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DTMF and Service Provider


ShadowAnt
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Hello!

How I can change DTMF method in PBXnSIP? when I try to dial extension - service provider do not transfer it. here is a log:

 

[7] 2010/01/12 14:32:00: SIP Rx tcp:1.1.7.7:58486:

INVITE sip:+78632370684@1.1.1.5;user=phone SIP/2.0

FROM: <sip:+78632688634100@s-case1c.case.ru;user=phone>;epid=77B47EEE06;tag=a6cc0a75

TO: <sip:+78632370684@1.1.1.5;user=phone>

CSEQ: 113 INVITE

CALL-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK62d377c1

CONTACT: <sip:s-case1c.case.ru:5060;transport=Tcp;maddr=1.1.7.7;ms-opaque=d799212d2afe4476>

CONTENT-LENGTH: 313

SUPPORTED: 100rel

USER-AGENT: RTCC/3.5.0.0 MediationServer

CONTENT-TYPE: application/sdp; charset=utf-8

ALLOW: UPDATE

ALLOW: Ack, Cancel, Bye,Invite

 

v=0

o=- 120 1 IN IP4 1.1.7.7

s=session

c=IN IP4 1.1.7.7

b=CT:1000

t=0 0

m=audio 63818 RTP/AVP 97 101 13 0 8

c=IN IP4 1.1.7.7

a=rtcp:63819

a=label:Audio

a=rtpmap:97 RED/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:13 CN/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=ptime:20

 

[5] 2010/01/12 14:32:00: Identify trunk (IP address and domain match) 2

[7] 2010/01/12 14:32:00: SIP Tx tcp:1.1.7.7:58486:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK62d377c1

From: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06

To: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b

Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

CSeq: 113 INVITE

Content-Length: 0

 

 

[7] 2010/01/12 14:32:00: Set packet length to 20

[6] 2010/01/12 14:32:00: Sending RTP for a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b to 1.1.7.7:63818

[5] 2010/01/12 14:32:00: Dialplan dialplan: Match +78632370684@1.1.1.5 to <sip:78632370684@sipnet.ru;user=phone> on trunk skype

[5] 2010/01/12 14:32:00: Using <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06 as redirect source address

[7] 2010/01/12 14:32:00: SIP Tx udp:212.53.40.40:5060:

INVITE sip:78632370684@sipnet.ru;user=phone SIP/2.0

Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-c79c0a24dde749085584480c72dabb69;rport

From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

To: <sip:78632370684@sipnet.ru;user=phone>

Call-ID: b92b6ef2@pbx

CSeq: 10060 INVITE

Max-Forwards: 70

Contact: <sip:0025879052@1.1.1.5:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.4.0.3201

Related-Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

Content-Type: application/sdp

Content-Length: 323

 

v=0

o=- 131 131 IN IP4 1.1.1.5

s=-

c=IN IP4 1.1.1.5

t=0 0

m=audio 59854 RTP/AVP 0 8 9 18 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2010/01/12 14:32:00: Set packet length to 20

[6] 2010/01/12 14:32:00: Send codec pcmu/8000

[7] 2010/01/12 14:32:00: SIP Tx tcp:1.1.7.7:58486:

SIP/2.0 183 Ringing

Via: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK62d377c1

From: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06

To: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b

Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

CSeq: 113 INVITE

Contact: <sip:Anonymous@1.1.1.5:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.4.0.3201

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 216

 

v=0

o=- 7916 7916 IN IP4 1.1.1.5

s=-

c=IN IP4 1.1.1.5

t=0 0

m=audio 50440 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[7] 2010/01/12 14:32:00: SIP Rx udp:212.53.40.40:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-c79c0a24dde749085584480c72dabb69;rport=58937;received=212.176.115.249

From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

To: <sip:78632370684@sipnet.ru;user=phone>

Call-ID: b92b6ef2@pbx

CSeq: 10060 INVITE

Server: CommuniGatePro/5.2.19

Content-Length: 0

 

 

[7] 2010/01/12 14:32:00: SIP Rx udp:212.53.40.40:5060:

