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CUBE SIP Trunk outbound call failing

uday kiran
Level 1
Level 1

Hello,

 i am working on getting new sip trunk up. I am able to make inbound calls but outbound calls are failing with cube sending CANCEL message.

Any help is much appreciated.

CUBE WAN : 192.168.1.2

CUBE LAN - 10.202.108.6

CUCM : 10.202.108.11

Here is the configuration

boot-start-marker
boot-end-marker
!
!
vrf definition Mgmt-intf
!
address-family ipv4
exit-address-family
!
address-family ipv6
exit-address-family
!
enable secret 5 $1$Qkte$i56jckul5oeJRVsJpVH2H1
!
aaa new-model
!
!
aaa authentication login aaa local
aaa authorization exec default local if-authenticated
!
!
!
!
!
aaa session-id common
clock timezone CST -6 0
clock summer-time CDT recurring
!
!
!
!
!
!
!
!
!
!
!


ip name-server 10.202.101.33

ip domain name Transcore.com
!
!
!
!
!
!
!
!
!
!
subscriber templating
multilink bundle-name authenticated
!
!
!
!
!
!
!
!
!
voice service voip
ip address trusted list
ipv4 0.0.0.0
ipv4 10.202.108.11
ipv4 192.168.1.1
address-hiding
mode border-element
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no fax-relay sg3-to-g3
sip
bind control source-interface GigabitEthernet0/0/1
bind media source-interface GigabitEthernet0/0/1
min-se 300 session-expires 300
header-passing
error-passthru
asserted-id pai
privacy pstn
early-offer forced
midcall-signaling passthru
!
voice class media 1
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
!
!
!
!
!
!
!
voice-card 0/1
no watchdog
!
voice-card 0/4
dsp services dspfarm
no watchdog
!
license udi pid ISR4331/K9 sn FDO193623S1
!
spanning-tree extend system-id
!
username uday privilege 15 secret 5 $1$NhjN$byaMfU4pQhjpZi8wTWikb.
username rogersj privilege 15 secret 5 $1$5wfw$s7/xU3Ik86QJAeJIAYmJq0
username obriene privilege 15 secret 5 $1$ftbO$ed4Qm.uFwAB83SMs2NoGu.
!
redundancy
mode none
!
!
vlan internal allocation policy ascending
!
!
!
!
!
!
interface Loopback0
no ip address
!
interface GigabitEthernet0/0/0
description Link-To-Adtran
ip address 192.168.1.2 255.255.255.0
negotiation auto
!
interface GigabitEthernet0/0/1
description Link-To-Core-Sw
ip address 10.202.108.6 255.255.255.0
negotiation auto
!
interface GigabitEthernet0/0/2
no ip address
negotiation auto
!
interface Service-Engine0/1/0
!
interface Service-Engine0/4/0
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
ip address 10.202.105.6 255.255.255.0
negotiation auto
!
interface Vlan1
no ip address
shutdown
!
ip forward-protocol nd
no ip http server
no ip http secure-server
ip tftp source-interface GigabitEthernet0
ip route 0.0.0.0 0.0.0.0 10.202.108.2
ip ssh version 2
ip ssh dh min size 2048
!
!
!
!
!
!
!
control-plane
!
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/1/2
!
voice-port 0/1/3
!
voice-port 0/1/4
!
voice-port 0/1/5
!
!
!
!
!
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 GigabitEthernet0/0/1
sccp ccm 10.202.108.11 identifier 2 priority 2 version 7.0
sccp ccm 10.202.108.21 identifier 1 priority 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 3 register MN-transocder
associate profile 1 register MN-Conferencing
!
!
!
dspfarm profile 3 transcode
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 20
associate application SCCP
!
dspfarm profile 1 conference
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 2
associate application SCCP
!
dial-peer voice 1 voip
preference 1
destination-pattern 65136332..
session protocol sipv2
session target ipv4:10.202.108.11
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 2 voip
preference 2
destination-pattern 65136332..
session protocol sipv2
session target ipv4:10.202.108.21
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 200 voip
session protocol sipv2
session target sip-server
incoming called-number .%
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 10 voip
destination-pattern 1[2-9]..[2-9]......
session protocol sipv2
session target sip-server
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
codec g711ulaw
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
!
sip-ua
retry invite 2
timers trying 100
sip-server ipv4:192.168.1.1:5060
!
!
line con 0
stopbits 1
line aux 0
stopbits 1
line vty 0 4
transport input ssh
line vty 5 15
transport input ssh

