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marc

By marc
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marc Newbie

marc

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Posted

Bonjour,

 

Je me suis mis il y a peu de temps à snom, j'ai réussi à comprendre, non sans quelques difficultés, la globalité des fonctions les plus courantes grâce au manuel et au wiki.

J'ai voulu tester un peu les trunks à l'aide de deux PC, donc deux serveurs snoms indépendants. Un tourne sous UBUNTU et l'autre WINDOWS (si ça peut aider).

J'ai donc cherché un peu et j'ai lu que dans ce cas il vaut mieux utiliser (corrigez moi si je me trompe) un SIP gateway, et que c’était très facile à configurer. Or le manuel et le wiki ne sont pas particulièrement(à mon avis) très clair au sujet des trunks.

J'ai donc essayé différents réglages (sans grand résultats) et je suis arrivé face à un problème. Le Dialplan reconnait bien l'appel et le dirige vers le bon trunk, un paquet est envoyé vers l'autre serveur (vérifié avec wireshark). Mais rien ne se passe. Au niveau du téléphone il n'y à pas de tonalité. Est-ce simplement un problème de configuration du serveur en face ?

 

Le tout est installé sur un réseau local, avec DHCP.

 

Si quelqu'un aurait la gentillesse de m'éclairer sur la configuration des trunks (plus précisément les SIP gateways), ça serait sympa ^^.

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marc Newbie

marc

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L'intérêt de la config ? Pas grand en effet, j'attend du matériel (du type gateway patton, d'autres tél.). Donc en attendant j'essaye de comprendre l'utilisation, et surtout la configuration des trunks.

 

Je suis sur du snom one free, avec quelques extensions de chaque côté, des auto-attentes et des services flags.

Dans ce cas j'essaye de contacter une auto-attente à partir d'un téléphone.

 

Au niveau des logs (j'en met peut-être trop):

 

 

[5] 2011/04/20 15:30:29: SIP Rx tls:192.168.1.5:4380:

INVITE sip:15600@pbx.fr;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.5:4380;branch=z9hG4bK-gdf0fkwvsz63;rport

From: "un UN" <sip:400@pbx.fr>;tag=ed58jb0pl6

To: <sip:15600@pbx.fr;user=phone>

Call-ID: 74d8263c51a2-umwvu8aeo9ai

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.5:4380;transport=tls;line=068beojl>;reg-id=1

X-Serialnumber: 000413455D50

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom821/8.4.18

Accept: application/sdp

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

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 522

 

v=0

o=root 1698489843 1698489843 IN IP4 192.168.1.5

s=call

c=IN IP4 192.168.1.5

t=0 0

m=audio 54908 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:lGM6KWJYNuZp08NlvHZJ+2Ag737+w+59YMRX/M3Z

a=rtpmap:9 g722/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt

a=sendrecv

[8] 2011/04/20 15:30:29: Packet authenticated by transport layer

[9] 2011/04/20 15:30:29: UDP: Opening socket on 0.0.0.0:64318

[9] 2011/04/20 15:30:29: UDP: Opening socket on 0.0.0.0:64319

[9] 2011/04/20 15:30:29: UDP: Opening socket on [::]:64318

[9] 2011/04/20 15:30:29: UDP: Opening socket on [::]:64319

[8] 2011/04/20 15:30:29: Could not find a trunk (2 trunks)

[5] 2011/04/20 15:30:29: SIP Rx tls:192.168.1.5:4380:

INVITE sip:15600@pbx.fr;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.5:4380;branch=z9hG4bK-gdf0fkwvsz63;rport

From: "un UN" <sip:400@pbx.fr>;tag=ed58jb0pl6

To: <sip:15600@pbx.fr;user=phone>

Call-ID: 74d8263c51a2-umwvu8aeo9ai

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.5:4380;transport=tls;line=068beojl>;reg-id=1

X-Serialnumber: 000413455D50

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom821/8.4.18

Accept: application/sdp

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

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 522

 

v=0

o=root 1698489843 1698489843 IN IP4 192.168.1.5

s=call

c=IN IP4 192.168.1.5

t=0 0

m=audio 54908 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:lGM6KWJYNuZp08NlvHZJ+2Ag737+w+59YMRX/M3Z

a=rtpmap:9 g722/8000

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt

a=sendrecv

[9] 2011/04/20 15:30:29: Using outbound proxy sip:192.168.1.5:4380;transport=tls because of flow-label

