Re: CUBE 500 internal server error

EcstaticDuck · ‎08-11-2015

Hi,

I don't understand the following:

A customer with CUCM and CUBE can call any number but only one number of a number range, he can't reach. If he calls this number, the CUBE sends an "500 internal server error" to the CUCM after the 200OK from the called destination arrived at the external interface of the CUBE. Does anybody have an idea why this could happen?

Best regards

ecstaticduck

Chris Deren · ‎08-11-2015

Can you post "sh run" and "debug ccsip messages"?

EcstaticDuck · ‎08-12-2015

5393779: Jul 13 07:04:08.475 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:1234@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/TCP 5.6.7.8:5060;branch=z9hG4bK8f6d4b06f77e
From: "TnC" <sip:5678@5.6.7.8>;tag=486574~7a508d76-4fd0-4164-905a-6c6e841f23a1-53739009
To: <sip:1234@1.2.3.4>
Date: Mon, 13 Jul 2015 07:04:08 GMT
Call-ID: 11-12-13-14@5.6.7.8
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.1
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
Cisco-Guid: 1504618880-0000065536-0000015064-0856203274
Session-Expires: 1800
P-Asserted-Identity: "TnC" <sip:5678@5.6.7.8>
Remote-Party-ID: "TnC" <sip:5678@5.6.7.8>;party=calling;screen=yes;privacy=off
Contact: <sip:5678@5.6.7.8:5060;transport=tcp>
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 211

v=0
o=CiscoSystemsCCM-SIP 486574 1 IN IP4 5.6.7.8
s=SIP Call
c=IN IP4 1.2.3.4
t=0 0
m=audio 17316 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

5393780: Jul 13 07:04:08.487 UTC: //10356620/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:1234@10.20.30.40:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.5:5060;branch=z9hG4bK92AB5B128C
From: "TnC" <sip:5678;phone-context=national@rt01.ab.local>;tag=DDAD9628-1C77
To: <sip:1234;phone-context=national@10.20.30.40>
Date: Mon, 13 Jul 2015 07:04:08 GMT
Call-ID: 21-22-23-24@rt01.ab.local
Supported: 100rel,timer,resource-priority,replaces,histinfo
Min-SE: 1800
Cisco-Guid: 1504618880-0000065536-0000015064-0856203274
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1436771048
Contact: <sip:5678@1.2.3.5:5060>
History-Info: <sip:1234@10.20.30.40:5060>;index=1
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires: 1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 243

v=0
o=CiscoSystemsSIP-GW-UserAgent 543 4395 IN IP4 1.2.3.5
s=SIP Call
c=IN IP4 1.2.3.5
t=0 0
m=audio 31758 RTP/AVP 8 101
c=IN IP4 1.2.3.5
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

5393781: Jul 13 07:04:08.487 UTC: //10356619/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 5.6.7.8:5060;branch=z9hG4bK8f6d4b06f77e
From: "TnC" <sip:5678@5.6.7.8>;tag=486574~7a508d76-4fd0-4164-905a-6c6e841f23a1-53739009
To: <sip:1234@1.2.3.4>
Date: Mon, 13 Jul 2015 07:04:08 GMT
Call-ID: 11-12-13-14@5.6.7.8
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.3.3.M
Content-Length: 0

5393782: Jul 13 07:04:08.491 UTC: //10356620/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 100 Trying
From: "TnC"<sip:5678;phone-context=national@rt01.ab.local>;tag=DDAD9628-1C77
To: <sip:1234;phone-context=national@10.20.30.40>
Call-ID: 21-22-23-24@rt01.ab.local
CSeq: 101 INVITE
Via: SIP/2.0/UDP 1.2.3.5:5060;branch=z9hG4bK92AB5B128C
Content-Length: 0

