CUBE: one way audio

[Hunor Szombatfalvi]

Hello everyone,

I'm having an issue with one way audio connecting to a sip provider. It looks like i'm not sending audio to them from the CUBE.

here is my CUBE setup:

version 15.4
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
!
hostname IACHZURVGW01
!
boot-start-marker
boot-end-marker
!
!
logging buffered 65536
enable secret 5 $1$LkVj$aXDs4n6xS77nlK5tNeitl0
!
aaa new-model
!
!
aaa group server tacacs+ EPMACS
 server name ACS_EU
 server name ACS_NA
!
aaa authentication login default group EPMACS group tacacs+ local
aaa authorization config-commands
aaa authorization exec default group EPMACS group tacacs+ local
aaa authorization commands 0 default group EPMACS group tacacs+ local
aaa authorization commands 1 default group EPMACS group tacacs+ local
aaa authorization commands 15 default group EPMACS group tacacs+ local
aaa accounting exec default start-stop group EPMACS group tacacs+
aaa accounting commands 1 default start-stop group EPMACS group tacacs+
aaa accounting commands 15 default start-stop group EPMACS group tacacs+
!
!
!
!
!
aaa session-id common
!
!
!
!
!
!
!
!
!
!
!
no ip dhcp conflict logging
ip dhcp excluded-address 10.75.32.129 10.75.32.140
!
ip dhcp pool VOIP_Zurich
 network 10.75.32.128 255.255.255.192
 dns-server 129.130.102.100 129.130.102.101
 default-router 10.75.32.129
 option 150 ip 10.96.108.12 10.96.109.11
!
!
!
ip multicast-routing
ip cef
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
!
!
cts logging verbose
!
!
voice-card 0
 dspfarm
 dsp services dspfarm
!
!
voice call send-alert
voice call convert-discpi-to-prog always
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
 ip address trusted list
  ipv4 10.96.108.11
  ipv4 10.96.108.12
  ipv4 10.96.108.13
  ipv4 10.96.109.11
  ipv4 10.75.32.128 255.255.255.192
  ipv4 10.75.32.240 255.255.255.240
 allow-connections sip to sip
 no supplementary-service sip handle-replaces
 redirect ip2ip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711alaw
 sip
  session transport tcp
  min-se 1100 session-expires 1100
  registrar server expires max 600 min 60
  asserted-id pai
  no anat
  early-offer forced
  no silent-discard untrusted
  history-info
  midcall-signaling passthru
  sip-profiles inbound
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g711alaw
 codec preference 3 g729r8
 codec preference 4 ilbc
!
!
voice class sip-profiles 11
 request INVITE sip-header SIP-Req-URI modify "sip:564044320(.*)@" "sip:88602243@"
 request INVITE sip-header From modify "sip:(.*);phone-context=international@" "sip:+\1@"
 request INVITE sip-header From modify "sip:(.*);phone-context=national@" "sip:+41\1@"
 request INVITE sip-header To modify "sip:564044320@" "sip:88602243@"
 request INVITE sip-header Contact modify "sip:(.*);phone-context=international@" "sip:+\1@"
 request INVITE sip-header Contact modify "sip:(.*);phone-context=national@" "sip:+41\1@"
 request INVITE sip-header P-Asserted-Identity modify "sip:(.*);phone-context=international@" "sip:+\1@"
 request INVITE sip-header P-Asserted-Identity modify "sip:(.*);phone-context=national@" "sip:+41\1@"
!
voice class sip-profiles 10
 request INVITE sip-header SIP-Req-URI modify "sip:000(.*)@" "sip:\1;phone-context=international@"
 request INVITE sip-header SIP-Req-URI modify "sip:00041(.*)@" "sip:\1;phone-context=national@"
 request INVITE sip-header SIP-Req-URI modify "sip:00(.*)@" "sip:\1;phone-context=national@"
 request INVITE sip-header From modify "sip:88602243@" "sip:564044320@"
 request INVITE sip-header To modify "sip:000(.*)@" "sip:\1;phone-context=international@"
 request INVITE sip-header To modify "sip:00041(.*)@" "sip:\1;phone-context=national@"
 request INVITE sip-header To modify "sip:00(.*)@" "sip:\1;phone-context=national@"
 request INVITE sip-header Contact modify "sip:88602243@" "sip:564044320@"
 request INVITE sip-header P-Asserted-Identity modify "sip:88602243@" "sip:564044320@"
!
!
!
voice iec syslog
voice register global
 mode srst
 timeouts interdigit 3
 system message Operating in Fallback mode
 max-dn 200
 max-pool 1
 ip qos dscp af31 signal
!
voice register pool  1
 busy-trigger-per-button 1
 id network 10.75.32.128 mask 255.255.255.192
!
!
!
!
!
license udi pid CISCO2911/K9 sn FCZ194171VP
hw-module pvdm 0/0
!
!
!