[9] 2011/04/20 15:30:29: Last message repeated 3 times

[6] 2011/04/20 15:30:29: Received searchRequest, cn=

[5] 2011/04/20 15:30:29: SIP Tx tls:192.168.1.5:4380:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.1.5:4380;branch=z9hG4bK-gdf0fkwvsz63;rport=4380

From: "un UN" <sip:400@pbx.fr>;tag=ed58jb0pl6

To: <sip:15600@pbx.fr;user=phone>;tag=5793247cef

Call-ID: 74d8263c51a2-umwvu8aeo9ai

CSeq: 1 INVITE

Content-Length: 0

 

[7] 2011/04/20 15:30:29: Set packet length to 20

[6] 2011/04/20 15:30:29: Sending RTP for 74d8263c51a2-umwvu8aeo9ai to 192.168.1.5:54908, codec not set yet

[8] 2011/04/20 15:30:29: Call from an user 400

[8] 2011/04/20 15:30:29: To is <sip:15600@pbx.fr;user=phone>, user 0, domain 1

[8] 2011/04/20 15:30:29: From user 400

[8] 2011/04/20 15:30:29: Set the To domain based on From user 400@pbx.fr

[8] 2011/04/20 15:30:29: Call state for call object 3: idle

[7] 2011/04/20 15:30:29: set_codecs: for 74d8263c51a2-umwvu8aeo9ai codecs "", codec_preference count 6

[9] 2011/04/20 15:30:29: Dialplan: Evaluating !^(15|17|18|112)@.*!sip:\1@\r;user=phone!i against 15600@pbx.fr

[9] 2011/04/20 15:30:29: Dialplan: Evaluating !^0([1-9]{8})!sip:0\1@\r;user=phone!i against 15600@pbx.fr

[9] 2011/04/20 15:30:29: Dialplan: Evaluating !^00([1-9][1-9][1-9]{8})!sip:00\1@\r;user=phone!i against 15600@pbx.fr

[9] 2011/04/20 15:30:29: Dialplan: Evaluating !^15([0-9]*)@.*!sip:\1@\r;user=phone!i against 15600@pbx.fr

[5] 2011/04/20 15:30:29: Dialplan "Standard Dialplan": Match 15600@pbx.fr to <sip:600@192.168.1.7;user=phone> on trunk aaa

[8] 2011/04/20 15:30:29: Play audio_moh/noise.wav

[9] 2011/04/20 15:30:29: UDP: Opening socket on 0.0.0.0:62736

[9] 2011/04/20 15:30:29: UDP: Opening socket on 0.0.0.0:62737

[9] 2011/04/20 15:30:29: UDP: Opening socket on [::]:62736

[9] 2011/04/20 15:30:29: UDP: Opening socket on [::]:62737

[7] 2011/04/20 15:30:29: set_codecs: for 301daad6@pbx codecs "", codec_preference count 6

[9] 2011/04/20 15:30:29: update_codecs for 301daad6@pbx: adding codec pcmu/8000 to available list

[9] 2011/04/20 15:30:29: update_codecs for 301daad6@pbx: adding codec pcma/8000 to available list

[9] 2011/04/20 15:30:29: update_codecs for 301daad6@pbx: adding codec g722/8000 to available list

[9] 2011/04/20 15:30:29: update_codecs for 301daad6@pbx: adding codec g726-32/8000 to available list

[9] 2011/04/20 15:30:29: update_codecs for 301daad6@pbx: adding codec gsm/8000 to available list

[9] 2011/04/20 15:30:29: update_codecs for 301daad6@pbx: codec_preference size 6, available codecs size 6

[9] 2011/04/20 15:30:29: Resolve 362: url sip:192.168.1.7

[9] 2011/04/20 15:30:29: Resolve 362: udp 192.168.1.7 5060

[5] 2011/04/20 15:30:29: SIP Tx udp:192.168.1.7:5060:

INVITE sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.3:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.0.3981

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 2051886958 2051886958 IN IP4 192.168.1.3

s=-

c=IN IP4 192.168.1.3

t=0 0

m=audio 62736 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[7] 2011/04/20 15:30:29: Set packet length to 20

