp

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE

Remote-Party-ID: "508" <sip:508@10.1.19.4>;party=calling;id-type=subscriber;privacy=off;screen=yes

Supported: replaces,join,norefersub

Content-Length: 274

Content-Type: application/sdp

Content-Disposition: session;handling=optional

 

v=0

o=Cisco-SIPUA 17805 0 IN IP4 10.1.19.218

s=SIP Call

t=0 0

m=audio 18382 RTP/AVP 0 8 18 101

c=IN IP4 10.1.19.218

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

 

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

--- (18 headers 13 lines) ---

Sending to 10.1.19.218 : 5060 (no NAT)

Using INVITE request as basis request - 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

Found peer '508' for '508' from 10.1.19.218:52662

 

<--- Reliably Transmitting (no NAT) to 10.1.19.218:5060 --->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 10.1.19.218:5060;branch=z9hG4bK26938e6f;received=10.1.19.218

From: "508" <sip:508@10.1.19.4>;tag=000b5f05273e17f97b117787-3a309321

To: <sip:507@10.1.19.4>;tag=as164ca2fa

Call-ID: 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

CSeq: 101 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="172e5f06"

Content-Length: 0

 

 

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

Scheduling destruction of SIP dialog '000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218' in 6784 ms (Method: INVITE)

 

<--- SIP read from UDP:10.1.19.218:52663 --->

ACK sip:507@10.1.19.4 SIP/2.0

Via: SIP/2.0/UDP 10.1.19.218:5060;branch=z9hG4bK26938e6f

From: "508" <sip:508@10.1.19.4>;tag=000b5f05273e17f97b117787-3a309321

To: <sip:507@10.1.19.4>;tag=as164ca2fa

Call-ID: 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

Date: Mon, 17 Feb 2014 07:57:04 GMT

CSeq: 101 ACK

Content-Length: 0

 

 

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

--- (8 headers 0 lines) ---

 

<--- SIP read from UDP:10.1.19.218:52664 --->

INVITE sip:507@10.1.19.4 SIP/2.0

Via: SIP/2.0/UDP 10.1.19.218:5060;branch=z9hG4bK1e7731af

From: "508" <sip:508@10.1.19.4>;tag=000b5f05273e17f97b117787-3a309321

To: <sip:507@10.1.19.4>

Call-ID: 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

Max-Forwards: 70

Date: Mon, 17 Feb 2014 07:57:04 GMT

CSeq: 102 INVITE

User-Agent: Cisco-CP7940G/8.0

Contact: <sip:508@10.1.19.218:5060;transport=udp>

Authorization: Digest username="508",realm="asterisk",uri="sip:507@10.1.19.4",response="bcfe970920c50645936310ccdb03f36b",nonce="172e5f06",algorithm=MD5

Expires: 180

Accept: application/sdp

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE

Remote-Party-ID: "508" <sip:508@10.1.19.4>;party=calling;id-type=subscriber;privacy=off;screen=yes

Supported: replaces,join,norefersub

Content-Length: 274

Content-Type: application/sdp

Content-Disposition: session;handling=optional

 

v=0

o=Cisco-SIPUA 17805 0 IN IP4 10.1.19.218

s=SIP Call

t=0 0

m=audio 18382 RTP/AVP 0 8 18 101

c=IN IP4 10.1.19.218

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

 

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

--- (19 headers 13 lines) ---

Sending to 10.1.19.218 : 5060 (no NAT)

Using INVITE request as basis request - 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

Found peer '508' for '508' from 10.1.19.218:52664

Found RTP audio format 0

Found RTP audio format 8

Found RTP audio format 18

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 G729 for ID 18

Found audio description format telephone-event for ID 101

Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/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 10.1.19.218:18382

Looking for 507 in dial_out (domain 10.1.19.4)

list_route: hop: <sip:508@10.1.19.218:5060;transport=udp>

 

<--- Transmitting (no NAT) to 10.1.19.218:5060 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.1.19.218:5060;branch=z9hG4bK1e7731af;received=10.1.19.218