SIP/2.0 401 Authentication required

Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-c79c0a24dde749085584480c72dabb69;rport=58937;received=212.176.115.249

From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

To: <sip:78632370684@sipnet.ru;user=phone>;tag=7A36DD5F

Call-ID: b92b6ef2@pbx

CSeq: 10060 INVITE

WWW-Authenticate: Digest realm="etc.tario.ru",nonce="740E87CA2557EC46E7DB",opaque="opaqueData",qop="auth",algorithm=MD5

Server: CommuniGatePro/5.2.19

Content-Length: 0

 

 

[7] 2010/01/12 14:32:00: SIP Tx udp:212.53.40.40:5060:

ACK sip:78632370684@sipnet.ru;user=phone SIP/2.0

Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-c79c0a24dde749085584480c72dabb69;rport

From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

To: <sip:78632370684@sipnet.ru;user=phone>;tag=7A36DD5F

Call-ID: b92b6ef2@pbx

CSeq: 10060 ACK

Max-Forwards: 70

Content-Length: 0

 

 

[7] 2010/01/12 14:32:00: SIP Tx udp:212.53.40.40:5060:

INVITE sip:78632370684@sipnet.ru;user=phone SIP/2.0

Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-62d178b026c98eea459c2f4874670522;rport

From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

To: <sip:78632370684@sipnet.ru;user=phone>

Call-ID: b92b6ef2@pbx

CSeq: 10061 INVITE

Max-Forwards: 70

Contact: <sip:0025879052@1.1.1.5:5060;transport=udp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.4.0.3201

Related-Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

Authorization: Digest realm="etc.tario.ru",nonce="740E87CA2557EC46E7DB",response="7c0198de271c502a6c46b23a7fb88568",username="0025879052",uri="sip:78632370684@sipnet.ru;user=phone",qop=auth,nc=00000001,cnonce="8dbfdfb6",opaque="opaqueData",algorithm=MD5

Content-Type: application/sdp

Content-Length: 323

 

v=0

o=- 131 131 IN IP4 1.1.1.5

s=-

c=IN IP4 1.1.1.5

t=0 0

m=audio 59854 RTP/AVP 0 8 9 18 2 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=sendrecv

 

[7] 2010/01/12 14:32:00: SIP Rx udp:212.53.40.40:5060:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-62d178b026c98eea459c2f4874670522;rport=58937;received=212.176.115.249

From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

To: <sip:78632370684@sipnet.ru;user=phone>

Call-ID: b92b6ef2@pbx

CSeq: 10061 INVITE

Server: CommuniGatePro/5.2.19

Content-Length: 0

 

 

[7] 2010/01/12 14:32:00: SIP Rx tcp:1.1.7.7:58486:

PRACK sip:Anonymous@1.1.1.5:5060;transport=tcp SIP/2.0

FROM: <sip:+78632688634100@s-case1c.case.ru;user=phone>;epid=77B47EEE06;tag=a6cc0a75

TO: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b

CSEQ: 114 PRACK

CALL-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK9fb71a0

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.5.0.0 MediationServer

RAck: 1 113 INVITE

 

 

[7] 2010/01/12 14:32:00: SIP Tx tcp:1.1.7.7:58486:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK9fb71a0

From: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06

To: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b

Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

CSeq: 114 PRACK

Contact: <sip:Anonymous@1.1.1.5:5060;transport=tcp>

User-Agent: pbxnsip-PBX/3.4.0.3201

Content-Length: 0

 

 

[7] 2010/01/12 14:32:01: SIP Rx udp:212.53.40.40:5060:

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 1.1.1.5:5060;received=212.176.115.249;rport=58937;branch=z9hG4bK-62d178b026c98eea459c2f4874670522

Record-Route: <sip:212.53.35.244:5060;lr>,<sip:197897-192.168.40.71.dialog.cgatepro;lr>