Here are the logs

R

eceived:
INVITE sip:19728342123@10.202.108.6:5060 SIP/2.0
Via: SIP/2.0/TCP 10.202.108.21:5060;branch=z9hG4bK1c131cc6e74
From: "Phone Test" <sip:6513633271@10.202.108.21>;tag=1394~badd9444-c4ac-48a0-81f5-ad86ecfcaa71-44642085
To: <sip:19728342123@10.202.108.6>
Date: Wed, 03 Feb 2016 17:19:59 GMT
Call-ID: 58e08f00-6b2136bf-99-156cca0a@10.202.108.21
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, kpml
Supported: X-cisco-srtp-fallback,X-cisco-original-called
Call-Info: <urn:x-cisco-remotecc:callinfo>; security= Unknown; gci= 2-70; isVoip, <sip:10.202.108.21:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Cisco-Guid: 1491111680-0000065536-0000000018-0359451146
Session-Expires: 1800
P-Asserted-Identity: "Phone Test" <sip:6513633271@10.202.108.21>
Remote-Party-ID: "Phone Test" <sip:6513633271@10.202.108.21>;party=calling;screen=yes;privacy=off
Contact: <sip:6513633271@10.202.108.21:5060;transport=tcp>
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 412

v=0
o=CiscoSystemsCCM-SIP 1394 1 IN IP4 10.202.108.21
s=SIP Call
c=IN IP4 10.202.112.64
b=TIAS:64000
b=AS:64
t=0 0
m=audio 28346 RTP/AVP 9 124 0 8 116 18 101
a=rtpmap:9 G722/8000
a=rtpmap:124 iSAC/16000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=maxptime:20
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Feb 3 11:19:59.126 CST: //48/58E08F000000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:19728342123@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2B2E6
Remote-Party-ID: "Phone Test" <sip:6513633271@192.168.1.2>;party=calling;screen=yes;privacy=off
From: "Phone Test" <sip:6513633271@192.168.1.2>;tag=3F6B85C6-1079
To: <sip:19728342123@192.168.1.1>
Date: Wed, 03 Feb 2016 17:19:59 GMT
Call-ID: 2FC1B4B9-C9D111E5-8099B878-6AC79315@192.168.1.2
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1491111680-0000065536-0000000018-0359451146
User-Agent: Cisco-SIPGateway/IOS-15.5.3.S1a
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1454519999
Contact: <sip:6513633271@192.168.1.2:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 68
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 243

v=0
o=CiscoSystemsSIP-GW-UserAgent 1302 7309 IN IP4 192.168.1.2
s=SIP Call
c=IN IP4 192.168.1.2
t=0 0
m=audio 8094 RTP/AVP 0 101
c=IN IP4 192.168.1.2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

Feb 3 11:19:59.151 CST: //48/58E08F000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
From: "Phone Test"<sip:6513633271@192.168.1.2>;tag=3F6B85C6-1079
To: <sip:19728342123@192.168.1.1>
Call-ID: 2FC1B4B9-C9D111E5-8099B878-6AC79315@192.168.1.2
CSeq: 101 INVITE
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2B2E6
Timestamp: 1454519999
Supported: replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_NetVanta_3430/R11.4.3.V (s00a0c8dd7cea)
Contact: <sip:19728342123@192.168.1.1:5060;transport=UDP>
Content-Length: 0


Feb 3 11:20:00.761 CST: //48/58E08F000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
From: "Phone Test"<sip:6513633271@192.168.1.2>;tag=3F6B85C6-1079
To: <sip:19728342123@192.168.1.1>;tag=73485f0-7f000001-13c4-1379cd-658d40be-1379cd
Call-ID: 2FC1B4B9-C9D111E5-8099B878-6AC79315@192.168.1.2
CSeq: 101 INVITE
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2B2E6
Contact: <sip:19728342123@192.168.1.1:5060;transport=UDP>
Supported: replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_NetVanta_3430/R11.4.3.V (s00a0c8dd7cea)
Content-Type: application/sdp
Content-Length: 211