From: "508" <sip:508@10.1.19.4>;tag=000b5f05273e17f97b117787-3a309321

To: <sip:507@10.1.19.4>

Call-ID: 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

CSeq: 102 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Contact: <sip:507@10.1.19.4>

Content-Length: 0

 

 

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

 

<--- Transmitting (no NAT) to 10.1.19.218:5060 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 10.1.19.218:5060;branch=z9hG4bK1e7731af;received=10.1.19.218

From: "508" <sip:508@10.1.19.4>;tag=000b5f05273e17f97b117787-3a309321

To: <sip:507@10.1.19.4>;tag=as588e58cb

Call-ID: 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

CSeq: 102 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Contact: <sip:507@10.1.19.4>

Content-Length: 0

 

 

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

Audio is at 10.1.19.4 port 15806

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 10.1.19.218:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.1.19.218:5060;branch=z9hG4bK1e7731af;received=10.1.19.218

From: "508" <sip:508@10.1.19.4>;tag=000b5f05273e17f97b117787-3a309321

To: <sip:507@10.1.19.4>;tag=as588e58cb

Call-ID: 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

CSeq: 102 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Contact: <sip:507@10.1.19.4>

Content-Type: application/sdp

Content-Length: 253

 

v=0

o=root 105493841 105493841 IN IP4 10.1.19.4

s=Asterisk PBX 1.6.2.13

c=IN IP4 10.1.19.4

t=0 0

m=audio 15806 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=ptime:20

a=sendrecv

 

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

 

<--- SIP read from UDP:10.1.19.218:52665 --->

ACK sip:507@10.1.19.4 SIP/2.0

Via: SIP/2.0/UDP 10.1.19.218:5060;branch=z9hG4bK132a8dc8

From: "508" <sip:508@10.1.19.4>;tag=000b5f05273e17f97b117787-3a309321

To: <sip:507@10.1.19.4>;tag=as588e58cb

Call-ID: 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

Max-Forwards: 70

Date: Mon, 17 Feb 2014 07:57:07 GMT

CSeq: 102 ACK

User-Agent: Cisco-CP7940G/8.0

Authorization: Digest username="508",realm="asterisk",uri="sip:507@10.1.19.4",response="bcfe970920c50645936310ccdb03f36b",nonce="172e5f06",algorithm=MD5

Remote-Party-ID: "508" <sip:508@10.1.19.4>;party=calling;id-type=subscriber;privacy=off;screen=yes

Content-Length: 0

 

 

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

--- (12 headers 0 lines) ---

Scheduling destruction of SIP dialog '000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218' in 6784 ms (Method: ACK)

set_destination: Parsing <sip:508@10.1.19.218:5060;transport=udp> for address/port to send to

set_destination: set destination to 10.1.19.218, port 5060

Reliably Transmitting (no NAT) to 10.1.19.218:5060:

BYE sip:508@10.1.19.218:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.1.19.4:5060;branch=z9hG4bK1812c225;rport

Max-Forwards: 70

From: <sip:507@10.1.19.4>;tag=as588e58cb

To: "508" <sip:508@10.1.19.4>;tag=000b5f05273e17f97b117787-3a309321

Call-ID: 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

CSeq: 102 BYE

User-Agent: Asterisk PBX 1.6.2.13

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

Content-Length: 0

 

 

---

 

<--- SIP read from UDP:10.1.19.218:52666 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.1.19.4:5060;branch=z9hG4bK1812c225;rport

From: <sip:507@10.1.19.4>;tag=as588e58cb

To: "508" <sip:508@10.1.19.4>;tag=000b5f05273e17f97b117787-3a309321

Call-ID: 000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218

Date: Mon, 17 Feb 2014 07:57:11 GMT

CSeq: 102 BYE

Server: Cisco-CP7940G/8.0

Content-Length: 0

RTP-RxStat: Dur=7,Pkt=185,Oct=29600,LatePkt=0,LostPkt=0,AvgJit=0

RTP-TxStat: Dur=4,Pkt=200,Oct=32000

 