Record-Route: <sip:192.168.40.71:5060;lr>

Record-Route: <sip:212.53.40.40:5060;lr>

From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

To: <sip:78632370684@sipnet.ru;user=phone>;tag=cb33c021-5282988

Call-ID: b92b6ef2@pbx

CSeq: 10061 INVITE

Content-Type: application/sdp

Server: TarioSoftswitch/3.2.11

Content-Length: 231

 

v=0

o=Tario-Softswitch 2749 101 IN IP4 212.53.40.91

s=SIP Call

c=IN IP4 212.53.40.71

t=0 0

m=audio 26520 RTP/AVP 8 97

c=IN IP4 212.53.40.71

a=rtpmap:8 PCMA/8000

a=rtpmap:97 telephone-event/8000

a=fmtp:97 0-15

a=ptime:20

 

[7] 2010/01/12 14:32:01: Set packet length to 20

[6] 2010/01/12 14:32:01: Send codec=pcma/8000 afrer answer

[6] 2010/01/12 14:32:01: Sending RTP for b92b6ef2@pbx#55270 to 212.53.40.71:26520

[7] 2010/01/12 14:32:01: b92b6ef2@pbx#55270: RTP pass-through mode

[7] 2010/01/12 14:32:01: a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b: RTP pass-through mode

[7] 2010/01/12 14:32:01: Cannot pass through on b92b6ef2@pbx#55270, falling back to transcoding

[7] 2010/01/12 14:32:04: SIP Rx udp:212.53.40.40:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 1.1.1.5:5060;received=212.176.115.249;rport=58937;branch=z9hG4bK-62d178b026c98eea459c2f4874670522

Record-Route: <sip:212.53.35.244:5060;lr>,<sip:197897-192.168.40.71.dialog.cgatepro;lr>

Record-Route: <sip:192.168.40.71:5060;lr>

Record-Route: <sip:212.53.40.40:5060;lr>

From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

To: <sip:78632370684@sipnet.ru;user=phone>;tag=cb33c021-5282988

Call-ID: b92b6ef2@pbx

CSeq: 10061 INVITE

Contact: <sip:proc-5282988@212.53.35.244>

Content-Type: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, INFO, OPTIONS

Server: TarioSoftswitch/3.2.11

Content-Length: 231

 

v=0

o=Tario-Softswitch 2749 101 IN IP4 212.53.40.91

s=SIP Call

c=IN IP4 212.53.40.71

t=0 0

m=audio 26520 RTP/AVP 8 97

c=IN IP4 212.53.40.71

a=rtpmap:8 PCMA/8000

a=rtpmap:97 telephone-event/8000

a=fmtp:97 0-15

a=ptime:20

 

[7] 2010/01/12 14:32:04: Call b92b6ef2@pbx#55270: Clear last INVITE

[7] 2010/01/12 14:32:04: Set packet length to 20

[7] 2010/01/12 14:32:04: SIP Tx udp:212.53.40.40:5060:

ACK sip:proc-5282988@212.53.35.244 SIP/2.0

Via: SIP/2.0/UDP 1.1.1.5:5060;branch=z9hG4bK-1d6e4f20cb81caac2bbcd1f8e52266dc;rport

Route: <sip:212.53.40.40:5060;lr>

Route: <sip:192.168.40.71:5060;lr>

Route: <sip:197897-192.168.40.71.dialog.cgatepro;lr>

Route: <sip:212.53.35.244:5060;lr>

From: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

To: <sip:78632370684@sipnet.ru;user=phone>;tag=cb33c021-5282988

Call-ID: b92b6ef2@pbx

CSeq: 10061 ACK

Max-Forwards: 70

Contact: <sip:0025879052@1.1.1.5:5060;transport=udp>

Authorization: Digest realm="etc.tario.ru",nonce="740E87CA2557EC46E7DB",response="aae322d85299e2b96309fcda4830667c",username="0025879052",uri="sip:proc-5282988@212.53.35.244",qop=auth,nc=00000002,cnonce="90704f9a",opaque="opaqueData",algorithm=MD5

Content-Length: 0

 

 

[7] 2010/01/12 14:32:04: Determine pass-through mode after receiving response

[7] 2010/01/12 14:32:04: a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b: RTP pass-through mode