[9] 2011/04/20 15:30:29: update_codecs for 74d8263c51a2-umwvu8aeo9ai: adding codec pcmu/8000 to available list

[9] 2011/04/20 15:30:29: update_codecs for 74d8263c51a2-umwvu8aeo9ai: adding codec pcma/8000 to available list

[9] 2011/04/20 15:30:29: update_codecs for 74d8263c51a2-umwvu8aeo9ai: adding codec g722/8000 to available list

[9] 2011/04/20 15:30:29: update_codecs for 74d8263c51a2-umwvu8aeo9ai: adding codec g726-32/8000 to available list

[9] 2011/04/20 15:30:29: update_codecs for 74d8263c51a2-umwvu8aeo9ai: adding codec gsm/8000 to available list

[9] 2011/04/20 15:30:29: update_codecs for 74d8263c51a2-umwvu8aeo9ai: codec_preference size 6, available codecs size 6

[6] 2011/04/20 15:30:29: Codec pcmu/8000 is chosen for call id 74d8263c51a2-umwvu8aeo9ai

[5] 2011/04/20 15:30:29: SIP Tx tls:192.168.1.5:4380:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TLS 192.168.1.5:4380;branch=z9hG4bK-gdf0fkwvsz63;rport=4380

From: "un UN" <sip:400@pbx.fr>;tag=ed58jb0pl6

To: <sip:15600@pbx.fr;user=phone>;tag=5793247cef

Call-ID: 74d8263c51a2-umwvu8aeo9ai

CSeq: 1 INVITE

Contact: <sip:400@192.168.1.3:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.0.3981

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 429

 

v=0

o=- 624984076 624984076 IN IP4 192.168.1.3

s=-

c=IN IP4 192.168.1.3

t=0 0

m=audio 64318 RTP/AVP 0 8 9 2 3 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:V5CJedrjSinoYIlSRtJ1rjuHXatq2qVqSuGpiKKW

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[5] 2011/04/20 15:30:30: SIP Tr udp:192.168.1.7:5060:

INVITE sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.3:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.0.3981

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 2051886958 2051886958 IN IP4 192.168.1.3

s=-

c=IN IP4 192.168.1.3

t=0 0

m=audio 62736 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[5] 2011/04/20 15:30:30: SIP Rx tls:192.168.1.5:4380:

PRACK sip:400@192.168.1.3:5061;transport=tls SIP/2.0

Via: SIP/2.0/TLS 192.168.1.5:4380;branch=z9hG4bK-ywcqgz69w4ul;rport

From: "un UN" <sip:400@pbx.fr>;tag=ed58jb0pl6

To: <sip:15600@pbx.fr;user=phone>;tag=5793247cef

Call-ID: 74d8263c51a2-umwvu8aeo9ai

CSeq: 2 PRACK

Max-Forwards: 70

Contact: <sip:400@192.168.1.5:4380;transport=tls;line=068beojl>;reg-id=1

RAck: 1 1 INVITE

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

Allow-Events: talk, hold, refer, call-info

Proxy-Require: buttons

Content-Length: 0

 

[8] 2011/04/20 15:30:30: Packet authenticated by transport layer

[5] 2011/04/20 15:30:30: SIP Tx tls:192.168.1.5:4380:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.5:4380;branch=z9hG4bK-ywcqgz69w4ul;rport=4380

From: "un UN" <sip:400@pbx.fr>;tag=ed58jb0pl6

To: <sip:15600@pbx.fr;user=phone>;tag=5793247cef

Call-ID: 74d8263c51a2-umwvu8aeo9ai

CSeq: 2 PRACK

Contact: <sip:400@192.168.1.3:5061;transport=tls>

User-Agent: snom-PBX/2011-4.2.0.3981

Content-Length: 0

 

[5] 2011/04/20 15:30:31: SIP Tr udp:192.168.1.7:5060:

INVITE sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.3:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.0.3981

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 2051886958 2051886958 IN IP4 192.168.1.3

s=-

c=IN IP4 192.168.1.3

t=0 0

m=audio 62736 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[5] 2011/04/20 15:30:32: SIP Rx tls:192.168.1.5:4380:

CANCEL sip:15600@pbx.fr;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.5:4380;branch=z9hG4bK-gdf0fkwvsz63;rport

