External calls to voicemail dropping after about 40 seconds while leaving a message with SIP trunks 1

b1-4431-a#show run
Building configuration...


Current configuration : 7830 bytes
!
! Last configuration change at 12:38:32 PDT Fri May 24 2019 by hardyr
! NVRAM config last updated at 15:57:34 PDT Tue May 28 2019 by hardyr
!
version 15.5
no service pad
service tcp-keepalives-in
service tcp-keepalives-out
service timestamps debug datetime msec localtime show-timezone
service timestamps log datetime msec localtime show-timezone
service password-encryption
service sequence-numbers
no platform punt-keepalive disable-kernel-core
!
hostname b1-4431-a
!
boot-start-marker
boot-end-marker
!
!
vrf definition Mgmt-intf
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address-family ipv4
exit-address-family
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address-family ipv6
exit-address-family
!
!
no aaa new-model
clock timezone PDT -8 0
clock summer-time PDT recurring
clock calendar-valid
!
!
!
ip name-server x.x.x.x x.x.x.x

ip domain name x
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!
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!
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!
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!
subscriber templating
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multilink bundle-name authenticated
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!
!
!
!
!
!
voice service voip
ip address trusted list
ipv4 10.129.193.10 255.255.255.255
ipv4 10.129.251.101 255.255.255.255
ipv4 10.129.193.11 255.255.255.255
ipv4 10.129.193.12 255.255.255.255
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
supplementary-service media-renegotiate
signaling forward unconditional
fax protocol pass-through g711ulaw
modem passthrough nse codec g711ulaw
h323
emptycapability
sip
rel1xx supported "rel100"
min-se 7200 session-expires 7200
header-passing
registrar server expires max 600 min 60
asserted-id pai
early-offer forced
midcall-signaling passthru
privacy-policy passthru
!
voice class codec 1
codec preference 1 g711ulaw
!
!
!
voice class server-group 1
ipv4 10.129.193.11 preference 1
ipv4 10.129.193.12 preference 2
ipv4 10.129.193.10 preference 3
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voice class server-group 2
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!
!
!
voice translation-rule 1
rule 1 /^9\(.*\)/ /+\1/
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voice translation-rule 2
rule 1 /^9\(.*\)/ /\1/
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!
voice translation-profile PSTN-OUT
translate called 1
!
voice translation-profile PSTN-OUT-2
translate called 2
!
!
!
!
voice-card 0/4
no watchdog
!
license udi pid xxxxxx
!
spanning-tree extend system-id
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username xxx privilege 15 secret 5 xxxx
username xxx privilege 15 secret 5 xxxx
username xxx privilege 15 secret 5 xxxxx
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redundancy
mode none
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vlan internal allocation policy ascending
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interface GigabitEthernet0/0/0
description --- 3750 port 43 ---
ip address 10.129.193.2 255.255.255.128
negotiation auto
!
interface GigabitEthernet0/0/1
description --- WindStream ---
ip address 10.129.251.100 255.255.255.254
negotiation auto
!
interface GigabitEthernet0/0/2
no ip address
shutdown
negotiation auto
!
interface GigabitEthernet0/0/3
no ip address
shutdown
negotiation auto
!
interface Service-Engine0/4/0
no ip address
shutdown
!
interface GigabitEthernet0
vrf forwarding Mgmt-intf
no ip address
shutdown
negotiation auto
!
interface Vlan1
no ip address
shutdown
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ip forward-protocol nd
no ip http server
no ip http secure-server
ip rtcp report interval 60000
ip route 0.0.0.0 0.0.0.0 10.129.193.1
ip ssh version 2
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!
!
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tftp-server bootflash:running-config
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control-plane
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!
!
!
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
!
!
!
!
dial-peer voice 2000 voip
description ***Inbound from PSTN - SIP; DNIS/DID***
session protocol sipv2
incoming called-number xxx.......
voice-class codec 1
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
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate disable
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 2001 voip
description ***Inbound from PSTN - SIP; DNIS/DID***
session protocol sipv2
incoming called-number xxx.......
voice-class codec 1
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
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate disable
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 1000 voip
description ***Inbound from CUCM Servers; PSTN Dialing ***
session protocol sipv2
incoming called-number ^9.T
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate disable
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 1100 voip
description *Outbound to Sub01,SUB02&PUB;DID/DNIS*
destination-pattern xxx.......
session protocol sipv2
session server-group 1
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate disable
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 1101 voip
description *Outbound to Sub01,SUB02&PUB;DID/DNIS*
destination-pattern xxx.......
session protocol sipv2
session server-group 1
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate disable
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 2091 voip
description ***Outbound to PSTN SIP; PSTN Dialing***
translation-profile outgoing PSTN-OUT-2
destination-pattern 9011T
session protocol sipv2
session target ipv4:10.129.251.101
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate disable
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 2092 voip
description ***Outbound to PSTN SIP; PSTN Dialing***
translation-profile outgoing PSTN-OUT-2
destination-pattern 9911
session protocol sipv2
session target ipv4:10.129.251.101
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate disable
fax protocol pass-through g711ulaw
no vad
!
dial-peer voice 2099 voip
description ***Outbound to PSTN SIP; PSTN Dialing***
translation-profile outgoing PSTN-OUT
destination-pattern 91[2-9].........
session protocol sipv2
session target ipv4:10.129.251.101
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate disable
fax protocol pass-through g711ulaw
no vad
!
!
gateway
media-inactivity-criteria all
timer receive-rtcp 5
timer receive-rtp 1200
!
sip-ua
retry invite 2
retry bye 2
retry cancel 2
timers trying 1000
timers expires 60000
timers connect 1000
timers buffer-invite 5000
reason-header override
g729-annexb override
!
!
line con 0
login local
stopbits 1
line aux 0
stopbits 1
line vty 0 4
login local
transport input ssh
!
ntp server x.x.x.x
!
end

b1-4431-a#
b1-4431-a#
b1-4431-a#
b1-4431-a#
b1-4431-a#

Have you done a SIP debug to see what the message is when the call is dropped?

Certifications:
A+
Network+
CCENT
CCNA Voice
CCNP Voice

No I have not but I will do that next. I am kind of new to Cisco after 20 years with Avaya.....would this be through the real time monitoring tool?

You could grab the CUCM logs from there yes. Or i would probably start at running debug ccsip messages on the router itself just to see if anything sticks out. If nothing did, then i would grab logs from CUCM and Unity and go from there.

Certifications:
A+
Network+
CCENT
CCNA Voice
CCNP Voice

Thank you so much gnrlash4life.

I just wanted to update you all, after running a debug it appears the our service provider is dropping the call after 60 seconds. We will now open a ticket with them and I will update this post when we get this resolved.

After running the ccsip messages debug and confirming our ip rtcp settings.

router(config)#ip rtcp report interval 60000
router(config)#gateway
router(config-gateway)#timer receive-rtcp 5

I opened a ticket with our SIP vendor uploaded the debug traces to convince the vendor to look at their gateways. They adjusted what I believe are the rtcp settings on their end and now everything is working fine.

Thank you gnrslash4life