I work for an ISP. We are delivering a SIP trunk from our Broadworks platform to our customer's Cisco Cube. The customer's Cube is sending our border controller a T38 invite sequence.
Failed Call:
INVITE sip:[REDACTED];cic=0432@10.154.0.30;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.156.0.30;branch=z9hG4bK+54a9a273bb0e927f13130682c596d1ce+000a1e9c+1
From: "ESSE HEALTH - M"<sip:3148511000@10.156.0.30;user=phone>;tag=000a1e9c+1+97ad0014+8bff99cc
To: <sip:[REDACTED]@10.154.0.30;user=phone>
CSeq: 831862043 INVITE
Expires: 90
Call-ID: 13697423000a1e9c
Remote-Party-ID: "ESSE HEALTH - M"<sip:[REDACTED]@10.156.0.30;user=phone>;party=calling;id-type=subscriber;privacy=off;screen=yes
Max-Forwards: 70
Contact: <sip:[REDACTED]@10.156.0.30;user=phone>
Content-Type: application/sdp
Content-Length: 192
v=0
o=- 3808740692 3808740692 IN IP4 10.156.0.30
s=-
c=IN IP4 10.158.0.10
t=0 0
m=audio 39688 RTP/AVP 0 18
a=sendrecv
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
Working call:
INVITE sip:+[REDACTED]@10.154.0.20:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.156.0.30;branch=z9hG4bK+cca243ed0e4ca0a89c11c43d4b809497+000a1e9c+1
From: "ESSE HEALTH - M"<sip:[REDACTED]@10.156.0.30;user=phone>;tag=000a1e9c+1+31500015+61f17373
To: <sip:[REDACTED]@10.155.0.30;user=phone>
CSeq: 257211438 INVITE
Expires: 90
Call-ID: 13659948000a1e9c
Remote-Party-ID: "ESSE HEALTH - M"<sip:[REDACTED]@10.156.0.30;user=phone>;party=calling;id-type=subscriber;privacy=off;screen=yes
Max-Forwards: 70
Contact: <sip:[REDACTED]@10.156.0.30;user=phone>
Content-Type: application/sdp
Content-Length: 246
v=0
o=- 3808737702 3808737702 IN IP4 10.156.0.30
s=-
c=IN IP4 10.158.0.10
t=0 0
m=audio 46996 RTP/AVP 0 18 101
a=sendrecv
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=fmtp:18 annexb=no
In the Working call, I see the DTMF Media Definitions applied :a=rtpmap:101 telephone-event/8000
Also on the working fax call, I am seeing the media format parameter :a=fmtp:101 0-15
I have little experience with the Cisco Cube and any assistance would be welcome.
The customer has provided us with the IOS config of the unit itself.
hostname co-cube1
!
boot-start-marker
boot system flash bootflash:/isr4300-universalk9.16.09.05.SPA.bin
boot-end-marker
!
!
vrf definition Mgmt-intf
!
address-family ipv4
exit-address-family
!
address-family ipv6
exit-address-family
!
logging buffered 10000000
no logging console
no logging monitor
!
aaa new-model
!
!
aaa session-id common
clock timezone CST -6 0
clock summer-time CDT recurring
!
!
!
ip name-server x.x.x.x
ip domain name mem-ins.com
!
!
!
login on-success log
!
!
!
!
!
!
!
subscriber templating
!
!
!
!
!
!
!
multilink bundle-name authenticated
!
!
!
!
!
!
voice service voip
ip address trusted list
ipv4 x.x.x.x
ipv4 x.x.x.x
ipv4 [REDACTED]
mode border-element license capacity 100
media anti-trombone
media statistics
media bulk-stats
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
h323
call service stop
sip
bind control source-interface Port-channel1
bind media source-interface Port-channel1
header-passing
asserted-id pai
asymmetric payload full
early-offer forced
call-route url
sip-profiles inbound
!
!
voice class uri CUCM sip
host ipv4:x.x.x.x
host ipv4:x.x.x.x
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g722-64
!
voice class codec 2
codec preference 1 g711ulaw
!
!
voice class sip-profiles 1
request INVITE sip-header From modify "From: (.*<)(.*>)" "From: <\2"
request INVITE sip-header From modify "(<.*:)(.*@)" "\[REDACTED]@"
!
voice class sip-profiles 2
request INVITE peer-header sip FROM copy ".*(x-.*>)" u01
request INVITE sip-header From modify "(.*)>(.*)" "\1;\u01\2"
!
!
voice class dpg 102
dial-peer 102
!
voice class dpg 202
dial-peer 202
!
voice class server-group 1
ipv4 x.x.x.x preference 1
ipv4 x.x.x.x preference 2
description *** CUCM ***
!
voice class sip-options-keepalive 1
!
!
voice iec syslog
!
!
voice translation-rule 1
rule 1 /.+\(....\)/ /\1/
!
!
voice translation-profile last4
translate called 1
!
!
!
!
voice-card 0/4
dsp services dspfarm
no watchdog
!
diagnostic bootup level minimal
!
spanning-tree extend system-id
!
!
!
redundancy
mode none
!
!
!
!
!
!
!
interface Port-channel1
ip address [REDACTED] 255.255.240.0
no negotiation auto
!
interface GigabitEthernet0/0/0
no ip address
negotiation auto
channel-group 1 mode active
!
interface GigabitEthernet0/0/1
no ip address
negotiation auto
channel-group 1 mode active
!
!
ip default-gateway 10.0.16.1
ip forward-protocol nd
no ip http server
ip http authentication local
no ip http secure-server
ip tftp source-interface GigabitEthernet0
ip route 0.0.0.0 0.0.0.0 [REDACTED]
!
ip ssh version 2
!
!
ip access-list extended CAP-FILTER
permit ip any any
logging origin-id hostname
logging host x.x.x.x
logging host x.x.x.x
!
!
!
!
!
control-plane
!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
sccp local Port-channel1
sccp ccm x.x.x.x identifier 1 version 7.0
sccp ccm x.x.x.x identifier 2 version 7.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register CUBE_COL_XCODE
keepalive retries 1
keepalive timeout 10
switchover method immediate
switchback method immediate
!
!
ccm-manager redundant-host x.x.x.x
ccm-manager config server x.x.x.x
ccm-manager sccp local Port-channel1
ccm-manager sccp
!
dspfarm profile 1 transcode
codec g711ulaw
codec g729r8
codec g722-64
maximum sessions 32
associate application SCCP
!
dial-peer voice 101 voip
description **In from ITSP**
translation-profile outgoing last4
session protocol sipv2
session target sip-server
destination dpg 102
incoming called-number .
voice-class codec 2
dtmf-relay rtp-nte sip-kpml
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
no vad
!
dial-peer voice 102 voip
description **Out to CUCM**
translation-profile outgoing last4
destination-pattern 111
session protocol sipv2
session server-group 1
voice-class codec 1
dtmf-relay rtp-nte sip-kpml
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
no vad
!
dial-peer voice 201 voip
description **In from CUCM**
session protocol sipv2
session server-group 1
destination dpg 202
incoming uri via CUCM
voice-class codec 1
dtmf-relay rtp-nte sip-kpml
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
no vad
!
dial-peer voice 202 voip
description **Out to ITSP**
destination-pattern 211
session protocol sipv2
session target sip-server
voice-class codec 2
voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte sip-kpml
fax-relay ecm disable
fax-relay sg3-to-g3
fax nsf 000000
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
no vad
!
!
