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TomH

T.38 pass through to Exchange 2007

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My issue is T.38 pass through to Exchange 2007.

 

Any help on missing pieces, trouble shooting, instructions, or examples would be appreciated.

Thanks to anyone for any help.

 

 

The flow is as follows

 

Fax Machine >> Broadvox >> PBXnSIP >> Exchange 2007

 

Some facts are:

• If called on a telephone line pbxnsip rings the phone and then passes the call on to exchange which answers and forward to exchange voice mail if no answer. Everything works great.

• Exchange is 172.x.x.81

• PBXnSIP is 172.x.x.75

 

 

When a fax machine rings:

 

*PBXnSIP rings phones

INVITE sip:103@be7.domain.com;user=phone SIP/2.0

.

.

.

*PBXnSIP Transfers to Exchange

[7] 2008/02/24 16:57:47: SIP Tx tcp:172.x.x.81:5065:

INVITE sip:103@be7.domain.com:5065;user=phone;transport=TCP SIP/2.0

.

.

.

*Exchange pickups

*Exchange recognizes fax

*Exchange transitions to T38

[7] 2008/02/24 16:57:58: SIP Rx tcp:172.x.x.81:5065:

INVITE sip:103@172.x.x.75:3444;transport=tcp SIP/2.0

FROM: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

TO: <sip:103@domain.com>;tag=20019

.

.

m=image 9200 udptl t38

a=T38FaxRateManagement:transferredTCF

a=T38FaxUdpEC:t38UDPFEC

 

 

*PBXnSIP transition to T38

[7] 2008/02/24 16:57:58: SIP Tx udp:64.152.60.75:5060:

UPDATE sip:5552907492@64.152.60.75:5060 SIP/2.0

Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport

From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

.

.

m=image 9086 udptl t38

a=T38FaxRateManagement:transferredTCF

a=T38FaxUdpEC:t38UDPFEC

 

 

 

*BroadVox transitions to T38

[7] 2008/02/24 16:57:58: SIP Rx udp:64.152.60.75:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport=5060

From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

Call-ID: 1459988973_58241008@64.152.60.75

.

.

m=image 0 udptl t38

a=T38FaxRateManagement:transferredTCF

a= T38FaxUdpEC:t38UDPFEC

a=sendrecv

 

 

Issues occurs now, it appear when Broadvox starts pinging port 0 instead of port given. In the m parameter of the SDP protocol I see a zero instead of a valid port# Could this be the issue.

 

Thanks again.

 

 

Full PBXnSIP Log follow:

 

[9] 2008/02/24 16:57:47: Resolve 1975: aaaa udp 64.152.60.75 5060

[9] 2008/02/24 16:57:47: Resolve 1975: a udp 64.152.60.75 5060

[9] 2008/02/24 16:57:47: Resolve 1975: udp 64.152.60.75 5060

[7] 2008/02/24 16:57:47: SIP Tx udp:64.152.60.75:5060:

SIP/2.0 183 Ringing

Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B04d42fb1611ec6f9

From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

Call-ID: 1459988973_58241008@64.152.60.75

CSeq: 18297 INVITE

Contact: <sip:Anonymous@172.x.x.75:5060;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/2.1.6.2448

Require: 100rel

RSeq: 1

Content-Type: application/sdp

Content-Length: 201

 

v=0

o=- 51427 51427 IN IP4 172.x.x.75

s=-

c=IN IP4 172.x.x.75

t=0 0

m=audio 9036 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

 

 

 

 

 

 

 

 

[8] 2008/02/24 16:57:47: DNS: Add dns_a be7.domain.com 172.x.x.81 (ttl=1200)

[9] 2008/02/24 16:57:47: Resolve 1974: a tcp be7.domain.com 5060

[9] 2008/02/24 16:57:47: Resolve 1974: tcp 172.x.x.81 5060

 

 

 

 

 

[7] 2008/02/24 16:57:47: SIP Tx tcp:172.x.x.81:5060:

INVITE sip:103@be7.domain.com;user=phone SIP/2.0

Via: SIP/2.0/TCP 172.x.x.75:3443;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport

From: "Tom Haselden" <sip:103@domain.com>;tag=20019

To: <sip:103@be7.domain.com;user=phone>

Call-ID: 708d4bca@pbx

CSeq: 8815 INVITE

Max-Forwards: 70

Contact: <sip:103@172.x.x.75:3443;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/2.1.6.2448