From: "un UN" <sip:400@pbx.fr>;tag=ed58jb0pl6

To: <sip:15600@pbx.fr;user=phone>

Call-ID: 74d8263c51a2-umwvu8aeo9ai

CSeq: 1 CANCEL

Max-Forwards: 70

Reason: SIP;cause=487;text="Request terminated by user"

Proxy-Require: buttons

Content-Length: 0

 

[8] 2011/04/20 15:30:32: Packet authenticated by transport layer

[5] 2011/04/20 15:30:32: SIP Tx tls:192.168.1.5:4380:

SIP/2.0 200 Ok

Via: SIP/2.0/TLS 192.168.1.5:4380;branch=z9hG4bK-gdf0fkwvsz63;rport=4380

From: "un UN" <sip:400@pbx.fr>;tag=ed58jb0pl6

To: <sip:15600@pbx.fr;user=phone>;tag=5793247cef

Call-ID: 74d8263c51a2-umwvu8aeo9ai

CSeq: 1 CANCEL

Contact: <sip:400@192.168.1.3:5061;transport=tls>

User-Agent: snom-PBX/2011-4.2.0.3981

Content-Length: 0

 

[5] 2011/04/20 15:30:32: SIP Tx tls:192.168.1.5:4380:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/TLS 192.168.1.5:4380;branch=z9hG4bK-gdf0fkwvsz63;rport=4380

From: "un UN" <sip:400@pbx.fr>;tag=ed58jb0pl6

To: <sip:15600@pbx.fr;user=phone>;tag=5793247cef

Call-ID: 74d8263c51a2-umwvu8aeo9ai

CSeq: 1 INVITE

Contact: <sip:400@192.168.1.3:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.0.3981

Content-Length: 0

 

[9] 2011/04/20 15:30:32: Resolve 367: udp 192.168.1.7 5060

[5] 2011/04/20 15:30:32: SIP Tx udp:192.168.1.7:5060:

CANCEL sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 CANCEL

Max-Forwards: 70

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Length: 0

 

[9] 2011/04/20 15:30:32: Remote site 192.168.1.5 closed the connection

[5] 2011/04/20 15:30:33: SIP Rx tls:192.168.1.5:4380:

ACK sip:15600@pbx.fr;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.1.5:4380;branch=z9hG4bK-gdf0fkwvsz63;rport

From: "un UN" <sip:400@pbx.fr>;tag=ed58jb0pl6

To: <sip:15600@pbx.fr;user=phone>;tag=5793247cef

Call-ID: 74d8263c51a2-umwvu8aeo9ai

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:400@192.168.1.5:4380;transport=tls;line=068beojl>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

 

[8] 2011/04/20 15:30:33: Packet authenticated by transport layer

[8] 2011/04/20 15:30:33: Hangup: Call 4 not found

[5] 2011/04/20 15:30:33: SIP Tr udp:192.168.1.7:5060:

INVITE sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.3:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.0.3981

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 2051886958 2051886958 IN IP4 192.168.1.3

s=-

c=IN IP4 192.168.1.3

t=0 0

m=audio 62736 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[5] 2011/04/20 15:30:33: SIP Tr udp:192.168.1.7:5060:

CANCEL sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 CANCEL

Max-Forwards: 70

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Length: 0

 

[5] 2011/04/20 15:30:37: Last message repeated 3 times

[5] 2011/04/20 15:30:37: SIP Tr udp:192.168.1.7:5060:

INVITE sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.3:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.0.3981

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 2051886958 2051886958 IN IP4 192.168.1.3

s=-

c=IN IP4 192.168.1.3

t=0 0

m=audio 62736 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[5] 2011/04/20 15:30:40: SIP Tr udp:192.168.1.7:5060:

CANCEL sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 CANCEL

Max-Forwards: 70

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Length: 0

 

[5] 2011/04/20 15:30:45: Last message repeated 2 times

[8] 2011/04/20 15:30:45: DNS: A * expired

[8] 2011/04/20 15:30:45: DNS: AAAA * expired

[5] 2011/04/20 15:30:45: SIP Tr udp:192.168.1.7:5060:

INVITE sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.3:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.0.3981