[7] 2010/01/12 14:32:04: Cannot pass through on b92b6ef2@pbx#55270, falling back to transcoding

[7] 2010/01/12 14:32:04: SIP Tx tcp:1.1.7.7:58486:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK62d377c1

From: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06

To: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b

Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

CSeq: 113 INVITE

Contact: <sip:Anonymous@1.1.1.5:5060;transport=tcp>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

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

Accept: application/sdp

User-Agent: pbxnsip-PBX/3.4.0.3201

Content-Type: application/sdp

Content-Length: 216

 

v=0

o=- 7916 7916 IN IP4 1.1.1.5

s=-

c=IN IP4 1.1.1.5

t=0 0

m=audio 50440 RTP/AVP 0 8 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

[7] 2010/01/12 14:32:04: a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b: RTP pass-through mode

[7] 2010/01/12 14:32:04: SIP Rx tcp:1.1.7.7:58486:

ACK sip:Anonymous@1.1.1.5:5060;transport=tcp SIP/2.0

FROM: <sip:+78632688634100@s-case1c.case.ru;user=phone>;epid=77B47EEE06;tag=a6cc0a75

TO: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b

CSEQ: 113 ACK

CALL-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 1.1.7.7:58486;branch=z9hG4bK7e1d4647

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.5.0.0 MediationServer

 

 

[7] 2010/01/12 14:32:04: Cannot pass through on b92b6ef2@pbx#55270, falling back to transcoding

[7] 2010/01/12 14:32:07: SIP Rx udp:1.1.5.149:5060:

SUBSCRIBE sip:1.1.1.5:53242;transport=tcp SIP/2.0

From: <sip:643@1.1.1.5>;tag=1010595-13c4-0-31b-2443

To: <sip:643@1.1.1.5>;tag=3f18c3a09b

Call-ID: 806e43ec-1010595-13c4-0-316-54fd@1.1.1.5

CSeq: 631 SUBSCRIBE

Via: SIP/2.0/UDP 1.1.5.149:5060;branch=z9hG4bK-763a-1cdd4c0-234

Expires: 47

Event: message-summary

Max-Forwards: 70

Supported: replaces,100rel

Accept: application/simple-message-summary

Contact: <sip:643@1.1.5.149:5060;transport=TCP>

Content-Length: 0

 

 

[7] 2010/01/12 14:32:07: SIP Tx udp:1.1.5.149:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 1.1.5.149:5060;branch=z9hG4bK-763a-1cdd4c0-234

From: <sip:643@1.1.1.5>;tag=1010595-13c4-0-31b-2443

To: <sip:643@1.1.1.5>;tag=3f18c3a09b

Call-ID: 806e43ec-1010595-13c4-0-316-54fd@1.1.1.5

CSeq: 631 SUBSCRIBE

Contact: <sip:1.1.1.5:5060;transport=udp>

Expires: 48

Content-Length: 0

 

 

[7] 2010/01/12 14:32:09: Cannot pass through on a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b, falling back to transcoding

[7] 2010/01/12 14:32:09: Received RFC4733 DTMF on codec 101

[5] 2010/01/12 14:32:10: Tuning to new SSRC

[5] 2010/01/12 14:32:14: Last message repeated 2 times

[7] 2010/01/12 14:32:14: SIP Rx udp:212.53.40.40:5060:

BYE sip:0025879052@1.1.1.5:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK480354-kmbdctj;cgp=etc.tario.ru;upaddr=212.53.35.244;rport

Via: SIP/2.0/UDP 212.53.35.244;branch=z9hG4bK263354062

Via: SIP/2.0/TCP 212.53.35.244:50666;branch=z9hG4bK-2c5d5000-5282988

Max-Forwards: 68

From: <sip:78632370684@sipnet.ru;user=phone>;tag=cb33c021-5282988

To: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

Call-ID: b92b6ef2@pbx

CSeq: 35426 BYE

User-Agent: TarioSoftswitch/3.2.11

Content-Length: 0

 

 