Diversion: <tel:103>;reason=no-answer;screen=no;privacy=off

Content-Type: application/sdp

Content-Length: 287

 

v=0

o=- 62722 62722 IN IP4 172.x.x.75

s=-

c=IN IP4 172.x.x.75

t=0 0

m=audio 9042 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=sendrecv

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:47: SIP Rx udp:64.152.60.75:5060:

PRACK sip:Anonymous@216.x.x.75:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05037402bed8a5c3

From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

Call-ID: 1459988973_58241008@64.152.60.75

CSeq: 18298 PRACK

Max-Forwards: 70

RAck: 1 18297 INVITE

Content-Length: 0

 

 

[9] 2008/02/24 16:57:47: Resolve 1976: aaaa udp 64.152.60.75 5060

[9] 2008/02/24 16:57:47: Resolve 1976: a udp 64.152.60.75 5060

[9] 2008/02/24 16:57:47: Resolve 1976: udp 64.152.60.75 5060

 

 

 

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:47: SIP Tx udp:64.152.60.75:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05037402bed8a5c3

From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

Call-ID: 1459988973_58241008@64.152.60.75

CSeq: 18298 PRACK

Contact: <sip:Anonymous@172.x.x.75:5060;transport=udp>

User-Agent: pbxnsip-PBX/2.1.6.2448

Content-Length: 0

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:47: SIP Rx tcp:172.x.x.81:5060:

SIP/2.0 100 Trying

FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

TO: <sip:103@be7.domain.com;user=phone>

CSEQ: 8815 INVITE

CALL-ID: 708d4bca@pbx

VIA: SIP/2.0/TCP 172.x.x.75:3443;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport

CONTENT-LENGTH: 0

 

 

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:47: SIP Rx tcp:172.x.x.81:5060:

SIP/2.0 302 Moved Temporarily

FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

TO: <sip:103@be7.domain.com;user=phone>;tag=60116e5511

CSEQ: 8815 INVITE

CALL-ID: 708d4bca@pbx

VIA: SIP/2.0/TCP 172.x.x.75:3443;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport

CONTACT: <sip:103@be7.domain.com:5065;user=phone;transport=TCP>

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

Diversion: <tel:103>;reason=no-answer;screen=no;privacy=off

 

 

[7] 2008/02/24 16:57:47: Call 708d4bca@pbx#20019: Clear last INVITE

[9] 2008/02/24 16:57:47: Resolve 1977: url sip:103@be7.domain.com;user=phone

[9] 2008/02/24 16:57:47: Resolve 1977: naptr be7.domain.com

[5] 2008/02/24 16:57:47: Redirecting call

[9] 2008/02/24 16:57:47: Resolve 1978: url sip:103@be7.domain.com:5065;user=phone;transport=TCP

[9] 2008/02/24 16:57:47: Resolve 1978: a tcp be7.domain.com 5065

[9] 2008/02/24 16:57:47: Resolve 1978: tcp 172.x.x.81 5065

 

 

 

 

 

 

[7] 2008/02/24 16:57:47: SIP Tx tcp:172.x.x.81:5065:

INVITE sip:103@be7.domain.com:5065;user=phone;transport=TCP SIP/2.0

Via: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport

From: "Tom Haselden" <sip:103@domain.com>;tag=20019

To: <sip:103@be7.domain.com;user=phone>

Call-ID: 708d4bca@pbx

CSeq: 8816 INVITE

Max-Forwards: 70

Contact: <sip:103@172.x.x.75:3444;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/2.1.6.2448

Diversion: <tel:103>;reason=no-answer;screen=no;privacy=off

Content-Type: application/sdp

Content-Length: 287

 

v=0

o=- 62722 62722 IN IP4 172.x.x.75

s=-

c=IN IP4 172.x.x.75

t=0 0

m=audio 9042 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=sendrecv

 

 

 

 

 

 

[8] 2008/02/24 16:57:47: DNS: Add dns_naptr be7.domain.com (ttl=3600)