5393792: Jul 13 07:04:08.915 UTC: //10356619/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 5.6.7.8:5060;branch=z9hG4bK8f6d4b06f77e
From: "TnC" <sip:5678@5.6.7.8>;tag=486574~7a508d76-4fd0-4164-905a-6c6e841f23a1-53739009
To: <sip:1234@1.2.3.4>;tag=DDAD97D8-995
Date: Mon, 13 Jul 2015 07:04:08 GMT
Call-ID: 11-12-13-14@5.6.7.8
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:1234@1.2.3.4:5060;transport=tcp>
Server: Cisco-SIPGateway/IOS-15.3.3.M
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 256

v=0
o=CiscoSystemsSIP-GW-UserAgent 5836 5971 IN IP4 1.2.3.4
s=SIP Call
c=IN IP4 1.2.3.4
t=0 0
m=audio 32614 RTP/AVP 8 101
c=IN IP4 1.2.3.4
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=label:Audio

5393793: Jul 13 07:04:09.271 UTC: //10356620/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 180 Ringing
From: "TnC"<sip:5678;phone-context=national@10.20.30.40>;tag=DDAD9628-1C77
To: <sip:1234;phone-context=national@10.20.30.40>;tag=75395
Call-ID: 21-22-23-24@rt01.ab.local
CSeq: 101 INVITE
Via: SIP/2.0/UDP 1.2.3.5:5060;branch=z9hG4bK92AB5B128C
Contact: <sip:1234@10.20.30.40:5060>
User-Agent: Nortel SESM 17.0.7.13
Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,gin
Timestamp: 1436771048
Content-Length: 0

5393794: Jul 13 07:04:09.271 UTC: //10356619/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 5.6.7.8:5060;branch=z9hG4bK8f6d4b06f77e
From: "TnC" <sip:5678@5.6.7.8>;tag=486574~7a508d76-4fd0-4164-905a-6c6e841f23a1-53739009
To: <sip:1234@1.2.3.4>;tag=DDAD97D8-995
Date: Mon, 13 Jul 2015 07:04:08 GMT
Call-ID: 11-12-13-14@5.6.7.8
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:01234@1.2.3.4:5060;transport=tcp>
Server: Cisco-SIPGateway/IOS-15.3.3.M
Content-Length: 0

5393799: Jul 13 07:04:15.635 UTC: //10356620/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
From: "TnC"<sip:5678;phone-context=national@10.20.30.40>;tag=DDAD9628-1C77
To: <sip:1234;phone-context=national@10.20.30.40>;tag=75395
Call-ID: 21-22-23-24@rt01.ab.local
CSeq: 101 INVITE
Via: SIP/2.0/UDP 1.2.3.5:5060;branch=z9hG4bK92AB5B128C
Content-Type: application/sdp
Contact: <sip:1234@10.20.30.40:5060;maddr=10.20.30.40>
User-Agent: Nortel SESM 17.0.7.13
Supported: replaces,tdialog
Allow: INVITE,BYE,CANCEL,ACK,REGISTER,SUBSCRIBE,NOTIFY,MESSAGE,INFO,REFER,OPTIONS,PUBLISH,PRACK
Require: timer
Timestamp: 1436771048
Session-Expires: 1800;refresher=uac
Content-Length: 223

v=0
o=- 5571 2 IN IP4 193.246.243.68
s=session
b=CT:1000
t=0 0
m=audio 45312 RTP/AVP 8 101
c=IN IP4 193.246.243.68
a=label:Audio
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

5393800: Jul 13 07:04:15.639 UTC: //10356619/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/TCP 5.6.7.8:5060;branch=z9hG4bK8f6d4b06f77e
From: "TnC" <sip:5678@5.6.7.8>;tag=486574~7a508d76-4fd0-4164-905a-6c6e841f23a1-53739009
To: <sip:1234@1.2.3.4>;tag=DDAD97D8-995
Call-ID: 11-12-13-14@5.6.7.8
CSeq: 101 INVITE
Content-Length: 0

5393801: Jul 13 07:04:15.639 UTC: //10356620/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:1234@10.20.30.40:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.5:5060;branch=z9hG4bK92AB5D4EB
From: "TnC" <sip:5678@rt01.ab.local>;tag=DDAD9628-1C77
To: <sip:1234;phone-context=national@10.20.30.40>;tag=75395
Date: Mon, 13 Jul 2015 07:04:08 GMT
Call-ID: 21-22-23-24@rt01.ab.local
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