[7] 2010/01/12 14:32:14: SIP Tx udp:212.53.40.40:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 212.53.40.40:5060;branch=z9hG4bK480354-kmbdctj;cgp=etc.tario.ru;upaddr=212.53.35.244;rport=5060

Via: SIP/2.0/UDP 212.53.35.244;branch=z9hG4bK263354062

Via: SIP/2.0/TCP 212.53.35.244:50666;branch=z9hG4bK-2c5d5000-5282988

From: <sip:78632370684@sipnet.ru;user=phone>;tag=cb33c021-5282988

To: <sip:99051000001869@sipnet.ru;user=phone>;tag=55270

Call-ID: b92b6ef2@pbx

CSeq: 35426 BYE

Contact: <sip:0025879052@1.1.1.5:5060;transport=udp>

User-Agent: pbxnsip-PBX/3.4.0.3201

RTP-RxStat: Dur=14,Pkt=723,Oct=109636,Underun=458

RTP-TxStat: Dur=10,Pkt=41,Oct=1904

Content-Length: 0

 

 

[7] 2010/01/12 14:32:14: Other Ports: 1

[7] 2010/01/12 14:32:14: Call Port: a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b

[7] 2010/01/12 14:32:14: SIP Tx tcp:1.1.7.7:58486:

BYE sip:s-case1c.case.ru:5060;transport=Tcp;maddr=1.1.7.7;ms-opaque=d799212d2afe4476 SIP/2.0

Via: SIP/2.0/TCP 1.1.1.5:5060;branch=z9hG4bK-06141a693318556b7acae6c0d7f66226;rport

From: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b

To: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06

Call-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

CSeq: 9721 BYE

Max-Forwards: 70

Contact: <sip:Anonymous@1.1.1.5:5060;transport=tcp>

RTP-RxStat: Dur=14,Pkt=39,Oct=1092,Underun=2

RTP-TxStat: Dur=10,Pkt=678,Oct=116616

Content-Length: 0

 

 

[7] 2010/01/12 14:32:14: SIP Rx tcp:1.1.7.7:58486:

SIP/2.0 200 OK

FROM: <sip:+78632370684@1.1.1.5;user=phone>;tag=dc9bb8e98b

TO: <sip:+78632688634100@s-case1c.case.ru;user=phone>;tag=a6cc0a75;epid=77B47EEE06

CSEQ: 9721 BYE

CALL-ID: a6546ac6-7af5-4a17-8856-0f66a5817956

VIA: SIP/2.0/TCP 1.1.1.5:5060;branch=z9hG4bK-06141a693318556b7acae6c0d7f66226;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.5.0.0 MediationServer

 

 

[7] 2010/01/12 14:32:14: Call a6546ac6-7af5-4a17-8856-0f66a5817956#dc9bb8e98b: Clear last request

[5] 2010/01/12 14:32:14: BYE Response: Terminate a6546ac6-7af5-4a17-8856-0f66a5817956

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What do you want to change? The log looks fine at first glance.

 

P.S. It is interesting that the provider reveals the routing to you. Now you are probably able to shoot SIP packets to any location in their network!

 

this is sipnet.ru. Our Russian provider...

 

I want to dial extension number after I dial telephone number.

They want SIP-INFO DTMF, but I find on PBXnSIP forum, that this is not supported...

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this is sipnet.ru. Our Russian provider...

 

I want to dial extension number after I dial telephone number.

They want SIP-INFO DTMF, but I find on PBXnSIP forum, that this is not supported...

 

Arghh..

 

Well, I think the SIP INFO pass-through is supported... If you phone sends it it might actally get through. The "transcoding" from media DTMF to signalling DTMF is not supported.

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Arghh..

 

Well, I think the SIP INFO pass-through is supported... If you phone sends it it might actally get through. The "transcoding" from media DTMF to signalling DTMF is not supported.

 

My phone is Office Communicator 2007 R2 :) but I think, if I start to use something like SNOM it will work fine. With Skype to SIP I don't have these problems, only with sipnet.ru.

Unfortunually I can't test hardware phone, because I don't have it...

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