[9] 2008/02/24 16:57:47: Resolve 1977: naptr be7.domain.com

[9] 2008/02/24 16:57:47: Resolve 1977: srv tls _sips._tcp.be7.domain.com

[8] 2008/02/24 16:57:47: DNS: Add dns_srv _sips._tcp.be7.domain.com (ttl=3600)

[9] 2008/02/24 16:57:47: Resolve 1977: srv tls _sips._tcp.be7.domain.com

[9] 2008/02/24 16:57:47: Resolve 1977: srv tcp _sip._tcp.be7.domain.com

[8] 2008/02/24 16:57:47: DNS: Add dns_srv _sip._tcp.be7.domain.com (ttl=3600)

[9] 2008/02/24 16:57:47: Resolve 1977: srv tcp _sip._tcp.be7.domain.com

[9] 2008/02/24 16:57:47: Resolve 1977: srv udp _sip._udp.be7.domain.com

[8] 2008/02/24 16:57:47: DNS: Add dns_srv _sip._udp.be7.domain.com (ttl=3600)

[9] 2008/02/24 16:57:47: Resolve 1977: srv udp _sip._udp.be7.domain.com

[9] 2008/02/24 16:57:47: Resolve 1977: a udp be7.domain.com 5060

[9] 2008/02/24 16:57:47: Resolve 1977: udp 172.x.x.81 5060

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:47: SIP Tx udp:172.x.x.81:5060:

ACK sip:103@be7.domain.com;user=phone SIP/2.0

Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-3a09d7e4ebaaeab8dbc1872cd2c1b08d;rport

From: "Tom Haselden" <sip:103@domain.com>;tag=20019

To: <sip:103@be7.domain.com;user=phone>;tag=60116e5511

Call-ID: 708d4bca@pbx

CSeq: 8815 ACK

Max-Forwards: 70

Contact: <sip:103@172.x.x.75:5060;transport=udp>

Content-Length: 0

 

 

 

[8] 2008/02/24 16:57:47: UDP: recvfrom receives ICMP message

[5] 2008/02/24 16:57:47: Connection refused on udp:172.x.x.81:5060

[6] 2008/02/24 16:57:47: Could not determine destination address on 1977

 

 

 

[7] 2008/02/24 16:57:47: SIP Rx tcp:172.x.x.81:5065:

SIP/2.0 100 Trying

FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

TO: <sip:103@be7.domain.com;user=phone>

CSEQ: 8816 INVITE

CALL-ID: 708d4bca@pbx

VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport

CONTENT-LENGTH: 0

 

 

 

[7] 2008/02/24 16:57:49: SIP Rx tcp:172.x.x.81:5065:

SIP/2.0 180 Ringing

FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

TO: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

CSEQ: 8816 INVITE

CALL-ID: 708d4bca@pbx

VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

 

[8] 2008/02/24 16:57:49: Play audio_en/ringback.wav

 

 

 

[7] 2008/02/24 16:57:52: SIP Rx tcp:172.x.x.81:5065:

SIP/2.0 200 OK

FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

TO: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

CSEQ: 8816 INVITE

CALL-ID: 708d4bca@pbx

VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-beeb0046e039369145a9685ba9818c99;rport

CONTACT: <sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81>;automata

CONTENT-LENGTH: 193

CONTENT-TYPE: application/sdp

ALLOW: UPDATE

SERVER: RTCC/3.0.0.0

ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

 

v=0

o=- 0 0 IN IP4 172.x.x.81

s=Microsoft Exchange Speech Engine

c=IN IP4 172.x.x.81

t=0 0

m=audio 35840 RTP/AVP 0 8 101

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:52: Call 708d4bca@pbx#20019: Clear last INVITE

[7] 2008/02/24 16:57:52: Set packet length to 20

[6] 2008/02/24 16:57:52: Sending RTP for 708d4bca@pbx#20019 to 172.x.x.81:35840

[9] 2008/02/24 16:57:52: Resolve 1979: aaaa tcp 172.x.x.81 5065

[9] 2008/02/24 16:57:52: Resolve 1979: a tcp 172.x.x.81 5065

[9] 2008/02/24 16:57:52: Resolve 1979: tcp 172.x.x.81 5065

 

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:52: SIP Tx tcp:172.x.x.81:5065:

ACK sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81 SIP/2.0

Via: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-abf7641b93baa986f47ec986f0906a1b;rport