 

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

--- (11 headers 0 lines) ---

SIP Response message for INCOMING dialog BYE arrived

Really destroying SIP dialog '000b5f05-273e0021-1273be66-7c3ade0a@10.1.19.218' Method: ACK

Reliably Transmitting (no NAT) to 10.1.19.218:5060:

OPTIONS sip:508@10.1.19.218:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 10.1.19.4:5060;branch=z9hG4bK06125c51;rport

Max-Forwards: 70

From: "asterisk" <sip:asterisk@10.1.19.4>;tag=as176b669e

To: <sip:508@10.1.19.218:5060;transport=udp>

Contact: <sip:asterisk@10.1.19.4>

Call-ID: 0d1f2fbc3e9b266e718007e122e67c7d@10.1.19.4

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.6.2.13

Date: Mon, 17 Feb 2014 07:57:15 GMT

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

Supported: replaces, timer

Content-Length: 0

 

 

---

 

<--- SIP read from UDP:10.1.19.218:52667 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.1.19.4:5060;branch=z9hG4bK06125c51;rport

From: "asterisk" <sip:asterisk@10.1.19.4>;tag=as176b669e

To: <sip:508@10.1.19.218:5060;transport=udp>;tag=000b5f05273e17fa0df658f9-6e03f93f

Call-ID: 0d1f2fbc3e9b266e718007e122e67c7d@10.1.19.4

Date: Mon, 17 Feb 2014 07:57:15 GMT

CSeq: 102 OPTIONS

Server: Cisco-CP7940G/8.0

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE

Accept: application/sdp,multipart/mixed,multipart/alternative

Accept-Encoding: identity

Accept-Language: en

Supported: replaces,join,norefersub

Content-Length: 236

Content-Type: application/sdp

Content-Disposition: session;handling=optional

 

v=0

o=Cisco-SIPUA 10787 0 IN IP4 10.1.19.218

s=SIP Call

t=0 0

m=audio 0 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

 

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

--- (16 headers 11 lines) ---

Really destroying SIP dialog '0d1f2fbc3e9b266e718007e122e67c7d@10.1.19.4' Method: OPTIONS

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Кодек должен быть везде 711, так же прилагаю дамп звонка.

Если интересно могу приложить дамп нормально звонка (вместо Cisco 7940 выступает модель 7911)

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Невидно что со стороны аваи происходит.

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Вот такой вот Debug с Астера. Дебажил по IP адрессам IP-телефонов.

Могу скинуть дамп не удачного звонка.

Edited by YagmothAgenda

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Лучше скинте дебаг звонка с аваи на циску

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Debug звонка с Avaya(507) на Cisco(508)

 

<--- SIP read from UDP:10.1.19.200:5060 --->

INVITE sip:508@10.1.19.4 SIP/2.0

Via: SIP/2.0/UDP 10.1.19.200:5060;branch=z9hG4bKeddfd3298

Max-Forwards: 70

Content-Length: 262

To: 508 <sip:508@10.1.19.4>

From: 507 <sip:507@10.1.19.4>;tag=16ec91b4db7b99e

Call-ID: d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

CSeq: 1666116718 INVITE

Supported: timer

Allow: NOTIFY

Allow: REFER

Allow: OPTIONS

Allow: INVITE

Allow: ACK

Allow: CANCEL

Allow: BYE

Content-Type: application/sdp

Contact: 507 <sip:507@10.1.19.200:5060>

Supported: replaces

User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26

 

v=0

o=MxSIP 0 1004016115 IN IP4 10.1.19.200

s=SIP Call

c=IN IP4 10.1.19.200

t=0 0

m=audio 34008 RTP/AVP 0 8 18 2 127

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:127 telephone-event/8000

a=ptime:20

 

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

--- (20 headers 12 lines) ---

Sending to 10.1.19.200 : 5060 (no NAT)

