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SIP-UA missing

LyRa
Level 1
Level 1

Hi all!

 

We have Cisco 881 routers at the branch offices and only on one of them we've got that problem when sip-ua suddenly disappears:

Annotation 2019-04-22 102839.png

Reboot solves the problem.
All configuration across routers is the same.
How can i troubleshoot this problem?

8 Replies 8

George Sotiropoulos
Cisco Employee
Cisco Employee

Hello,

Can you check if you use the same IOS as with the rest of the routers?

G

Please Rate Posts (by clicking on Star) and/or Mark Solutions as Accepted, when applies

Yep, it's 15.4(3)M7.

Hi,

Can you please post the sip-ua and dial-peers configuration here?

Thanks

 

Regards

 

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

Here is the sip-ua config:

Spoiler
RT-Novosibirsk#sh run | s sip-ua
sip-ua
credentials username 067000 password blablabla realm voip.uiscom.ru
authentication username 067000 password 7 blablabla
no remote-party-id
retry invite 3
retry response 3
retry bye 3
retry cancel 3
registrar ipv4:195.211.120.9:9060 expires 900
sip-server ipv4:195.211.120.9:9060
no suspend-resume

And dial-peer configuration:

Spoiler
voice translation-rule 6000
rule 1 /067000/ /9998/
voice translation-rule 67000
rule 1 /.*/ /067000/
voice translation-profile NOVOSIBIRSK-to-PROVIDER
translate calling 67000
voice translation-profile UCCX
translate called 6000
dial-peer voice 1002 voip
description -[OUTGOING TO PSTN]-
translation-profile outgoing NOVOSIBIRSK-to-PROVIDER
destination-pattern .T
session protocol sipv2
session target sip-server
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 2000 voip
description -[INCOMING FROM PSTN]-
translation-profile incoming UCCX
session protocol sipv2
session target sip-server
incoming called-number 067000
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 1000 voip
description -[OUTGOING TO CUCM]-
destination-pattern 9998
session protocol sipv2
session target ipv4:172.31.2.15
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 1003 voip
description -[OUTGOING TO MSK through CUCM]-
destination-pattern 849[59].......$
session protocol sipv2
session target ipv4:172.31.2.15
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 1004 voip
description -[OUTGOING TO SPB through CUCM]-
destination-pattern 8812.......$
session protocol sipv2
session target ipv4:172.31.2.15
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 1005 voip
description -[OUTGOING TO KRD through CUCM]-
destination-pattern 8861.......$
session protocol sipv2
session target ipv4:172.31.2.15
voice-class codec 1
dtmf-relay rtp-nte
no vad
dial-peer voice 6103 voip
description -[To CUCM_MSK]-
destination-pattern ....
session protocol sipv2
session target ipv4:172.31.2.15
voice-class codec 1
dtmf-relay rtp-nte
no vad

 

 

 

Hi,

On registrar configuration, add a refresh ratio timer.

Eg.

registrar ipv4:195.211.120.9:9060 expires 900 refresh-ratio 50

 

 

HTH

 

 

Regards

 

 

Carlo

 

 

 

Please rate all helpful posts "The more you help the more you learn"

Thanks, i'll try that.

Unfortunately, today sip-ua gone missing again :(

Maybe i can retrieve some logs about sip-ua somewhere?

What is the output of show sip-ua statistics? You many need to clear the statistics so you can show new errors.

I'm also thinking that debug ccsip errors or debug ccsip non-call might show what's happening with the sip-ua.

Maren

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