Jump to content

OVH Trunk - no sound


Jeremy Salmon
 Share

Recommended Posts

Hi All,

 

I have a trunk to OVH SIP Provider (ovh.fr).

 

Trunk is registered but when I place a call, I ear just the beginning of the first ring and after nothing else ...

 

I played with trunk option, sip replacement list, ... without success

 

snom ONE IP : 192.168.1.13

snom320 IP : 192.168.1.12

OVH sip server : 91.121.129.17

Called number : 0972101112

 

Here my SIP log :

 

=====

[5] 2011/12/28 10:28:32: Last message repeated 6 times

[5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:

INVITE sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1

X-Serialnumber: 000413318D7F

P-Key-Flags: keys="3"

User-Agent: snom320/8.4.18

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 469490837 469490837 IN IP4 192.168.1.12

s=call

c=IN IP4 192.168.1.12

t=0 0

m=audio 62652 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:fpsUzc8Bo1L9ecyoUesYEHcSpeeJL9NY265vLFVc

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

[5] 2011/12/28 10:28:32: Last message repeated 2 times

[5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport=2081

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 1 INVITE

Content-Length: 0

 

[5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport=2081

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 1 INVITE

User-Agent: snom-PBX/2011-4.2.0.3981

WWW-Authenticate: Digest realm="societe4.topsystem.be",nonce="1a6d0303a5486a12117e695eaaaf8663",domain="sip:0972101112@societe4.topsystem.be;user=phone",algorithm=MD5

Content-Length: 0

 

[5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:

ACK sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-gcbxcf2bivmr;rport

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

 

[5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:

INVITE sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 2 INVITE

Max-Forwards: 70

Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1

X-Serialnumber: 000413318D7F

P-Key-Flags: keys="3"

User-Agent: snom320/8.4.18

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

Authorization: Digest username="400",realm="societe4.topsystem.be",nonce="1a6d0303a5486a12117e695eaaaf8663",uri="sip:0972101112@societe4.topsystem.be;user=phone",response="1f49712e3eb260cf6b2c006d031e9218",algorithm=MD5

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 522

 

v=0

o=root 469490837 469490837 IN IP4 192.168.1.12

s=call

c=IN IP4 192.168.1.12

t=0 0

m=audio 62652 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:fpsUzc8Bo1L9ecyoUesYEHcSpeeJL9NY265vLFVc

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

[5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 2 INVITE

Content-Length: 0

 

[5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060:

INVITE sip:0972101112@91.121.129.17;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8e16178196f7d0955897860527ca4960;rport

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>

Call-ID: 0f4287ed@pbx

CSeq: 12053 INVITE

Max-Forwards: 70

Contact: <sip:0033184190197@192.168.1.13:5060;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

Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes

Content-Type: application/sdp

Content-Length: 323

 

v=0

o=- 801070725 801070725 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 60346 RTP/AVP 0 8 3 2 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:30

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

a=sendrecv

[5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:

SIP/2.0 183 Session Progress

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 2 INVITE

Contact: <sip:400@192.168.1.13:5060;transport=tcp>

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: 298

 

v=0

o=- 1825812257 1825812257 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 57150 RTP/AVP 0 8 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/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] 2011/12/28 10:28:32: SIP Rx udp:91.121.129.17:5060:

SIP/2.0 407 authentication required

Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8e16178196f7d0955897860527ca4960

Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a364c-25053b736

Call-ID: 0f4287ed@pbx

CSeq: 12053 INVITE

Contact: <sip:0972101112@41.141.84.105:5060;user=phone>

Proxy-Authenticate: Digest realm="sip.ovh.net", nonce="001a360d6b5d399f4797a37b635627ac", opaque="0010abd225acd41", stale=false, algorithm=MD5

server: Cirpack/v4.42j (gw_sip)

Allow: UPDATE, REFER, INFO

Content-Length: 0

 

[5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060:

ACK sip:0972101112@91.121.129.17;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8e16178196f7d0955897860527ca4960;rport

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a364c-25053b736

Call-ID: 0f4287ed@pbx

CSeq: 12053 ACK

Max-Forwards: 70

Content-Length: 0

 

[5] 2011/12/28 10:28:32: SIP Tx udp:91.121.129.17:5060:

INVITE sip:0972101112@91.121.129.17;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>

Call-ID: 0f4287ed@pbx

CSeq: 12054 INVITE

Max-Forwards: 70

Contact: <sip:0033184190197@192.168.1.13:5060;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

Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes

Proxy-Authorization: Digest realm="sip.ovh.net",nonce="001a360d6b5d399f4797a37b635627ac",response="16890c1b4a1beaf825a0f2e3fd1234fc",username="0033184190197",uri="sip:0972101112@91.121.129.17;user=phone",opaque="0010abd225acd41",algorithm=MD5

Content-Type: application/sdp

Content-Length: 323

 

v=0

o=- 801070725 801070725 IN IP4 192.168.1.13

s=-

c=IN IP4 192.168.1.13

t=0 0

m=audio 60346 RTP/AVP 0 8 3 2 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:3 gsm/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:30