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 2051886958 2051886958 IN IP4 192.168.1.3

s=-

c=IN IP4 192.168.1.3

t=0 0

m=audio 62736 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[5] 2011/04/20 15:30:48: SIP Tr udp:192.168.1.7:5060:

CANCEL sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 CANCEL

Max-Forwards: 70

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Length: 0

 

[5] 2011/04/20 15:31:01: Last message repeated 4 times

[5] 2011/04/20 15:31:01: SIP Tr udp:192.168.1.7:5060:

INVITE sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.3:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.0.3981

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Type: application/sdp

Content-Length: 335

 

v=0

o=- 2051886958 2051886958 IN IP4 192.168.1.3

s=-

c=IN IP4 192.168.1.3

t=0 0

m=audio 62736 RTP/AVP 0 8 9 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[5] 2011/04/20 15:31:04: SIP Tr udp:192.168.1.7:5060:

CANCEL sip:600@192.168.1.7;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK-46a6a9426fd716efca588ed58cb3af18;rport

From: "un UN" <sip:400@pbx.fr>;tag=1964218684

To: <sip:600@192.168.1.7;user=phone>

Call-ID: 301daad6@pbx

CSeq: 10083 CANCEL

Max-Forwards: 70

P-Asserted-Identity: "A" <sip:400@pbx.fr>

Content-Length: 0

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marc Newbie

marc

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Posted

Au niveau du manuel, c'est justement la partie que je ne trouvais pas très claire ^^, sinon merci pour le sujet qui à l'air intéressant, j'était passé à coté en cherchant. Je vais voir ce que je peu faire avec ça.

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marc Newbie

marc

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Suite à a modification des trunk selon l'exemple donné dans le sujet conseillé j'ai une erreur "408 Request Timeout (Registration failed, retry after 60 seconds)", qui se produit durant l'enregistrement des deux serveurs.

D'où cela peut-il venir ?

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marc Newbie

marc

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  • Author
Posted

Ouais ben du coup c'est pas exactement ce que je voulais faire :P

À la base je voulais faire ça pour étudier un peu les gateway en prévision de l’installation d'une passerelle patton, à priori une 4112. Quelqu'un à-t-il déjà manipulé ce type de matériel, est-ce facile à mettre en place ?

 

edit : Au niveau de l'extension j'ai rien fait de spécial. Il y quelque chose de plus à configurer par rapport à une extension classique ?

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marc Newbie

marc

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  • Author
Posted

bon, j'ai trouvé le "problème". J'ai pensé que j'avais enregistré une licence sur le deuxième seveur, or il se trouve que non (j'ai honte :( ), donc snom refusait l'enregistrement de l'autre serveur. Du coup tout marche, la config avec des gateways aussi ... désolé pour le dérangement, mais merci encore pour les infos pour la config de la patton.

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  • 2 weeks later...
oge Newbie

oge

Posted
Posted

Bonsoir,

 

Le trunk multi-sites sur snomONE fonctionne de la maniàre suivante ; pour mettre en réseau 2 snomONE

1/ Créer un SIP Trunk (Sip Gateway) 1Vers2 sur le snomONE1

1.1/ Domain: Adresse IP du snomONE2

1.2/ Proxy: Adresse IP du snomONE2

1.3/ Valider la case Inter-sites trunk

1.4/Explicitly list addresses for inbound traffic:Addresse IP du snomONE2

2/ Créer un SIP Trunk (Sip Gateway) 2Vers1 sur le snomONE2

1.1/ Domain: Adresse IP du snomONE1

1.2/ Proxy: Adresse IP du snomONE1

1.3/ Valider la case Inter-sites trunk

1.4/Explicitly list addresses for inbound traffic:Addresse IP du snomONE1

 

Ensuite, il reste a faire un plan de numérotation cohérent

Abonnés du snomONE1: 100 à 199 par exemple

Abonnés du snomONE2: 200 à 299 par exemple

 

Dans le plan Num du snomONE1

Pref Trunk Pattern Replacement

100 1vers2 2* 2* (ou 2xx 2xx)

 

Dans le plan Num du snomONE2

Pref Trunk Pattern Replacement

100 2vers1 1* 1*

 

Et voilà. Associer un trunk du snomONE1 à une extension du snomONE2 comme discuté dans vos postes peut fonctionner mais ce n'est pas la méthode idéale

 

B)

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