username netadmin secret 5 $1$v6/J$DAhOpzE./YOWg/8UH004p/
!
redundancy
!
!
ip ssh version 2
!
class-map match-any VOIP-SIG
 match protocol skinny
 match protocol h323
 match protocol sip
class-map match-all VOIP-RTP
 match protocol rtp
!
policy-map VOIP
 class VOIP-RTP
  priority percent 70
 class VOIP-SIG
  bandwidth 100
!
!
!
!
!
interface Loopback0
 ip address 10.75.32.241 255.255.255.240
!
interface Embedded-Service-Engine0/0
 no ip address
 shutdown
!
interface GigabitEthernet0/0
 description primary
 backup interface GigabitEthernet0/1
 ip address 10.75.32.130 255.255.255.192
 ip pim sparse-mode
 duplex auto
 speed auto
 service-policy output VOIP
!
interface GigabitEthernet0/1
 description back up
 ip address 10.75.32.130 255.255.255.192
 ip pim sparse-mode
 duplex auto
 speed auto
 service-policy output VOIP
!
interface GigabitEthernet0/2
 ip address 192.168.10.2 255.255.255.0
 duplex auto
 speed auto
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 10.75.32.129
ip route 192.168.10.0 255.255.255.0 192.168.10.1
ip route 193.246.242.0 255.255.255.0 192.168.10.1
ip tacacs source-interface Loopback0
!
logging source-interface Loopback0
logging host 10.69.97.7
logging host 10.69.97.169
!
!
tacacs server ACS_EU
 address ipv4 10.69.97.6
 single-connection
tacacs server ACS_NA
 address ipv4 10.223.51.12
 single-connection
access-list 51 permit 10.69.97.2
access-list 51 permit 10.69.131.3
access-list 51 remark *** SNMP_Control_ACL ***
access-list 51 permit 10.69.0.0 0.0.252.255
!
!
!
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 Loopback0
sccp ccm 10.96.109.11 identifier 1 priority 1 version 7.0
sccp ccm 10.96.108.13 identifier 2 priority 2 version 7.0
sccp ccm 10.72.83.129 identifier 3 priority 3 version 7.0
sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate ccm 2 priority 2
 associate ccm 3 priority 3
 associate profile 3 register IACHZURCONF01
 associate profile 2 register IACHZURXCODE01
 associate profile 1 register IACHZURSIPMTP
!
ccm-manager music-on-hold bind Loopback0
!
!
dspfarm profile 2 transcode
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 codec ilbc
 codec g729r8
 codec g729br8
 maximum sessions 5
 associate application SCCP
!
dspfarm profile 3 conference
 codec g729br8
 codec g729r8
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 codec ilbc
 maximum sessions 1
 associate application SCCP
!
dspfarm profile 1 mtp
 codec g711alaw
 maximum sessions software 150
 associate application SCCP
!
dial-peer voice 20 voip
 description to CUCM03
 destination-pattern 88602T
 session protocol sipv2
 session target ipv4:10.96.109.11
 incoming called-number 0T
 voice-class codec 1
 voice-class sip bind control source-interface Loopback0
 voice-class sip bind media source-interface Loopback0
 dtmf-relay rtp-nte
 fax rate disable
!
dial-peer voice 21 voip
 description to CUCM04
 translation-profile incoming VOIP_IN_CUCM
 preference 2
 destination-pattern 88602T
 session protocol sipv2
 session target ipv4:10.96.108.13
 incoming called-number 0T
 voice-class codec 1
 voice-class sip bind control source-interface Loopback0
 voice-class sip bind media source-interface Loopback0
 dtmf-relay rtp-nte
 fax rate disable
!
dial-peer voice 10 voip
 description to SIP Provider
 destination-pattern .T
 no modem passthrough
 session protocol sipv2
 session target ipv4:193.246.242.84
 session transport udp
 incoming called-number [1-9]T
 voice-class sip localhost dns:sbc01.ch
 voice-class sip profiles 10
 voice-class sip profiles 11 inbound
 voice-class sip options-keepalive up-interval 30 down-interval 60
 voice-class sip bind control source-interface GigabitEthernet0/2
 voice-class sip bind media source-interface GigabitEthernet0/2
 dtmf-relay rtp-nte
 codec g711alaw
 fax rate 14400
 no vad
!
!
gateway
 timer receive-rtp 1200
!
sip-ua
 no remote-party-id
 set pstn-cause 31 sip-status 503
 retry invite 3
 retry bye 3
 retry cancel 3
 timers trying 200
 connection-reuse
!
!
!
gatekeeper
 shutdown
!
!
!
line con 0
line aux 0
line 2
 no activation-character
 no exec
 transport preferred none
 transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
 stopbits 1
line vty 0 4
 transport input ssh
line vty 5 15
 transport input ssh
!
scheduler allocate 20000 1000
ntp server 10.16.64.124
ntp server 10.20.64.124
!
end