Using INVITE request as basis request - d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

Found peer '507' for '507' from 10.1.19.200:5060

 

<--- Reliably Transmitting (no NAT) to 10.1.19.200:5060 --->

SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 10.1.19.200:5060;branch=z9hG4bKeddfd3298;received=10.1.19.200

From: 507 <sip:507@10.1.19.4>;tag=16ec91b4db7b99e

To: 508 <sip:508@10.1.19.4>;tag=as3c61cd1b

Call-ID: d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

CSeq: 1666116718 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="68439023"

Content-Length: 0

 

 

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

Scheduling destruction of SIP dialog 'd4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200' in 6400 ms (Method: INVITE)

 

<--- SIP read from UDP:10.1.19.200:5060 --->

ACK sip:508@10.1.19.4 SIP/2.0

Via: SIP/2.0/UDP 10.1.19.200:5060;branch=z9hG4bKeddfd3298

Max-Forwards: 70

Content-Length: 0

To: 508 <sip:508@10.1.19.4>;tag=as3c61cd1b

From: 507 <sip:507@10.1.19.4>;tag=16ec91b4db7b99e

Call-ID: d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

CSeq: 1666116718 ACK

User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26

 

 

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

--- (9 headers 0 lines) ---

 

<--- SIP read from UDP:10.1.19.200:5060 --->

INVITE sip:508@10.1.19.4 SIP/2.0

Via: SIP/2.0/UDP 10.1.19.200:5060;branch=z9hG4bKb2a342000

Max-Forwards: 70

Content-Length: 262

To: 508 <sip:508@10.1.19.4>

From: 507 <sip:507@10.1.19.4>;tag=16ec91b4db7b99e

Call-ID: d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

CSeq: 1666116719 INVITE

Supported: timer

Allow: NOTIFY

Allow: REFER

Allow: OPTIONS

Allow: INVITE

Allow: ACK

Allow: CANCEL

Allow: BYE

Contact: 507 <sip:507@10.1.19.200:5060>

Content-Type: application/sdp

Supported: replaces

Authorization:Digest response="12ba77eb6a533a5e772fd82f97e13625",username="507",realm="asterisk",nonce="68439023",algorithm=MD5,uri="sip:508@10.1.19.4"

User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26

 

v=0

o=MxSIP 0 1004016115 IN IP4 10.1.19.200

s=SIP Call

c=IN IP4 10.1.19.200

t=0 0

m=audio 34008 RTP/AVP 0 8 18 2 127

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:127 telephone-event/8000

a=ptime:20

 

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

--- (21 headers 12 lines) ---

Sending to 10.1.19.200 : 5060 (no NAT)

Using INVITE request as basis request - d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

Found peer '507' for '507' from 10.1.19.200:5060

Found RTP audio format 0

Found RTP audio format 8

Found RTP audio format 18

Found RTP audio format 2

Found RTP audio format 127

Found audio description format PCMU for ID 0

Found audio description format PCMA for ID 8

Found audio description format G729 for ID 18

Found audio description format G726-32 for ID 2

Found audio description format telephone-event for ID 127

Capabilities: us - 0x8 (alaw), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)

Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

Peer audio RTP is at port 10.1.19.200:34008

Looking for 508 in dial_out (domain 10.1.19.4)

list_route: hop: <sip:507@10.1.19.200:5060>

 

<--- Transmitting (no NAT) to 10.1.19.200:5060 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.1.19.200:5060;branch=z9hG4bKb2a342000;received=10.1.19.200

From: 507 <sip:507@10.1.19.4>;tag=16ec91b4db7b99e

To: 508 <sip:508@10.1.19.4>

Call-ID: d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

CSeq: 1666116719 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Require: timer

Session-Expires: 1800;refresher=uas

Contact: <sip:508@10.1.19.4>

Content-Length: 0

 

 

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

 

<--- Transmitting (no NAT) to 10.1.19.200:5060 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 10.1.19.200:5060;branch=z9hG4bKb2a342000;received=10.1.19.200

