03-24-2017 03:10 AM - edited 03-17-2019 09:53 AM
Hello,
I tried to transfer an inbound fax to an outband fax as below :
Fax1 -------ITSP-------SIP-trunk--------CUBE--------CUCM -------- SIP-trunk---ITSP-------Fax2
Fax 2 is available when i dial him directly.
When i tried to send a fax from Fax1 to Fax2 there is a cancel in SIP messages
On Cube i just apply this command for T38 and inbound outband faxes works :
voice service voip
address-hiding
mode border-element license capacity 525
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
rel1xx disable
session refresh
header-passing
error-passthru
asserted-id pai
privacy pstn
early-offer forced
midcall-signaling passthru
privacy-policy passthru
Below some SIP messages ( in attach complete logs):
Received:
INVITE sip:+331708XXXXX@21.159.X.X:5060 SIP/2.0
Via: SIP/2.0/UDP 21.159.X.X:5060;branch=z9hG4bK0cBdd041bcd4da81538
From: <sip:+337XXXXXX@21.159.X.X>;tag=gK0c64f835
To: <sip:+33170xXXXXX@21.159.X.X>
Call-ID: 755798237_121622568@21.159.X.X
CSeq: 2123 INVITE
Max-Forwards: 70
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
Contact: <sip:+3375XXXXXX@21.159.X.X:5060>
Supported: timer,100rel
Session-Expires: 1800
Min-SE: 90
Content-Length: 310
Content-Disposition: session; handling=required
Content-Type: application/sdp
v=0
o=Sonus_UAC 21749 2003 IN IP4 21.159.X.X
s=SIP Media Capabilities
c=IN IP4 21.159.236.196
t=0 0
m=audio 16330 RTP/AVP 18 2 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
*****************************************************************************************
Invite with divertion header
INVITE sip:+331408XXXXX@21.159.236.57 SIP/2.0
Via: SIP/2.0/UDP 21.159.X.X:5060;branch=z9hG4bK479731ED2
From: <sip:+337XXXXXX@21.159.X.X>;tag=7875C030-4EF
To: <sip:+331408XXXXXX@21.159.X.X>
Date: Thu, 23 Mar 2017 17:48:47 GMT
Call-ID: CCCE8F52-F2711E7-BE8DA4FF-C9587607@21.159.X.X
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 4124897024-0000065536-0000086622-1862304021
User-Agent: Cisco-SIPGateway/IOS-15.5.3.S3
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1490291327
Contact: <sip:+33751808@21.159.X.X:5060>
Expires: 180
Allow-Events: kpml, telephone-event
Max-Forwards: 67
P-Asserted-Identity: <sip:+337XXXXXXX@21.159.X.X>
Diversion: "Test Fax"<sip:+331708XXXXX@21.159.X.X>;privacy=off;reason=unconditional;screen=yes
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserA
4431-LON2-01F-CUBE-U1#gent 7173 7938 IN IP4 21.159.X.X
s=SIP Call
c=IN IP4 21.159.X.X
t=0 0
m=audio 16954 RTP/AVP 8 101
c=IN IP4 21.159.X.X
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*********************************************************************
Received:
SUBSCRIBE sip:+3314087XXXXX21.138.0.30:5060 SIP/2.0
Via: SIP/2.0/UDP 21.129.X.X:5060;branch=z9hG4bK72a1db7a9b91c7
From: <sip:+337XXXXXXXX@21.129.X.X>;tag=153710321~1f78a770-7fe4-81a3-46fb-0d284cb2492c-39200425
To: <sip:+3314087XXXX@21.138.X.X>;tag=7875C2AF-1457
Call-ID: f5dcef00-8d410a7f-5a4b12-6f008115@21.129.X.X
CSeq: 102 SUBSCRIBE
Date: Thu, 23 Mar 2017 17:48:47 GMT
User-Agent: Cisco-CUCM10.5
Event: kpml
Expires: 7200
Contact: <sip:21.129.X.X:5060>
P-Asserted-Identity: <sip:+337XXXXXXXX@21.129.X.X>
Accept: application/kpml-response+xml
Max-Forwards: 70
Content-Type: application/kpml-request+xml
Content-Length: 370
<?xml version="1.0" encoding="UTF-8" ?>
<kpml-request xmlns="urn:ietf:params:xml:ns:kpml-request" xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:schemaLocation="urn:ietf:params:xml:ns:kpml-request kpml-request.xsd" version="1.0">
<pattern interdigittimer="7260000" persist="persist">
<regex tag="dtmf">[x*#ABCD]</regex>
</pattern>
</kpml-request>
May i need to enable Fax Relay or something else (passtrought)?
Regards
Thomas
03-24-2017 04:20 AM
Logically you should see a complete dialogue for the audio call getting connected and then a re-invite for the switch-over to fax protocol T38 where fax capabilities will be exchanged.
Regards
Abhay
03-27-2017 02:19 AM
Hello,
Thanks for your answers,
As we configured divert option, in the diversion header there is the DDI belong to us , so the call is authorized.
Abhay, it's true the call is not establish , before modify config of T38 , i have to check why the call is not establish.
In this kind of calls , MTP is maybe required ? that's maybe why i received a cancel
We don't enable MTP required as we are in full G711 on trunk beetween CUBE and CUCM
Regards
Thomas
03-24-2017 06:17 AM
Is your ITSP carrier allowing you to send calls with caller ID not belonging to you? It is often times not allowed and one needs to insert owned caller ID in diversion header.
To test it can you make a similar call from external phone that is forwarded to another external destination?
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