From: "Tom Haselden" <sip:103@domain.com>;tag=20019

To: <sip:103@be7.domain.com;user=phone>;tag=f1f4aa832

Call-ID: 708d4bca@pbx

CSeq: 8816 ACK

Max-Forwards: 70

Contact: <sip:103@172.x.x.75:3444;transport=tcp>

Content-Length: 0

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:52: Determine pass-through mode after receiving response

[9] 2008/02/24 16:57:52: Resolve 1980: aaaa udp 64.152.60.75 5060

[9] 2008/02/24 16:57:52: Resolve 1980: a udp 64.152.60.75 5060

[9] 2008/02/24 16:57:52: Resolve 1980: udp 64.152.60.75 5060

 

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:52: SIP Tx udp:64.152.60.75:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B04d42fb1611ec6f9

From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

Call-ID: 1459988973_58241008@64.152.60.75

CSeq: 18297 INVITE

Contact: <sip:Anonymous@172.x.x.75:5060;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/2.1.6.2448

Content-Type: application/sdp

Content-Length: 201

 

v=0

o=- 51427 51427 IN IP4 172.x.x.75

s=-

c=IN IP4 172.x.x.75

t=0 0

m=audio 9036 RTP/AVP 0 101

a=rtpmap:0 pcmu/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:52: 708d4bca@pbx#20019: RTP pass-through mode

[7] 2008/02/24 16:57:52: 1459988973_58241008@64.152.60.75#a9a6181f02: RTP pass-through mode

[7] 2008/02/24 16:57:52: SIP Rx udp:64.152.60.75:5060:

ACK sip:Anonymous@172.x.x.75:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B051200e6bed8a5c3

From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

Call-ID: 1459988973_58241008@64.152.60.75

CSeq: 18297 ACK

Max-Forwards: 70

Content-Length: 0

 

 

[5] 2008/02/24 16:57:55: Call bf515e7b@pbx#19778: Last request not finished

[9] 2008/02/24 16:57:55: Resolve 1981: tcp 172.x.x.130 5060

 

 

 

 

 

[7] 2008/02/24 16:57:55: SIP Tx tcp:172.x.x.130:5060:

CANCEL sip:55552907492@sip:172.x.x.130:5060;transport=tcp;user=phone SIP/2.0

Via: SIP/2.0/TCP 172.x.x.75:3442;branch=z9hG4bK-3c718048209c673690214a2681b04258;rport

From: "Tom Haselden" <sip:103@domain.com>;tag=19778

To: <sip:55552907492@sip:172.x.x.130:5060;transport=tcp;user=phone>

Call-ID: bf515e7b@pbx

CSeq: 26279 CANCEL

Max-Forwards: 70

Content-Length: 0

 

 

[8] 2008/02/24 16:57:55: Hangup: Call bf515e7b@pbx#19778 not found

 

 

 

 

 

 

[7] 2008/02/24 16:57:58: SIP Rx tcp:172.x.x.81:5065:

INVITE sip:103@172.x.x.75:3444;transport=tcp SIP/2.0

FROM: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

TO: <sip:103@domain.com>;tag=20019

CSEQ: 1 INVITE

CALL-ID: 708d4bca@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK2e2c2c66

CONTACT: <sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81;ms-opaque=c9e23a1203e9a49b>;automata

CONTENT-LENGTH: 276

USER-AGENT: RTCC/3.0.0.0

CONTENT-TYPE: application/sdp

 

v=0

o=- 0 1 IN IP4 172.x.x.81

s=session

c=IN IP4 172.x.x.81

t=0 0

m=audio 0 RTP/AVP 0 8 101 13

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

m=image 9200 udptl t38

a=T38FaxRateManagement:transferredTCF

a=T38FaxUdpEC:t38UDPFEC

 

 

 

 

[7] 2008/02/24 16:57:58: UDP: Opening socket on port 9060

[7] 2008/02/24 16:57:58: UDP: Opening socket on port 9086

 

 

 

[9] 2008/02/24 16:57:58: Resolve 1982: url sip:5552907492@64.152.60.75:5060

[9] 2008/02/24 16:57:58: Resolve 1982: udp 64.152.60.75 5060

 

 

 

 

 

[7] 2008/02/24 16:57:58: SIP Tx udp:64.152.60.75:5060:

UPDATE sip:5552907492@64.152.60.75:5060 SIP/2.0

Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport

From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

Call-ID: 1459988973_58241008@64.152.60.75

CSeq: 707 UPDATE

Max-Forwards: 70

Contact: <sip:Anonymous@172.x.x.75:5060;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/2.1.6.2448

Content-Type: application/sdp

Content-Length: 276

v=0

o=- 51427 51428 IN IP4 172.x.x.75

s=-

c=IN IP4 172.x.x.75

t=0 0

m=audio 0 RTP/AVP 0 8 101 13

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

m=image 9086 udptl t38

a=T38FaxRateManagement:transferredTCF

a=T38FaxUdpEC:t38UDPFEC

 

 

 

 

[9] 2008/02/24 16:57:58: Resolve 1983: tcp 172.x.x.81 5065

 

 

 

 

 

[7] 2008/02/24 16:57:58: SIP Tx tcp:172.x.x.81:5065:

SIP/2.0 100 Trying

Via: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK2e2c2c66

From: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

To: <sip:103@domain.com>;tag=20019

Call-ID: 708d4bca@pbx

CSeq: 1 INVITE

Content-Length: 0

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:58: SIP Rx udp:64.152.60.75:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport=5060

From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

Call-ID: 1459988973_58241008@64.152.60.75

CSeq: 707 UPDATE

Contact: "Haselden Tom " <sip:5552907492@64.152.60.75:5060>

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

Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed

Supported: timer

Session-Expires: 1800;refresher=uas

Content-Length: 329

Content-Disposition: session; handling=required

Content-Type: application/sdp

 

v=0

o=Sonus_UAC 825 15704 IN IP4 64.152.60.75

s=SIP Media Capabilities

c=IN IP4 64.152.60.71

t=0 0

m=audio 32370 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

 

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

a=ptime:20

m=image 0 udptl t38

a=T38FaxRateManagement:transferredTCF

a= T38FaxUdpEC:t38UDPFEC

a=sendrecv

 

 

 

 

 

[7] 2008/02/24 16:57:58: Call 1459988973_58241008@64.152.60.75#a9a6181f02: Clear last request

[9] 2008/02/24 16:57:58: Resolve 1984: tcp 172.x.x.81 5065

 

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:58: SIP Tx tcp:172.x.x.81:5065:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK2e2c2c66

From: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

To: <sip:103@domain.com>;tag=20019

Call-ID: 708d4bca@pbx

CSeq: 1 INVITE

Contact: <sip:103@172.x.x.75:3444;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/2.1.6.2448

Content-Type: application/sdp

Content-Length: 275

 

v=0

o=- 62722 62723 IN IP4 172.x.x.75

s=-

c=IN IP4 172.x.x.75

t=0 0

m=audio 9060 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

m=image 0 udptl t38

a=T38FaxRateManagement:transferredTCF

a= T38FaxUdpEC:t38UDPFEC

 

 

 

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:57:58: SIP Rx tcp:172.x.x.81:5065:

ACK sip:103@172.x.x.75:3444;transport=tcp SIP/2.0

FROM: <sip:103@be7.domain.com;user=phone>;epid=AA79D0A609;tag=f1f4aa832

TO: <sip:103@domain.com>;tag=20019

CSEQ: 1 ACK

CALL-ID: 708d4bca@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 172.x.x.81:5065;branch=z9hG4bK68d5d4f

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.0.0.0

 

 

 

[5] 2008/02/24 16:58:02: SIP port accept from 172.x.x.81:18364

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:58:02: SIP Rx tcp:172.x.x.81:18364:

OPTIONS sip:172.x.x.75:5060 SIP/2.0

FROM: <sip:be7.domain.com:5060;transport=Tcp;ms-opaque=95944f3af7203520>;epid=7BE54968BB;tag=e59d6c85f0

TO: <sip:172.x.x.75:5060>

CSEQ: 6 OPTIONS

CALL-ID: ff3f5d4ff10b4315a6865919470a673f

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 172.x.x.81:18364;branch=z9hG4bK8948f21

ACCEPT: application/sdp

CONTENT-LENGTH: 0

USER-AGENT: RTCC/3.0.0.0

 

 