the phones can't reach the ip of the IPTS directly but i have REDIRECT IP2IP configured and DSPFARM, which should do the trick if i understood correctly.

when i place a call if i do a SHOW SIP-UA CALLS it looks like on the outbound stream the source and destination ip are the same and doing a DEBUG VOIP RTP confirms this, there are only TX stream from 10.75.32.241 to 10.75.32.241.

Total SIP call legs:2, User Agent Client:1, User Agent Server:1
SIP UAC CALL INFO
Call 1
SIP Call ID                : 596C02E6-8E2511E5-8280E351-94B9191D@sbc01.ch
   State of the call       : STATE_ACTIVE (7)
   Substate of the call    : SUBSTATE_NONE (0)
   Calling Number          : 88602243
   Called Number           : 00040756039061
   Called URI              : sip:00040756039061@193.246.242.84:5060
   Bit Flags               : 0xC04018 0x90000100 0x0
   CC Call ID              : 564
   Source IP Address (Sig ): 192.168.10.2
   Destn SIP Req Addr:Port : [193.246.242.84]:5060
   Destn SIP Resp Addr:Port: [193.246.242.84]:5060
   Destination Name        : 193.246.242.84
   Number of Media Streams : 1
   Number of Active Streams: 1
   RTP Fork Object         : 0x0
   Media Mode              : flow-through
   Media Stream 1
     State of the stream      : STREAM_ACTIVE
     Stream Call ID           : 564
     Stream Type              : voice+dtmf (1)
     Stream Media Addr Type   : 1
     Negotiated Codec         : g711alaw (160 bytes)
     Codec Payload Type       : 8
     Negotiated Dtmf-relay    : rtp-nte
     Dtmf-relay Payload Type  : 101
     QoS ID                   : -1
     Local QoS Strength       : BestEffort
     Negotiated QoS Strength  : BestEffort
     Negotiated QoS Direction : None
     Local QoS Status         : None
     Media Source IP Addr:Port: [192.168.10.2]:16534
     Media Dest IP Addr:Port  : [193.246.244.68]:48808