5393802: Jul 13 07:04:15.659 UTC: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:1234@1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/TCP 5.6.7.8:5060;branch=z9hG4bK8f6d4b06f77e
From: "TnC" <sip:5678@5.6.7.8>;tag=486574~7a508d76-4fd0-4164-905a-6c6e841f23a1-53739009
To: <sip:1234@1.2.3.4>;tag=DDAD97D8-995
Date: Mon, 13 Jul 2015 07:04:08 GMT
Call-ID: 11-12-13-14@5.6.7.8
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: presence, kpml
Content-Length: 0

5393803: Jul 13 07:04:15.663 UTC: //10356620/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Sent:
BYE sip:1234@10.20.30.40:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.3.5:5060;branch=z9hG4bK92AB5E123F
From: "TnC" <sip:5678@rt01.ab.local>;tag=DDAD9628-1C77
To: <sip:1234;phone-context=national@10.20.30.40>;tag=75395
Date: Mon, 13 Jul 2015 07:04:08 GMT
Call-ID: 21-22-23-24@rt01.ab.local
User-Agent: Cisco-SIPGateway/IOS-15.3.3.M
Max-Forwards: 70
Timestamp: 1436771055
CSeq: 102 BYE
Reason: Q.850;cause=16
P-RTP-Stat: PS=332,OS=53120,PR=0,OR=0,PL=0,JI=0,LA=0,DU=0
Content-Length: 0

5393804: Jul 13 07:04:15.671 UTC: //10356620/59AEA9800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
From: "TnC"<sip:5678@rt01.ab.local>;tag=DDAD9628-1C77
To: <sip:1234;phone-context=national@10.20.30.40>;tag=75395
Call-ID: 21-22-23-24@rt01.ab.local
CSeq: 102 BYE
Via: SIP/2.0/UDP 1.2.3.5:5060;branch=z9hG4bK92AB5E123F
Content-Length: 0

Mohammed al Baqari · ‎08-12-2015

Hi,

Please share your config along with debug ccsip verb, debug voice ccapi, debug voice dialpeer

EcstaticDuck · ‎08-12-2015

no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec show-timezone
service timestamps log datetime msec show-timezone
service password-encryption
service sequence-numbers
!
hostname rt01
!
boot-start-marker
boot-end-marker
!
aqm-register-fnf
!
logging buffered 64000
no logging console
enable secret 4 XXYYZZ
!
aaa new-model
!
!
aaa authentication login VTY group radius local
aaa authentication login CONSOLE local
!
!
!
!
!
aaa session-id common