[9] 2008/02/24 16:58:02: Resolve 1985: tcp 172.x.x.81 18364

 

 

 

 

 

 

[7] 2008/02/24 16:58:02: SIP Tx tcp:172.x.x.81:18364:

SIP/2.0 200 Ok

Via: SIP/2.0/TCP 172.x.x.81:18364;branch=z9hG4bK8948f21

From: <sip:be7.domain.com:5060;transport=Tcp;ms-opaque=95944f3af7203520>;epid=7BE54968BB;tag=e59d6c85f0

To: <sip:172.x.x.75:5060>;tag=b3edb24a21

Call-ID: ff3f5d4ff10b4315a6865919470a673f

CSeq: 6 OPTIONS

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

Content-Length: 0

 

 

 

 

 

 

 

[7] 2008/02/24 16:58:31: SIP Rx udp:64.152.60.75:5060:

BYE sip:Anonymous@172.x.x.75:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05813609bed8a5c3

From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

Call-ID: 1459988973_58241008@64.152.60.75

CSeq: 18299 BYE

Max-Forwards: 70

Supported: 100rel

Content-Length: 0

 

 

 

 

 

 

[9] 2008/02/24 16:58:31: Resolve 1986: aaaa udp 64.152.60.75 5060

[9] 2008/02/24 16:58:31: Resolve 1986: a udp 64.152.60.75 5060

[9] 2008/02/24 16:58:31: Resolve 1986: udp 64.152.60.75 5060

 

 

 

 

 

 

 

[7] 2008/02/24 16:58:31: SIP Tx udp:64.152.60.75:5060:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 64.152.60.75:5060;branch=z9hG4bK05B05813609bed8a5c3

From: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

To: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

Call-ID: 1459988973_58241008@64.152.60.75

CSeq: 18299 BYE

Contact: <sip:Anonymous@172.x.x.75:5060;transport=udp>

User-Agent: pbxnsip-PBX/2.1.6.2448

RTP-RxStat: Dur=88,Pkt=272,Oct=46784,Underun=0

RTP-TxStat: Dur=63,Pkt=272,Oct=46784

Content-Length: 0

 

 

 

 

 

[7] 2008/02/24 16:58:31: 708d4bca@pbx#20019: Media-aware pass-through mode

[7] 2008/02/24 16:58:31: Other Ports: 1

[7] 2008/02/24 16:58:31: Call Port: 708d4bca@pbx#20019

[8] 2008/02/24 16:58:31: UDP: recvfrom receives ICMP message

[8] 2008/02/24 16:58:31: Last message repeated 13 times

 

[9] 2008/02/24 16:58:31: Resolve 1987: aaaa tcp 172.x.x.81 5065

[9] 2008/02/24 16:58:31: Resolve 1987: a tcp 172.x.x.81 5065

[9] 2008/02/24 16:58:31: Resolve 1987: tcp 172.x.x.81 5065

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:58:31: SIP Tx tcp:172.x.x.81:5065:

BYE sip:be7.domain.com:5065;transport=Tcp;maddr=172.x.x.81 SIP/2.0

Via: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-846f98ba102960c739a4a8f2c078531a;rport

From: "Tom Haselden" <sip:103@domain.com>;tag=20019

To: <sip:103@be7.domain.com;user=phone>;tag=f1f4aa832

Call-ID: 708d4bca@pbx

CSeq: 8817 BYE

Max-Forwards: 70

Contact: <sip:103@172.x.x.75:3444;transport=tcp>

RTP-RxStat: Dur=68,Pkt=4,Oct=688,Underun=0

RTP-TxStat: Dur=63,Pkt=285,Oct=49020

Content-Length: 0

 

 

 

 

 

[8] 2008/02/24 16:58:31: UDP: recvfrom receives ICMP message

[8] 2008/02/24 16:58:31: Last message repeated 3 times

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:58:31: SIP Rx tcp:172.x.x.81:5065:

SIP/2.0 200 OK

FROM: "Tom Haselden"<sip:103@domain.com>;tag=20019

TO: <sip:103@be7.domain.com;user=phone>;tag=f1f4aa832;epid=AA79D0A609

CSEQ: 8817 BYE

CALL-ID: 708d4bca@pbx

VIA: SIP/2.0/TCP 172.x.x.75:3444;branch=z9hG4bK-846f98ba102960c739a4a8f2c078531a;rport