Options-Ping    ENABLED:NO    ACTIVE:NO
   Number of SIP User Agent Client(UAC) calls: 1

SIP UAS CALL INFO
Call 1
SIP Call ID                : 82076f80-64e11a24-1725dc-b6d600a@10.96.109.11
   State of the call       : STATE_ACTIVE (7)
   Substate of the call    : SUBSTATE_NONE (0)
   Calling Number          : 88602243
   Called Number           : 00040756039061
   Called URI              : sip:00040756039061@10.75.32.241:5060
   Bit Flags               : 0xC0401E 0x10000000 0x4
   CC Call ID              : 563
   Source IP Address (Sig ): 10.75.32.241
   Destn SIP Req Addr:Port : [10.96.109.11]:5060
   Destn SIP Resp Addr:Port: [10.96.109.11]:58729
   Destination Name        : 10.96.109.11
   Number of Media Streams : 1
   Number of Active Streams: 1
   RTP Fork Object         : 0x0
   Media Mode              : flow-through
   Media Stream 1
     State of the stream      : STREAM_ACTIVE
     Stream Call ID           : 563
     Stream Type              : voice+dtmf (0)
     Stream Media Addr Type   : 1
     Negotiated Codec         : g711alaw (160 bytes)
     Codec Payload Type       : 8
     Negotiated Dtmf-relay    : rtp-nte
     Dtmf-relay Payload Type  : 101
     QoS ID                   : -1
     Local QoS Strength       : BestEffort
     Negotiated QoS Strength  : BestEffort
     Negotiated QoS Direction : None
     Local QoS Status         : None
     Media Source IP Addr:Port: [10.75.32.241]:16532
     Media Dest IP Addr:Port  : [10.75.32.241]:16530


Options-Ping    ENABLED:NO    ACTIVE:NO
   Number of SIP User Agent Server(UAS) calls: 1

This is my first CUBE configuration and I'm a bit lost, anyone have any idea what i'm doing wrong?

Thanks in advance :)

[Ayodeji Okanlawon]

It is possible that your sip packets are not been sourced from the correct interface..

Media Source IP Addr:Port: [192.168.10.2]:16534
     Media Dest IP Addr:Port  : [193.246.244.68]:48808

Can the ip address on 192.168.10.2 reach the ip 193.246.244.68 ?

How is your CUBE setup? Do you have two itnerfaces? Which interface connects to your CUCM and which connects to your ITSP..

Please post a sh run here

[Hunor Szombatfalvi]

192.168.10.2 is my gi0/2 interface and 193.246.244.68 is somewhere in the ITSP network but is reachable.

interface Loopback0
 ip address 10.75.32.241 255.255.255.240
!
interface Embedded-Service-Engine0/0
 no ip address
 shutdown
!
interface GigabitEthernet0/0
 description primary
 backup interface GigabitEthernet0/1
 ip address 10.75.32.130 255.255.255.192
 ip pim sparse-mode
 duplex auto
 speed auto
 service-policy output VOIP
!
interface GigabitEthernet0/1
 description back up
 ip address 10.75.32.130 255.255.255.192
 ip pim sparse-mode
 duplex auto
 speed auto
 service-policy output VOIP
!
interface GigabitEthernet0/2
 ip address 192.168.10.2 255.255.255.0
 duplex auto
 speed auto
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 10.75.32.129
ip route 192.168.10.0 255.255.255.0 192.168.10.1
ip route 193.246.242.0 255.255.255.0 192.168.10.1

(there's a full runing config in the original post)

[Ayodeji Okanlawon]

Thanks for the update. I can see that your configuration looks correct. So the 10.75.32.241 is the interface that connects to your cucm and the 192.168.10.2 is the one that connects to your ITSP.

Lets start by looking at your the sip messages..