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

a=sendrecv

[5] 2011/12/28 10:28:32: SIP Rx tcp:192.168.1.12:2081:

PRACK sip:400@192.168.1.13:5060;transport=tcp SIP/2.0

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-ckw8xvotqm55;rport

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 3 PRACK

Max-Forwards: 70

Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1

RAck: 1 2 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

 

[5] 2011/12/28 10:28:32: SIP Tx tcp:192.168.1.12:2081:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-ckw8xvotqm55;rport=2081

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 3 PRACK

Contact: <sip:400@192.168.1.13:5060;transport=tcp>

User-Agent: snom-PBX/2011-4.2.0.3981

Content-Length: 0

 

[5] 2011/12/28 10:28:32: SIP Rx udp:91.121.129.17:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>

Call-ID: 0f4287ed@pbx

CSeq: 12054 INVITE

Contact: <sip:41.141.84.105:5060>

server: Cirpack/v4.42j (gw_sip)

Allow: UPDATE, REFER, INFO

Content-Length: 0

 

[5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

Call-ID: 0f4287ed@pbx

CSeq: 12054 INVITE

Contact: <sip:41.141.84.105:5060>

server: Cirpack/v4.42j (gw_sip)

Allow: UPDATE, REFER, INFO

Content-Length: 0

 

[5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060:

SIP/2.0 183 Media change

Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

Call-ID: 0f4287ed@pbx

CSeq: 12054 INVITE

Contact: <sip:41.141.84.105:5060>

server: Cirpack/v4.42j (gw_sip)

Allow: UPDATE, REFER, INFO

Content-Type: application/sdp

Content-Length: 262

 

v=0

o=cp10 132506811328 132506811330 IN IP4 192.168.1.1

s=SIP Call

c=IN IP4 192.168.1.1

t=0 0

m=audio 7072 RTP/AVP 0 8 101

b=AS:82

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

[5] 2011/12/28 10:28:33: SIP Rx udp:91.121.129.17:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-05b06979574af555d61e176134a2f476

Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

From: <sip:0033184190197@91.121.129.17>;tag=1452189312

To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-07824-001a1068-6632914c5

Call-ID: ad9d1f30@pbx

CSeq: 18918 INVITE

Contact: <sip:41.141.84.105:5060>

p-asserted-identity: <sip:0972101112@91.121.129.17;user=phone>

server: Cirpack/v4.42j (gw_sip)

Allow: UPDATE, REFER, INFO

Content-Type: application/sdp

Content-Length: 262

 

v=0

o=cp10 132506808498 132506808500 IN IP4 192.168.1.1

s=SIP Call

c=IN IP4 192.168.1.1

t=0 0

m=audio 7070 RTP/AVP 0 8 101

b=AS:82

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

a=sendrecv

[5] 2011/12/28 10:28:34: SIP Rx tcp:192.168.1.12:2081:

CANCEL sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 2 CANCEL

Max-Forwards: 70

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

Proxy-Require: buttons

Content-Length: 0

 

[5] 2011/12/28 10:28:34: SIP Tx tcp:192.168.1.12:2081:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 2 CANCEL

Contact: <sip:400@192.168.1.13:5060;transport=tcp>

User-Agent: snom-PBX/2011-4.2.0.3981

Content-Length: 0

 

[5] 2011/12/28 10:28:34: SIP Tx tcp:192.168.1.12:2081:

SIP/2.0 487 Request Terminated

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport=2081

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 2 INVITE

Contact: <sip:400@192.168.1.13:5060;transport=tcp>

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

 

[5] 2011/12/28 10:28:34: SIP Tx udp:91.121.129.17:5060:

CANCEL sip:0972101112@91.121.129.17;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>

Call-ID: 0f4287ed@pbx

CSeq: 12054 CANCEL

Max-Forwards: 70

Remote-Party-ID: "Standardiste" <sip:400@societe4.topsystem.be;user=phone>;party=calling;screen=yes

Content-Length: 0

 

[5] 2011/12/28 10:28:34: SIP Rx tcp:192.168.1.12:2081:

ACK sip:0972101112@societe4.topsystem.be;user=phone SIP/2.0

Via: SIP/2.0/TCP 192.168.1.12:2081;branch=z9hG4bK-50qfvr8zu02i;rport

From: "Standardiste" <sip:400@societe4.topsystem.be>;tag=ffbkc6h3xn

To: <sip:0972101112@societe4.topsystem.be;user=phone>;tag=f2e4c5a710

Call-ID: 3c267cc8ba57-s7m7z0rilz9z

CSeq: 2 ACK

Max-Forwards: 70

Contact: <sip:400@192.168.1.12:2081;transport=tcp;line=syoxynt0>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

 

[5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>

Call-ID: 0f4287ed@pbx

CSeq: 12054 CANCEL

server: Cirpack/v4.42j (gw_sip)

Content-Length: 0

 