v=0
o=BroadWorks 244870600 1 IN IP4 192.168.1.1
s=-
c=IN IP4 192.168.1.1
t=0 0
m=audio 10088 RTP/AVP 0 101
a=ptime:20
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Feb 3 11:20:01.500 CST: //48/58E08F000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 183 Session Progress
From: "Phone Test"<sip:6513633271@192.168.1.2>;tag=3F6B85C6-1079
To: <sip:19728342123@192.168.1.1>;tag=73485f0-7f000001-13c4-1379cd-658d40be-1379cd
Call-ID: 2FC1B4B9-C9D111E5-8099B878-6AC79315@192.168.1.2
CSeq: 101 INVITE
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2B2E6
Contact: <sip:19728342123@192.168.1.1:5060;transport=UDP>
Supported: replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_NetVanta_3430/R11.4.3.V (s00a0c8dd7cea)
Content-Type: application/sdp
Content-Length: 211

v=0
o=BroadWorks 244870600 1 IN IP4 192.168.1.1
s=-
c=IN IP4 192.168.1.1
t=0 0
m=audio 10088 RTP/AVP 0 101
a=ptime:20
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

Feb 3 11:20:04.123 CST: //48/58E08F000000/SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:19728342123@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2B2E6
From: "Phone Test" <sip:6513633271@192.168.1.2>;tag=3F6B85C6-1079
To: <sip:19728342123@192.168.1.1>
Date: Wed, 03 Feb 2016 17:19:59 GMT
Call-ID: 2FC1B4B9-C9D111E5-8099B878-6AC79315@192.168.1.2
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1454520004
Reason: Q.850;cause=38
Content-Length: 0


Feb 3 11:20:04.134 CST: //48/58E08F000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
From: "Phone Test"<sip:6513633271@192.168.1.2>;tag=3F6B85C6-1079
To: <sip:19728342123@192.168.1.1>;tag=73485f0-7f000001-13c4-1379cd-658d40be-1379cd
Call-ID: 2FC1B4B9-C9D111E5-8099B878-6AC79315@192.168.1.2
CSeq: 101 CANCEL
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2B2E6
Supported: replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_NetVanta_3430/R11.4.3.V (s00a0c8dd7cea)
Content-Length: 0


Feb 3 11:20:04.139 CST: //48/58E08F000000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 487 Request Terminated
From: "Phone Test"<sip:6513633271@192.168.1.2>;tag=3F6B85C6-1079
To: <sip:19728342123@192.168.1.1>;tag=73485f0-7f000001-13c4-1379cd-658d40be-1379cd
Call-ID: 2FC1B4B9-C9D111E5-8099B878-6AC79315@192.168.1.2
CSeq: 101 INVITE
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2B2E6
Supported: replaces
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER
User-Agent: ADTRAN_NetVanta_3430/R11.4.3.V (s00a0c8dd7cea)
Content-Length: 0


Feb 3 11:20:04.140 CST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:19728342123@192.168.1.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK2B2E6
From: "Phone Test" <sip:6513633271@192.168.1.2>;tag=3F6B85C6-1079
To: <sip:19728342123@192.168.1.1>;tag=73485f0-7f000001-13c4-1379cd-658d40be-1379cd
Date: Wed, 03 Feb 2016 17:19:59 GMT
Call-ID: 2FC1B4B9-C9D111E5-8099B878-6AC79315@192.168.1.2
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

1 Reply 1

Vivek Batra
VIP Alumni
VIP Alumni

Hi,

When provider is sending 183 Session Progress, this message is not sent further to CUCM by CUBE and most probable reason is incorrect dial peer selection and SIP bind command.

As you can see that only dial-peer 200 is configured with incoming called number command and bind with 192.168.1.2 interface, this means that when CUCM sends INVITE to CUBE, this dial peer will be selected however on this case leg, we want to use 10.202.108.6 interface.

You need to use more specific incoming called-number command like as follows and add this command in either dial peer 1 or 2;

incoming called-number 1[2-9]..[2-9]......

Other thing which you would like to avoid is to use different transport protocols on both the call legs. As you can see CUCM call leg is on TCP whereas provider call leg is using UDP. This should not create any issue however I always avoid it unless not really required. I suggest you to change this under SIP Trunk Security Profile (in CUCM) and modify outgoing transport type to UDP.

Please share the result show it goes.

- Vivek