Please do a test call and send the ff:

debug ccsip messages

(include calling and called number)

[Hunor Szombatfalvi]

here's a debug ccsip messages from internal phone to external mobile:

Nov 19 21:17:31.794 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:00040756039061@10.75.32.241:5060 SIP/2.0
Via: SIP/2.0/TCP 10.96.109.11:5060;branch=z9hG4bK3f5649204c1e38
From: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;tag=33493864~e09d3604-04c0-45ad-9523-7316cdf33dab-56965700
To: <sip:00040756039061@10.75.32.241>
Date: Thu, 19 Nov 2015 21:17:31 GMT
Call-ID: f0562d00-64e13c6b-172a5b-b6d600a@10.96.109.11
Supported: timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM10.5
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 101 INVITE
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Cisco-Guid: 4032179456-0000065536-0000088650-0191717386
Session-Expires:  1800
P-Asserted-Identity: "don't pick up, we're testing" <sip:88602243@10.96.109.11>
Remote-Party-ID: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;party=calling;screen=yes;privacy=off
Contact: <sip:88602243@10.96.109.11:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 204

v=0
o=CiscoSystemsCCM-SIP 33493864 1 IN IP4 10.96.109.11
s=SIP Call
c=IN IP4 10.75.32.241
t=0 0
m=audio 16446 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Nov 19 21:17:31.806 UTC: //162/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:40756039061;phone-context=international@193.246.242.84:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK4EF7F
From: "don't pick up, we're testing" <sip:564044320@sbc01.ch>;tag=13882C-1B0C
To: <sip:40756039061;phone-context=international@193.246.242.84>
Date: Thu, 19 Nov 2015 21:17:31 GMT
Call-ID: C79EC9A8-8E3911E5-80CABD72-5A9C1A9B@sbc01.ch
Supported: 100rel,timer,resource-priority,replaces,histinfo
Min-SE:  1800
Cisco-Guid: 4032179456-0000065536-0000088650-0191717386
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1447967851
Contact: <sip:564044320@192.168.10.2:5060>
History-Info: <sip:00040756039061@193.246.242.84:5060>;index=1
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
P-Asserted-Identity: "don't pick up, we're testing" <sip:564044320@sbc01.ch>
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 9645 2729 IN IP4 192.168.10.2
s=SIP Call
c=IN IP4 192.168.10.2
t=0 0
m=audio 16450 RTP/AVP 8 101
c=IN IP4 192.168.10.2
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Nov 19 21:17:31.806 UTC: //161/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 10.96.109.11:5060;branch=z9hG4bK3f5649204c1e38
From: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;tag=33493864~e09d3604-04c0-45ad-9523-7316cdf33dab-56965700
To: <sip:00040756039061@10.75.32.241>
Date: Thu, 19 Nov 2015 21:17:31 GMT
Call-ID: f0562d00-64e13c6b-172a5b-b6d600a@10.96.109.11
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Length: 0


Nov 19 21:17:31.822 UTC: //162/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
From: "don't pick up, we're testing"<sip:564044320@sbc01.ch>;tag=13882C-1B0C
To: <sip:40756039061;phone-context=international@193.246.242.84>
Call-ID: C79EC9A8-8E3911E5-80CABD72-5A9C1A9B@sbc01.ch
CSeq: 101 INVITE
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK4EF7F
Content-Length: 0


Nov 19 21:17:31.902 UTC: //162/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Description
From: "don't pick up, we're testing"<sip:564044320@193.246.242.84>;tag=13882C-1B0C
To: <sip:40756039061;phone-context=international@193.246.242.84>;tag=75561
Call-ID: C79EC9A8-8E3911E5-80CABD72-5A9C1A9B@sbc01.ch
CSeq: 101 INVITE
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK4EF7F
Content-Type: application/sdp
Contact: <sip:40756039061@193.246.242.84:5060>
User-Agent:  Nortel SESM 17.0.7.13
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin
Timestamp: 1447967851
Content-Length: 287