From: 507 <sip:507@10.1.19.4>;tag=16ec91b4db7b99e

To: 508 <sip:508@10.1.19.4>;tag=as5ca3a165

Call-ID: d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

CSeq: 1666116719 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Require: timer

Session-Expires: 1800;refresher=uas

Contact: <sip:508@10.1.19.4>

Content-Length: 0

 

 

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

Audio is at 10.1.19.4 port 13954

Adding codec 0x8 (alaw) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

 

<--- Reliably Transmitting (no NAT) to 10.1.19.200:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.1.19.200:5060;branch=z9hG4bKb2a342000;received=10.1.19.200

From: 507 <sip:507@10.1.19.4>;tag=16ec91b4db7b99e

To: 508 <sip:508@10.1.19.4>;tag=as5ca3a165

Call-ID: d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

CSeq: 1666116719 INVITE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Require: timer

Session-Expires: 1800;refresher=uas

Contact: <sip:508@10.1.19.4>

Content-Type: application/sdp

Content-Length: 229

 

v=0

o=root 465416510 465416510 IN IP4 10.1.19.4

s=Asterisk PBX 1.6.2.13

c=IN IP4 10.1.19.4

t=0 0

m=audio 13954 RTP/AVP 8 127

a=rtpmap:8 PCMA/8000

a=rtpmap:127 telephone-event/8000

a=fmtp:127 0-16

a=ptime:20

a=sendrecv

 

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

 

<--- SIP read from UDP:10.1.19.200:5060 --->

ACK sip:508@10.1.19.4 SIP/2.0

Via: SIP/2.0/UDP 10.1.19.200:5060;branch=z9hG4bK9f6c8270b

Max-Forwards: 70

Content-Length: 0

To: 508 <sip:508@10.1.19.4>;tag=as5ca3a165

From: 507 <sip:507@10.1.19.4>;tag=16ec91b4db7b99e

Call-ID: d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

CSeq: 1666116719 ACK

Contact: 507 <sip:507@10.1.19.200:5060>

Authorization:Digest response="48ad0e265bd246dc06413e0e4067f80d",username="507",realm="asterisk",nonce="68439023",algorithm=MD5,uri="sip:508@10.1.19.4"

User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26

 

 

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

--- (11 headers 0 lines) ---

 

<--- SIP read from UDP:10.1.19.200:5060 --->

BYE sip:508@10.1.19.4 SIP/2.0

Via: SIP/2.0/UDP 10.1.19.200:5060;branch=z9hG4bKcff361644

Max-Forwards: 70

Content-Length: 0

To: 508 <sip:508@10.1.19.4>;tag=as5ca3a165

From: 507 <sip:507@10.1.19.4>;tag=16ec91b4db7b99e

Call-ID: d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

CSeq: 1666116720 BYE

Supported: timer

Supported: replaces

Authorization:Digest response="cfb2e843fb3b19756e1b4cb4272c6e1d",username="507",realm="asterisk",nonce="68439023",algorithm=MD5,uri="sip:508@10.1.19.4"

User-Agent: Avaya SIP R2.2 Endpoint Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26

 

 

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

--- (12 headers 0 lines) ---

Sending to 10.1.19.200 : 5060 (no NAT)

 

<--- Transmitting (no NAT) to 10.1.19.200:5060 --->

SIP/2.0 200 OK

Via: SIP/2.0/UDP 10.1.19.200:5060;branch=z9hG4bKcff361644;received=10.1.19.200

From: 507 <sip:507@10.1.19.4>;tag=16ec91b4db7b99e

To: 508 <sip:508@10.1.19.4>;tag=as5ca3a165

Call-ID: d4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200

CSeq: 1666116720 BYE

Server: Asterisk PBX 1.6.2.13

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

Supported: replaces, timer

Content-Length: 0

 

 

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

Really destroying SIP dialog 'd4b2164f66b7052b0bf63d0f18ae0a0e@10.1.19.200' Method: BYE

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