errdisable recovery cause udld
errdisable recovery cause bpduguard
errdisable recovery cause rootguard
errdisable recovery cause pagp-flap
errdisable recovery cause dtp-flap
errdisable recovery cause link-flap
errdisable recovery interval 180
!
no ip source-route
ip icmp rate-limit unreachable 1000
!
!
!
!
!
!
no ip domain lookup
ip domain name kunde.ab
ip cef
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice-card 0
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
no ip address trusted authenticate
address-hiding
mode border-element
allow-connections sip to sip
fax protocol pass-through g711alaw
sip
min-se 1100 session-expires 1100
source filter
no anat
early-offer forced
history-info
midcall-signaling passthru
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
voice class sip-profiles 110
request INVITE sip-header From modify "sip:([1-9]........[0-9].*)" "sip:000\1;"
request INVITE sip-header From modify "sip:([1-9].......[0-9])" "sip:00\1;"
request INVITE sip-header Remote-Party-ID modify "sip:([1-9]........[0-9].*)" "sip:000\1;"
request INVITE sip-header Remote-Party-ID modify "sip:([1-9].......[0-9])" "sip:00\1;"
request INVITE sip-header Contact modify "sip:([1-9]........[0-9].*)" "sip:000\1;"
request INVITE sip-header Contact modify "sip:([1-9].......[0-9])" "sip:00\1;"
!
voice class sip-profiles 100
request INVITE sip-header To modify "sip:0([1-9].*)@" "sip:\1;phone-context=national@"
request INVITE sip-header To modify "sip:00(.*)@" "sip:\1;phone-context=international@"
request INVITE sip-header History-Info modify "Reason=sip;" "Reason=sip%3b"
request INVITE sip-header History-Info modify "cause=" "cause%3d"
request INVITE sip-header History-Info modify "text=" "text%3d"
request INVITE sip-header History-Info modify "sip:00([1-9].*" "sip:\1;phone-context=national"
request INVITE sip-header P-Asserted-Identity modify "sip:zzzzz([4-8][0-9]..)@" "sip:81844\1;phone-context=national@"
request INVITE sip-header P-Asserted-Identity modify "sip:zzzzz(8[0-9]..)@" "sip:zzzzz\1;phone-context=national@"
request INVITE sip-header P-Asserted-Identity modify "sip:zzzzz(9[0-9]..)@" "sip:zzzzz\1;phone-context=national@"
request INVITE sip-header Diversion modify "sip:00(.*)@" "sip:\1;phone-context=national@"
request INVITE sip-header From modify "sip:(.........)@" "sip:\1;phone-context=national@"
request INVITE sip-header From modify "sip:(.........[0-9].*)@" "sip:\1;phone-context=international@"
request REINVITE sip-header SIP-Req-URI modify "anonymous" "zzzzzzzzzzz"
!
!
!
!
voice translation-rule 1
rule 1 /$.+$/ /0\1/
rule 2 /^$/ /xxxxxxxxxxx/
!
voice translation-rule 2
rule 1 /wwwww$[4-8]...$$/ /xxxxxxx\1/
rule 2 /wwwww$40[5-7].$/ /xxxxxxx\1/
rule 3 /wwwww$437[3-5]$/ /xxxxxxx\1/
!
voice translation-rule 3
rule 1 /^0/ //
!
voice translation-rule 4
rule 1 /^yy/ //
!
voice translation-rule 5
!
voice translation-rule 10
rule 1 /^yy/ //
rule 2 /000yy$.........$/ /\1/
rule 3 /000/ /00\1/
rule 4 /00yy$.......$/ /yy\1/
rule 5 /00/ /\1/
rule 6 /[+]yy$.........$/ /\1/
!
voice translation-rule 11
rule 1 /^yy/ //
rule 2 /000yy$a[a-c].......$/ /zzzzzzzz\1/
rule 3 /000yy$.........$/ /0\1/
rule 4 /000/ /00\1/
rule 5 /00yy$.......$/ /041\1/
rule 6 /00$a[a-c].......$/ /zzzzzzzz\1/
rule 7 /00/ /0\1/
!
voice translation-rule 30
rule 1 /000yy$a[a-c].......$/ /zzzzzzzz\1/
rule 2 /00$a[a-c].......$/ /zzzzzzzz\1/
rule 3 /^0/ //
!
!
voice translation-profile voip-gate_IN
translate calling 1
translate called 2
!
voice translation-profile voip-gate_OUT
translate calling 10
translate called 3
!
!
!
license udi pid CISCO2911/K9 sn FGL17431077
hw-module pvdm 0/0
!
!
!
object-group network Provider_Servers
ggg.hhh.iii.0 255.255.255.0
ggg.hhh.jjj.0 255.255.255.0
ggg.hhh.kkk.0 255.255.255.0
!
vtp domain kunde.ab
vtp mode transparent
username AABBCC secret 5 XXYYZZ
!
redundancy
!
!
!
!
!
ip tcp window-size 65535
ip tcp synwait-time 10
ip tcp path-mtu-discovery
ip tftp source-interface GigabitEthernet0/0
ip ssh time-out 30
ip ssh source-interface GigabitEthernet0/0
ip ssh version 2
!
!
!
!
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0

ip address 1.2.3.4 255.255.255.0
no ip redirects
no ip proxy-arp
duplex auto
speed auto
arp timeout 300
!
interface GigabitEthernet0/1

ip address 1.2.3.5 255.255.255.0
ip access-group Provider_IN in
duplex auto
speed auto
!