v=0
o=PVG 562979328 562979328 IN IP4 193.246.244.68
s=-
c=IN IP4 193.246.244.68
t=0 0
a=sqn:0
a=cdsc: 1 image udptl t38
a=cpar: a=T38FaxVersion:0
a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
m=audio 45364 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Nov 19 21:17:31.906 UTC: //161/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 10.96.109.11:5060;branch=z9hG4bK3f5649204c1e38
From: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;tag=33493864~e09d3604-04c0-45ad-9523-7316cdf33dab-56965700
To: <sip:00040756039061@10.75.32.241>;tag=138894-1381
Date: Thu, 19 Nov 2015 21:17:31 GMT
Call-ID: f0562d00-64e13c6b-172a5b-b6d600a@10.96.109.11
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:00040756039061@10.75.32.241:5060;transport=tcp>
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 5947 1996 IN IP4 10.75.32.241
s=SIP Call
c=IN IP4 10.75.32.241
t=0 0
m=audio 16448 RTP/AVP 8 101
c=IN IP4 10.75.32.241
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Nov 19 21:17:31.906 UTC: //161/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
UPDATE sip:88602243@10.96.109.11:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.75.32.241:5060;branch=z9hG4bK4F249E
From: <sip:00040756039061@10.75.32.241>;tag=138894-1381
To: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;tag=33493864~e09d3604-04c0-45ad-9523-7316cdf33dab-56965700
Date: Thu, 19 Nov 2015 21:17:31 GMT
Call-ID: f0562d00-64e13c6b-172a5b-b6d600a@10.96.109.11
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3
Max-Forwards: 70
Supported: timer,resource-priority,replaces,histinfo
Timestamp: 1447967851
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 UPDATE
Contact: <sip:00040756039061@10.75.32.241:5060;transport=tcp>
Min-SE:  1800
Content-Length: 0


Nov 19 21:17:31.942 UTC: //161/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.75.32.241:5060;branch=z9hG4bK4F249E
From: <sip:00040756039061@10.75.32.241>;tag=138894-1381
To: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;tag=33493864~e09d3604-04c0-45ad-9523-7316cdf33dab-56965700
Date: Thu, 19 Nov 2015 21:17:31 GMT
Call-ID: f0562d00-64e13c6b-172a5b-b6d600a@10.96.109.11
Server: Cisco-CUCM10.5
CSeq: 101 UPDATE
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
P-Asserted-Identity: "don't pick up, we're testing" <sip:88602243@10.96.109.11>
Remote-Party-ID: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;party=called;screen=yes;privacy=off
Contact: <sip:88602243@10.96.109.11:5060;transport=tcp>
Content-Length: 0


Nov 19 21:17:40.938 UTC: //162/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
From: "don't pick up, we're testing"<sip:564044320@193.246.242.84>;tag=13882C-1B0C
To: <sip:40756039061;phone-context=international@193.246.242.84>;tag=75561
Call-ID: C79EC9A8-8E3911E5-80CABD72-5A9C1A9B@sbc01.ch
CSeq: 101 INVITE
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK4EF7F
Content-Type: application/sdp
Contact: <sip:40756039061@193.246.242.84:5060>
User-Agent:  Nortel SESM 17.0.7.13
Supported: tdialog,replaces
Allow: INVITE,BYE,CANCEL,ACK,REGISTER,SUBSCRIBE,NOTIFY,MESSAGE,INFO,REFER,OPTIONS,PUBLISH,PRACK
Require: timer
Timestamp: 1447967851
Session-Expires: 1800;refresher=uac
Content-Length: 287

v=0
o=PVG 562979328 562979328 IN IP4 193.246.244.68
s=-
c=IN IP4 193.246.244.68
t=0 0
a=sqn:0
a=cdsc: 1 image udptl t38
a=cpar: a=T38FaxVersion:0
a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
m=audio 45364 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Nov 19 21:17:40.946 UTC: //161/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.96.109.11:5060;branch=z9hG4bK3f5649204c1e38
From: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;tag=33493864~e09d3604-04c0-45ad-9523-7316cdf33dab-56965700
To: <sip:00040756039061@10.75.32.241>;tag=138894-1381
Date: Thu, 19 Nov 2015 21:17:31 GMT
Call-ID: f0562d00-64e13c6b-172a5b-b6d600a@10.96.109.11
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:40756039061@10.75.32.241:5060;transport=tcp>
Supported: replaces
Server: Cisco-SIPGateway/IOS-15.4.3.M3
Session-Expires:  1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247