[5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060:

SIP/2.0 487 Session canceled

Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

Call-ID: 0f4287ed@pbx

CSeq: 12054 INVITE

Contact: <sip:41.141.84.105:5060>

server: Cirpack/v4.42j (gw_sip)

Allow: UPDATE, REFER, INFO

Content-Length: 0

 

[5] 2011/12/28 10:28:34: SIP Tx udp:192.168.1.1:5060:

ACK sip:41.141.84.105:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport

Route: <sip:siproxd@192.168.1.1:5060;lr>

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

Call-ID: 0f4287ed@pbx

CSeq: 12054 ACK

Max-Forwards: 70

Contact: <sip:0033184190197@192.168.1.13:5060;transport=udp>

Remote-Party-ID: <sip:0033184190197@91.121.129.17>;party=calling;screen=yes

Content-Length: 0

 

[5] 2011/12/28 10:28:34: INVITE Response 487 Session canceled: Terminate 0f4287ed@pbx

[5] 2011/12/28 10:28:34: SIP Rx udp:91.121.129.17:5060:

ACK sip:192.168.1.13:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bKb094afeb726b070e8ab8f4e7222edbb0

Via: SIP/2.0/UDP 41.141.84.105:5060;branch=z9hG4bKc0e36b0d24e61351467780d2975f9369

Via: SIP/2.0/UDP 192.168.1.13:5060;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39;rport

Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

Call-ID: 0f4287ed@pbx

CSeq: 12054 ACK

Contact: <sip:0033184190197@41.141.84.105>

max-forwards: 68

remote-party-id: <sip:0033184190197@91.121.129.17>;party=calling;screen=yes

Content-Length: 0

 

[5] 2011/12/28 10:28:35: SIP Rx udp:91.121.129.17:5060:

SIP/2.0 487 Session canceled

Via: SIP/2.0/UDP 192.168.1.13:5060;rport;branch=z9hG4bK-8b328816c83c0d8cfd89225109466e39

Record-Route: <sip:siproxd@192.168.1.1:5060;lr>

From: <sip:0033184190197@91.121.129.17>;tag=1119664557

To: <sip:0972101112@91.121.129.17;user=phone>;tag=00-08093-001a366d-70574b090

Call-ID: 0f4287ed@pbx

CSeq: 12054 INVITE

Contact: <sip:41.141.84.105:5060>

server: Cirpack/v4.42j (gw_sip)

Allow: UPDATE, REFER, INFO

Content-Length: 0

=====

 

 

 

Any ideas ?

 

I thing sound disappear just after the "183 Media Change" ....

 

Thanks in advance,

 

Jeremy

Link to comment
Share on other sites

It's a DLINK Dir300. But I tried with two others. :(

 

 

 

Here my Trunk settings:

 

# Trunk 3 in domain societe4.topsystem.be

Name: OVH_TRUNK

Type: register

To: sip

RegPass: ********

Direction:

Disabled:

Global: false

Display:

RegAccount: 0033184190197

RegRegistrar: sip.ovh.net

RegKeep:

RegUser: 0033184190197

Icid:

Require:

OutboundProxy: sip.ovh.net

Ani:

DialExtension:

Prefix:

Trusted: false

AcceptRedirect: false

RfcRtp: false

Analog: false

SendEmail:

UseUuid: false

Ring180: false

Failover: never

Privacy: false

Glob:

RequestTimeout:

Codecs:

CodecLock: true

Expires: 3600

FromUser:

Tel: true

TranscodeDtmf: false

AssociatedAddresses:

InterOffice: false

DialPlan:

Colines:

DialogPermission:

Link to comment
Share on other sites

For info firmware version of the DIR300 is ME_2.01.

 

Under "Advanced Setup", "NAT" I have ALG. By default it's activated.

 

I turned off "SIP Enabled". It seem to work fine.

 

Thanks for your patience !!!

 

Just a last question : why SIP Provider IPPI.fr worked and OVH no ? For info OVH have a CIRPACK and Ippi an Asterisk.

 

Regards,

 

JS

Link to comment
Share on other sites

The problem with the ALG is that the router manufacturers have to deal with so many different SIP interpretations and even buggy implementations that it becomes impossible to please everyone. SIP was not designed for ALG at all, and you need to do really difficult things to get this done properly. IMHO SIP ALG are not very useful anyway because practically all SIP providers today support a far-end NAT solution (SBC), and there is no need to fix things on the router.

 

Sometimes it is even counter productive. :)

Link to comment
Share on other sites

Join the conversation

You can post now and register later. If you have an account, sign in now to post with your account.
Note: Your post will require moderator approval before it will be visible.

Guest
Reply to this topic...

×   Pasted as rich text.   Paste as plain text instead

  Only 75 emoji are allowed.

×   Your link has been automatically embedded.   Display as a link instead

×   Your previous content has been restored.   Clear editor

×   You cannot paste images directly. Upload or insert images from URL.

 Share

×
×
  • Create New...