!

!
ip access-list extended Provider_IN
permit udp object-group Provider_Servers host 1.2.3.5
permit icmp any any
deny ip any any log
!
logging source-interface GigabitEthernet0/0
logging host sag.ich.sicher.nicht
!
!
!
!
!
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 GigabitEthernet0/0
sccp ccm cucm1 identifier 1 version 7.0
sccp ccm cucm2 identifier 2 version 7.0
sccp
!
sccp ccm group 1
bind interface GigabitEthernet0/0
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 2 register ABCD-rt01
associate profile 1 register MTP-rt01
!
dspfarm profile 2 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 12
associate application SCCP
!
dspfarm profile 1 mtp
codec g711alaw
maximum sessions software 150
associate application SCCP
!
dial-peer voice 10 voip
description *** SIP to Provider ***
translation-profile outgoing Provider_OUT
preference 1
destination-pattern 0T
session protocol sipv2
session target ipv4:10.20.30.40
session transport udp
voice-class sip localhost dns:rt01.ab.local
voice-class sip profiles 100
voice-class sip options-keepalive up-interval 30 down-interval 15 retry 3
dtmf-relay rtp-nte
codec g711alaw
fax protocol pass-through g711alaw
no vad
!
dial-peer voice 101 voip
description *** SIP to Primary CUCM ***
preference 1
destination-pattern zzzzzzz....
session protocol sipv2
session target ipv4:5.6.7.8
session transport udp
voice-class sip profiles 110
voice-class sip options-keepalive up-interval 30 down-interval 15 retry 3
dtmf-relay rtp-nte
codec g711alaw
fax rate 9600
fax protocol pass-through g711alaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 102 voip
description *** SIP to Secondary CUCM ***
preference 2
destination-pattern zzzzzzz....
session protocol sipv2
session target ipv4:5.6.7.7
session transport udp
voice-class sip profiles 110
voice-class sip options-keepalive up-interval 30 down-interval 15 retry 3
dtmf-relay rtp-nte
codec g711alaw
fax rate 9600
fax protocol pass-through g711alaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 100 voip
description *** SIP from Provider ***
translation-profile incoming Provider_IN
session protocol sipv2
session transport udp
incoming called-number zzzzz....
voice-class sip localhost dns:rt01.ab.local
dtmf-relay rtp-nte
codec g711alaw
fax rate 9600
fax protocol pass-through g711alaw
no vad
!
dial-peer voice 20 voip
description *** SIP from CUCM ***
session protocol sipv2
session transport udp
incoming called-number 0T
dtmf-relay rtp-nte
codec g711alaw
fax protocol pass-through g711alaw
no vad
!
dial-peer voice 110 voip
description *** SIP from Provider ***
translation-profile incoming Provider_IN
session protocol sipv2
session transport udp
incoming called-number zzzzzzz[a-c].
voice-class sip localhost dns:rt01.ab.local
dtmf-relay rtp-nte
codec g711alaw
fax rate 9600
fax protocol pass-through g711alaw
no vad
!
dial-peer voice 111 voip
description *** SIP to Primary CUCM ***
preference 1
destination-pattern zzzzzzzzz[a-c].
session protocol sipv2
session target ipv4:5.6.7.8
session transport udp
voice-class sip profiles 110
voice-class sip options-keepalive up-interval 30 down-interval 15 retry 3
dtmf-relay rtp-nte
codec g711alaw
fax rate 9600
fax protocol pass-through g711alaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 112 voip
description *** SIP to Secondary CUCM ***
preference 2
destination-pattern zzzzzzzzz[a-c].
session protocol sipv2
session target ipv4:5.6.7.7
session transport udp
voice-class sip profiles 110
voice-class sip options-keepalive up-interval 30 down-interval 15 retry 3
dtmf-relay rtp-nte
codec g711alaw
fax rate 9600
fax protocol pass-through g711alaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 120 voip
description *** SIP from Provider ***
translation-profile incoming Provider_IN
session protocol sipv2
session transport udp
incoming called-number zzzzzzzz[a-c]
voice-class sip localhost dns:rt01.ab.local
dtmf-relay rtp-nte
codec g711alaw
fax rate 9600
fax protocol pass-through g711alaw
no vad
!
dial-peer voice 121 voip
description *** SIP to Primary CUCM ***
preference 1
destination-pattern zzzzzzzzzz[a-c]
session protocol sipv2
session target ipv4:5.6.7.8
session transport udp
voice-class sip profiles 110
voice-class sip options-keepalive up-interval 30 down-interval 15 retry 3
dtmf-relay rtp-nte
codec g711alaw
fax rate 9600
fax protocol pass-through g711alaw
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 122 voip
description *** SIP to Secondary CUCM ***
preference 2
destination-pattern zzzzzzzzzz[a-c]
session protocol sipv2
session target ipv4:5.6.7.7
session transport udp
voice-class sip profiles 110
voice-class sip options-keepalive up-interval 30 down-interval 15 retry 3
dtmf-relay rtp-nte
codec g711alaw
fax rate 9600
fax protocol pass-through g711alaw
ip qos dscp cs3 signaling
no vad
!
!
gateway
timer receive-rtp 1200
!
sip-ua
no remote-party-id
retry invite 3
connection-reuse
!
!
!
gatekeeper
shutdown
!
!