v=0
o=CiscoSystemsSIP-GW-UserAgent 5947 1996 IN IP4 10.75.32.241
s=SIP Call
c=IN IP4 10.75.32.241
t=0 0
m=audio 16448 RTP/AVP 8 101
c=IN IP4 10.75.32.241
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Nov 19 21:17:40.946 UTC: //162/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:40756039061@193.246.242.84:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.2:5060;branch=z9hG4bK5120A2
From: "don't pick up, we're testing" <sip:88602243@sbc01.ch>;tag=13882C-1B0C
To: <sip:40756039061;phone-context=international@193.246.242.84>;tag=75561
Date: Thu, 19 Nov 2015 21:17:31 GMT
Call-ID: C79EC9A8-8E3911E5-80CABD72-5A9C1A9B@sbc01.ch
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0


Nov 19 21:17:40.982 UTC: //161/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:40756039061@10.75.32.241:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.96.109.11:5060;branch=z9hG4bK3f564b6f66c193
From: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;tag=33493864~e09d3604-04c0-45ad-9523-7316cdf33dab-56965700
To: <sip:00040756039061@10.75.32.241>;tag=138894-1381
Date: Thu, 19 Nov 2015 21:17:31 GMT
Call-ID: f0562d00-64e13c6b-172a5b-b6d600a@10.96.109.11
User-Agent: Cisco-CUCM10.5
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence
Content-Length: 0


Nov 19 21:17:42.802 UTC: //162/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Received:
BYE sip:564044320@192.168.10.2:5060 SIP/2.0
From: <sip:40756039061;phone-context=international@193.246.242.84>;tag=75561
To: "don't pick up, we're testing"<sip:564044320@193.246.242.84>;tag=13882C-1B0C
Call-ID: C79EC9A8-8E3911E5-80CABD72-5A9C1A9B@sbc01.ch
CSeq: 102 BYE
Via: SIP/2.0/UDP 193.246.242.84:5060;branch=z9hG4bK-101d17c-ef1a4cb8-2c07319a
User-Agent:  Nortel SESM 17.0.7.13
Max-Forwards: 20
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin
P-Charging-Vector: icid-value=8003c91215b5d05f11fffa06466eaa2789e6f5a;icid-generated-at=193.246.242.84
Content-Length: 0


Nov 19 21:17:42.806 UTC: //161/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:88602243@10.96.109.11:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 10.75.32.241:5060;branch=z9hG4bK526FC
From: <sip:00040756039061@10.75.32.241>;tag=138894-1381
To: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;tag=33493864~e09d3604-04c0-45ad-9523-7316cdf33dab-56965700
Date: Thu, 19 Nov 2015 21:17:40 GMT
Call-ID: f0562d00-64e13c6b-172a5b-b6d600a@10.96.109.11
User-Agent: Cisco-SIPGateway/IOS-15.4.3.M3
Max-Forwards: 70
Timestamp: 1447967862
CSeq: 102 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=542,OS=86720,PR=514,OR=82240,PL=0,JI=0,LA=0,DU=1
Content-Length: 0


Nov 19 21:17:42.806 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 193.246.242.84:5060;branch=z9hG4bK-101d17c-ef1a4cb8-2c07319a
From: <sip:40756039061;phone-context=international@193.246.242.84>;tag=75561
To: "don't pick up, we're testing" <sip:88602243@sbc01.ch>;tag=13882C-1B0C
Date: Thu, 19 Nov 2015 21:17:42 GMT
Call-ID: C79EC9A8-8E3911E5-80CABD72-5A9C1A9B@sbc01.ch
Server: Cisco-SIPGateway/IOS-15.4.3.M3
CSeq: 102 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=514,OS=82240,PR=542,OR=86720,PL=0,JI=0,LA=0,DU=1
Content-Length: 0