CONTENT-LENGTH: 0

SERVER: RTCC/3.0.0.0

 

 

 

 

 

 

 

 

 

[7] 2008/02/24 16:58:31: Call 708d4bca@pbx#20019: Clear last request

[5] 2008/02/24 16:58:31: BYE Response: Terminate 708d4bca@pbx

[3] 2008/02/24 16:58:32: SMTP: Cannot resolve mail.domain.com

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*BroadVox transitions to T38

[7] 2008/02/24 16:57:58: SIP Rx udp:64.152.60.75:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 172.x.x.75:5060;branch=z9hG4bK-b070d615504a1056cba841938ea3e077;rport=5060

From: <sip:5554102925@172.x.x.75>;tag=a9a6181f02

To: "Haselden Tom " <sip:5552907492@64.152.60.75>;tag=gK056974f4

Call-ID: 1459988973_58241008@64.152.60.75

.

.

m=image 0 udptl t38

a=T38FaxRateManagement:transferredTCF

a=T38FaxUdpEC:t38UDPFEC

a=sendrecv

 

Yes that is definitevely a problem. Broadvox is usually quite responsive, would be great to have them provide this kind of service! Did you also ask them if they can do anything about it?

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Yes that is definitevely a problem. Broadvox is usually quite responsive, would be great to have them provide this kind of service! Did you also ask them if they can do anything about it?

 

 

Yes, I provided them the same info and will let you know what they say. From your standpoint this is something that should work and the issues appears to be on their side?

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Yes, I provided them the same info and will let you know what they say. From your standpoint this is something that should work and the issues appears to be on their side?

 

Well, sending to port 0 is not an option in IP, so that must be changed anyway. If that was the last obstable? You never know!

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Well, sending to port 0 is not an option in IP, so that must be changed anyway. If that was the last obstable? You never know!

 

BroadVox has two questions.

 

1) see below we recieve an "INVITE" from exchange but passs it through to broadvox as a "UPDATE" can it be passed through as an "INVITE"?

 

2) and more important can we limit the protocols to just T38 so there is no choise?

 

Thanks

 

Receives INVITE from 172.26.1.81 (Exchange) to go to T38

[7] 2008/02/28 16:45:39: SIP Rx tcp:172.26.1.81:5065:

INVITE sip:103@172.26.1.75:3006;transport=tcp SIP/2.0

FROM: <sip:103@be7.ezoutlook.com;user=phone>;epid=AA79D0A609;tag=a9fab924e2

TO: <sip:103@ezoutlook.com>;tag=59403

CSEQ: 1 INVITE

CALL-ID: a0b0509d@pbx

MAX-FORWARDS: 70

VIA: SIP/2.0/TCP 172.26.1.81:5065;branch=z9hG4bK2dfacb75

CONTACT: <sip:be7.ezoutlook.com:5065;transport=Tcp;maddr=172.26.1.81;ms-opaque=c9e23a1203e9a49b>;automata

CONTENT-LENGTH: 276

USER-AGENT: RTCC/3.0.0.0

CONTENT-TYPE: application/sdp

 

v=0

o=- 0 1 IN IP4 172.26.1.81

s=session

c=IN IP4 172.26.1.81

t=0 0

m=audio 0 RTP/AVP 0 8 101 13

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

m=image 9200 udptl t38

a=T38FaxRateManagement:transferredTCF

a=T38FaxUdpEC:t38UDPFEC

Sends UPDATE to BroadVox

 

[7] 2008/02/28 16:45:39: SIP Tx udp:64.152.60.75:5060:

UPDATE sip:2163736227@64.152.60.75:5060 SIP/2.0

Via: SIP/2.0/UDP 172.26.1.75:5060;branch=z9hG4bK-7dcdaaf32ca71adac0facc4140d0d153;rport

From: <sip:5024102925@172.26.1.75>;tag=c98233afea

To: "JERGENS INC " <sip:2163736227@64.152.60.75>;tag=gK0229cd68

Call-ID: 1661104583_128358926@64.152.60.75

CSeq: 2771 UPDATE

Max-Forwards: 70

Contact: <sip:Anonymous@172.26.1.75:5060;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/2.1.6.2448

Content-Type: application/sdp

Content-Length: 276

 

v=0

o=- 43466 43467 IN IP4 172.26.1.75

s=-

c=IN IP4 172.26.1.75

t=0 0

m=audio 0 RTP/AVP 0 8 101 13

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

m=image 9098 udptl t38

a=T38FaxRateManagement:transferredTCF

a=T38FaxUdpEC:t38UDPFEC

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1) see below we recieve an "INVITE" from exchange but passs it through to broadvox as a "UPDATE" can it be passed through as an "INVITE"?