Mohammed al Baqari · ‎08-12-2015

Hi,

I don't see bind interface command anyware. You need to configure bind interface command on dialpeers facing ITSP and dialpeers facing CUCM to use the right interfaces using the command 'voice-class sip bind control source-interface x/x'

Nadeem Ahmed · ‎08-11-2015

along with what Chris suggest, Please collect the debug voice ccapi inout as well. Make a test to the number which is not working.

enable these debugs and post the same.

Br,

Nadeem

Br, Nadeem Please rate all useful post.

pwenger · ‎09-04-2017

Hello

I have exactly the same problem and I wonder if you ever found a solution?

If so, I appricate to share the solution for this problem.

Thanks

Peter

Abhay Singh Reyal · ‎09-04-2017

Collect the debugs from the CUBE. Provide calling and called number with time stamp after reproducing the issue. Post the debugs and show run here. You can attach it on the post
Debugs :-
debug ccsip messages
debug voip ccapi inout

Regards
Abhay

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle

pwenger · ‎09-04-2017

Hello Abhay

Attached you find the traces you requested.

Regards

Peter

Abhay Singh Reyal · ‎09-04-2017

Make another test call with the working scenario and check the difference. As per this debug output:-
Incoming Dial-peer=201
Outgoing Dial-peer=100
++++
Sep 4 12:47:28: //1073714/6E2E0B800000/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=313067201
----- ccCallInfo IE subfields -----
cisco-ani=sip:313067201@172.29.10.20
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=0316333019

Invite sent to the ITSP has number as To: <sip:316333019;phone-context=national@193.246.242.84>
Make sure if this is correct or not.
Collect the debugs of a working scenario and post it here. Make sure to run both the debugs together on the same call.
PS:- Provide show run as well.

Regards
Abhay

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle

pwenger · ‎09-04-2017

Hello Abhay

Find attached the traces for the following calls:

working call:

calling number: 0313067201

called number: 0319972357

not working call:

calling number: 0313067201

called number: 0316333231

Interesting is, that there is an Voicemail behind the called number of the not working call and it's only not working for some numbers.

Attached you find a snipset of the configuration as well.