Nov 19 21:17:42.842 UTC: //161/F0562D000001/SIP/Msg/ccsipDisplayMsg:
Received:

IACHZURVGW01#SIP/2.0 200 OK
Via: SIP/2.0/TCP 10.75.32.241:5060;branch=z9hG4bK526FC
From: <sip:00040756039061@10.75.32.241>;tag=138894-1381
To: "don't pick up, we're testing" <sip:88602243@10.96.109.11>;tag=33493864~e09d3604-04c0-45ad-9523-7316cdf33dab-56965700
Date: Thu, 19 Nov 2015 21:17:42 GMT
Call-ID: f0562d00-64e13c6b-172a5b-b6d600a@10.96.109.11
Server: Cisco-CUCM10.5
CSeq: 102 BYE
Content-Length: 0

[Ayodeji Okanlawon · Accepted solution]

Okay before we go any further..Here is what we see...its exactly what we have in the show sip-ua command..

Leg 1 of the call

Iinternal ip phone is sending media to10.75.32.241 (it looks like cucm is invoking a xcoder/mtp for this call), the CUBE is also sending its media to the xcoder which is itself.

leg 2 of the call..

CUBE is sending media to 193.246.244.68 and ITSP is sending media to 192.168.10.2

So here are the questions before we start taking packet captures..

1. Is 192.168.10.2 reachable from 193.246.244.68 (ITSP)

2. Is the CUBE IP 10.75.32.241 reachable from IP phone subnet..

If the answer is yes, then please do the following to take packet captures from the CUBE..

1.       

. Configure capture profile

               !

               ip traffic-export profile TAC mode capture

               bidirectional

               !

               interface gig0/2  ----> Interface which routes the traffic to ITSP

               ip traffic-export apply TAC 99999999

2. Capture traffic with these exec (enable) level commands

Note: The exec cmds don’t appear until a profile has been configured

router#traffic-export interface fa0/0 clear

router#traffic-export interface fa0/0 start

router#traffic-export interface fa0/0 stop

3. Export the pcap file to a server

router#traffic-export interface fa0/0 copy ftp://x.x.x.x/capture.pcap

1. Configure the logging buffer

                                Router#configuration terminal

                                Router(config)#no logging console

                                Router(config)#service timestamps debug datetime msec

                                Router(config)#logging buffered 30000000 debugging

                                Router(config)#service sequence

                                Router(config)#no logging rate-limit

                                Router(config)#exit

                                Router#

2. Enable the debugs below

Debug voip ccapi inout

Debug ccsip message

Debug ccsip error

3. Do “clear log”.
4. Make a test call.
5. After 3 minutes, stop the debugs by entering “undebug all”
6. Collect the output by entering “show log”

Please attach the log and the pcap file..

NB: we may need to also capture traffic from CUCM side later.

[Hunor Szombatfalvi]

Ok, I solved it, the #%$!@ gave me the wrong IP :) 

As you can see from my dial-peers I was meant to send traffic to 193.246.242.84, I saw from the debug commands that I was getting traffic back from a different IP but at first glance it looked like it was from the same subnet. I couldn't test this since all the provider devices seem to have ICMP disabled so I couldn't ping.

Only now did I notice that 193.246.244.68 is actually from a different subnet. I added a route, everything works =/

[Ayodeji Okanlawon]

That is good to know. This is why I asked if that IP was reachable. Feel free to mark the thread as answered to help others in the future. 

[Hunor Szombatfalvi]

could you please copy my post with the resolution and re-post it? i can't mark my own stuff as "correct answer"

[Ayodeji Okanlawon]

I am sure of my response above helped you to see where the issue is. That may just do the trick. If not, then don't worry about it. Enjoy your day and please come back more often as we are more than happy to help