 

2) and more important can we limit the protocols to just T38 so there is no choise?

 

1) There is a global setting called "support_update", if you set it to false the PBX will use INVITE (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File on how to set it).

 

2) Yea, I agree that the audio with port 0 is really strange (well, that's how we get it!). Maybe Broadvox can try with the INVITE and if that does no work we can make a version that takes the audio part out in this case.

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1) There is a global setting called "support_update", if you set it to false the PBX will use INVITE (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File on how to set it).

 

2) Yea, I agree that the audio with port 0 is really strange (well, that's how we get it!). Maybe Broadvox can try with the INVITE and if that does no work we can make a version that takes the audio part out in this case.

 

Well, just heard back from BroadVox. They are saying that the issue is that there are two codex in the SDP:

 

m=audio 0 RTP/AVP 0 8 101 13

a=rtpmap:0 PCMU/8000/1

a=rtpmap:8 PCMA/8000/1

a=rtpmap:101 telephone-event/8000

m=image 9200 udptl t38

a=T38FaxRateManagement:transferredTCF

a=T38FaxUdpEC:t38UDPFEC

 

I need to get rid of the audio which does have port 0. I know you all are just passing this through from exchange but can you offer any help on correcting it on exchange or correcting it in the pass through. I believe they are correct and if I can remove it the fax will pass through correctly.

 

Also, are there other PBXNSIP users sucessfully doing inbound faxing. If so maybe I need to reinstall.

 

 

Tom

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I also found this it appears CISCO call manager is having the same issue can you help?

 

http://forums.microsoft.com/TechNet/ShowPo...1&SiteID=17

 

Following is exerpt from microsoft forum.

 

"We've been down this path, and unfortunately were not able to reach any sort of solution via CUCM 6.01 & Ex07 SP1 (regardless of inband detection enabled). Neither or Cisco or Microsoft are really able to help on this unless they come to some sort of agreement. The details are basically as follows:

 

Microsoft and Cisco use two different standards for handling t.38 calls over SIP. The SIP signaling used by Microsoft Exchange to switch a call from audio to T.38 fax is not understood by CallManager, causing fax calls to fail.

 

Specifically, Exchange sends SDP with two "m=" lines, one to terminate audio and one to enable fax session. However, CallManager interprets it as audio requests that terminate the audio channels.

 

Currently CallManager only supports send/receive SIP INVITE signal to switch an audio call to T.38 call by a single image m=line in SIP SDP portion.

 

This SDP signals the endpoint to replace the existing channels (audio in our case) and establish a new channel for T.38.

 

Both implementations by Cisco CallManager and Microsoft Exchange conform to the standards (RFC3264 (MS) and ITU T.38 (Cisco)).

 

Support for both standards is in development for CallManager, This functionality will be fully supported in CallManager 7.0. It is currently in a 'resolved' state meaning this functionality is tested and confirmed in development builds of 7.0.

 

This is tracked via BugID CSCsg60357, but was private the last I checked.

 

Hope this helps. If anyone can use this information to get this functional, I'd love to see it."

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1) There is a global setting called "support_update", if you set it to false the PBX will use INVITE (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File on how to set it).

 

2) Yea, I agree that the audio with port 0 is really strange (well, that's how we get it!). Maybe Broadvox can try with the INVITE and if that does no work we can make a version that takes the audio part out in this case.

 

 

Here is where it stands Broadvox is not going to fix the issue with multiple m records in the sdp any time soon. I need to move on. I am going to look for a SIP trunk vendor that does support this. At the same time does PBXnSIP have the abilility to recieve and forward inbound faxes? this is another alternative for us.

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I am having the same issue with a provider using genband.

 

seems MS needs to remove the =m audio part to make most of the providers happy.

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