Having an issue with receiving faxes, but when I pass through the fax.
Currently, I receive the fax with Digium's Fax for Asterisk, store it and
the initiate an outbound call to our fax server. (XMedius Fax). This
works, but we would prefer to have Asterisk simply route the call directly
to the fax server and take the store and forward out of the equation.
When I do that, however, the fax is never properly negotiated. One thing I
have noticed is that XMedius Fax tells Asterisk it has
'a=T38maxBitRate:14400' and Asterisk immediately turns around and tells
our upstream provider 'a=T38MaxBitRate:2400' on the invites (full invite
text below). Is the fact that XMedius is not capitalizing the 'm' in
'T38maxBitRate' the cause of Asterisk telling the upstream provider that
the 'T38MaxBitRate' is 2400?
This should be the relevant sip debug. I have replaced the IP addresses
with XXX.XXX.XXX.XXX (or WWW or YYY or ZZZ) as appropriate.
<--- SIP read from UDP:WWW.WWW.WWW.WWW:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP WWW.WWW.WWW.WWW:5060;branch=z9hG4bK-40A939C7AE8C
From: sip:XMFAX2.mydomain.world;tag=095775A0931E
To: sip:[email protected];tag=as09ca5622
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Max-Forwards: 70
Contact: sip:WWW.WWW.WWW.WWW:5060
User-Agent: XMediusFAX/7.0.0.298
Content-Type: application/sdp
Content-Length: 315
v=0
o=XMedius-Fax-Gateway 76811410 411 IN IP4 WWW.WWW.WWW.WWW
s=Asterisk PBX 10.5.0-digiumphones
c=IN IP4 WWW.WWW.WWW.WWW
t=0 0
m=image 54296 udptl t38
a=T38FaxVersion:0
a=T38maxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:8192
a=T38FaxMaxDatagram:1008
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (11 headers 12 lines) ---
Sending to WWW.WWW.WWW.WWW:5060 (no NAT)
== Using UDPTL CoS mark 5
Got T.38 offer in SDP in dialog
[email protected]:5060
Capabilities: us - (ulaw), peer -
audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0
(nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
<--- Transmitting (no NAT) to WWW.WWW.WWW.WWW:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
WWW.WWW.WWW.WWW:5060;branch=z9hG4bK-40A939C7AE8C;received=WWW.WWW.WWW.WWW
From: sip:XMFAX2.mydomain.world;tag=095775A0931E
To: sip:[email protected];tag=as09ca5622
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: Asterisk PBX 10.5.0-digiumphones
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
== Using UDPTL CoS mark 5
set_destination: Parsing <sip:[email protected]:5060> for
address/port to send to
set_destination: set destination to XXX.XXX.XXX.XXX:5060
Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK7ef2185a;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as40d4ca92
To: <sip:[email protected];isup-oli=0>;tag=gK020b0efc
Contact: <sip:6024667281;[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 10.5.0-digiumphones
Session-Expires: 1800;refresher=uas
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 505811356 505811358 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=Asterisk PBX 10.5.0-digiumphones
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=image 4464 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:2400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:507
a=T38FaxUdpEC:t38UDPFEC
---
<--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK7ef2185a;rport=5060
From: <sip:[email protected]>;tag=as40d4ca92
To: <sip:[email protected];isup-oli=0>;tag=gK020b0efc
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:XXX.XXX.XXX.XXX:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bK7ef2185a;rport=5060
From: <sip:[email protected]>;tag=as40d4ca92
To: <sip:[email protected];isup-oli=0>;tag=gK020b0efc
Call-ID: [email protected]
CSeq: 102 INVITE
Accept: application/sdp, application/isup, application/dtmf,
application/dtmf-relay, multipart/mixed
Contact: <sip:[email protected]:5060>
Allow:
INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH
Supported: timer
Session-Expires: 1800;refresher=uas
Content-Length: 303
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 28889 23043 IN IP4 XXX.XXX.XXX.XXX
s=SIP Media Capabilities
c=IN IP4 208.49.73.36
t=0 0
m=image 25030 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:2400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:262
a=T38FaxMaxDatagram:176
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv
<------------->
--- (14 headers 13 lines) ---
Got T.38 offer in SDP in dialog [email protected]
Capabilities: us - (ulaw|alaw|g729), peer -
audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0
(nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
set_destination: Parsing <sip:[email protected]:5060> for
address/port to send to
set_destination: set destination to XXX.XXX.XXX.XXX:5060
Transmitting (NAT) to XXX.XXX.XXX.XXX:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK0dbf9f05;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as40d4ca92
To: <sip:[email protected];isup-oli=0>;tag=gK020b0efc
Contact: <sip:6024667281;[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 10.5.0-digiumphones
Content-Length: 0
---
<--- Reliably Transmitting (no NAT) to WWW.WWW.WWW.WWW:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
WWW.WWW.WWW.WWW:5060;branch=z9hG4bK-40A939C7AE8C;received=WWW.WWW.WWW.WWW
From: sip:XMFAX2.mydomain.world;tag=095775A0931E
To: sip:[email protected];tag=as09ca5622
Call-ID: [email protected]:5060
CSeq: 103 INVITE
Server: Asterisk PBX 10.5.0-digiumphones
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 495936988 495936989 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
s=Asterisk PBX 10.5.0-digiumphones
c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ
t=0 0
m=image 4241 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:2400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:339
a=T38FaxUdpEC:t38UDPRedundancy
<------------>
<--- SIP read from UDP:WWW.WWW.WWW.WWW:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP
WWW.WWW.WWW.WWW:5060;branch=z9hG4bK-40A939C7AE8C;received=WWW.WWW.WWW.WWW
From: sip:XMFAX2.mydomain.world;tag=095775A0931E
To: sip:[email protected];tag=as09ca5622
Call-ID: [email protected]:5060
Max-Forwards: 70
CSeq: 103 ACK
Contact: sip:WWW.WWW.WWW.WWW:5060
Content-Length: 0
<------------->
--- (9 headers
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
--
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