Thanks

Peter

Abhay Singh Reyal · ‎09-04-2017

Not working Invite sent to ITSP
++++++
Sent:
INVITE sip:0316333231@193.246.242.84:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.38.2:5060;branch=z9hG4bKA0E321344
From: <sip:313067201;phone-context=national@172.29.38.2>;tag=6B3FCDB8-C4F
To: <sip:316333231;phone-context=national@193.246.242.84>
Date: Mon, 04 Sep 2017 15:04:45 GMT
Call-ID: 39226386-90B911E7-B5B0C128-D2B7144D@172.29.38.2

Working Invite Sent to ITSP
+++++++
Sent:
INVITE sip:0319972356@193.246.242.84:5060 SIP/2.0
Via: SIP/2.0/UDP 172.29.38.2:5060;branch=z9hG4bKA0E0910A2
From: "4961 Nicola Tolic" <sip:313067201;phone-context=international;phone-context=national@172.29.38.2>;tag=6B31B10C-969
To: <sip:319972356;phone-context=national@193.246.242.84>
Date: Mon, 04 Sep 2017 14:49:21 GMT
Call-ID: 11E18DC1-90B711E7-B556C128-D2B7144D@172.29.38.2

Non Working 183 Session Progress
+++++++
Received:
SIP/2.0 183 Session Description
From: <sip:313067201;phone-context=national@193.246.242.84>;tag=6B3FCDB8-C4F
To: <sip:316333231;phone-context=national@193.246.242.84>;tag=49840
Call-ID: 39226386-90B911E7-B5B0C128-D2B7144D@172.29.38.2
CSeq: 101 INVITE
Via: SIP/2.0/UDP 172.29.38.2:5060;branch=z9hG4bKA0E321344
Content-Type: application/sdp
Contact: <sip:0316333231@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: 1504537485
Content-Length: 223

v=0
o=vop-q20-bei-00 12992 1 IN IP4 193.246.244.68
s=sip call
b=CT:1000
t=0 0
m=audio 44042 RTP/AVP 8 101
c=IN IP4 193.246.244.68

Working 183 Session Progress
+++++++
Received:
SIP/2.0 183 Session Description
From: "4961 Nicola Tolic"<sip:313067201;phone-context=national@193.246.242.84>;tag=6B31B10C-969
To: <sip:319972356;phone-context=national@193.246.242.84>;tag=56965
Call-ID: 11E18DC1-90B711E7-B556C128-D2B7144D@172.29.38.2
CSeq: 101 INVITE
Via: SIP/2.0/UDP 172.29.38.2:5060;branch=z9hG4bKA0E0910A2
Content-Type: application/sdp
Contact: <sip:0319972356@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: 1504536561
Content-Length: 289

v=0
o=PVG 1300786660 1300786660 IN IP4 193.246.244.75
s=-
c=IN IP4 193.246.244.75

----->>>> Difference is regarding the originator and session identifier which is different in both the cases [193.246.244.68 and 193.246.244.75]

Can you add below command and see if it makes any difference

voice service voip
sip
pass-thru content sdp

And also bind the correct interface with the command below
bind control source-interface Loopback or interface
bind media source-interface Loopback or interface

HTH
Regards
Abhay

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle

pwenger · ‎09-05-2017

Hello

As you mentioned in the previous post I applied the bind commands and the pass-thru on the CUBE.

Now I see no more the internal server error, but the caller has no ringbacktone.

For this I changed the SIP Rel1XX Options parameter from disabled to Send PRACK if 1XX contains SDP.

But it changed nothing, still no ringbacktone for certain dialed numbers.

Do you have an idea what to change?

Regards

Peter

Abhay Singh Reyal · ‎09-05-2017

Try removing the below command and see what do you get then, as the interface has been bind. Check if the call still works or not and if it does and still there is no ringback tone, provide the debugs as provided earlier.
voice service voip
sip
no pass-thru content sdp

PS :- We have come to the point now where CUBE is not sending 500 internal server error anymore.
Keep this thread posted after making the changes.

HTH
Regards
Abhay

Regards
Abhay Singh Reyal
The Only Way To Do Great Work Is To Love What You Do. If You Haven’t Found It Yet, Keep Looking. Don’t Settle