Ask the Expert event- CUCM: Configuration & Troubleshooting

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This topic is a chance to discuss more about Cisco Unified Communications Manager (CUCM) best configuration and troubleshooting practices. This session will provide you with a better understanding of media resources, voice protocols, IP phones, integrations with other voice systems like Unity Connection, UCCX, Recording Servers and Inform cast among others. 
Learn more about voice gateway integration like H323, SIP, MGCP or SCCP call routing issues.

In addition, experts will cover questions related to security features on CUCM, such as TLS, certificates and encryption.

Ask questions from Monday 1st to Friday 12th of October, 2018

Featured experts

Oscar Martinez has been a Cisco TAC engineer for CUCM team for half-decade, currently he is a Team Lead of CUCM team at Mexico City Offices. He specialises on Gateways and CUCM and he has over 7 years of experience working with voice technologies. Before, he worked at Ericsson for 2 years managing IMS technology, which is the voice part of LTE technology. Oscar holds a Bachelor’s Degree in Electronic engineers in Instituto Tecnologico de Culiacan, in Mexico. He loves traveling and playing soccer is his greatest hobby.

 Jay Robinson has been part of the Voice technology team at Cisco’s TAC for the last six years, he recently moved to Mexico to become a Team Lead for CUCM technology at this location. He has experience across different areas of CUCM, particularly on its configuration and troubleshooting. Before Cisco, he worked at Skyes in Costa Rica as Support Engineer for five years. Jay holds a degree in Business Management from the Concordia College, NY, USA. He holds a CCNA certification and is currently pursuing a CCIE in collaboration. On his spare time, he enjoys traveling and playing basketball. 

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[License]

CUCME and SIP Trunk Dialing Issues

Hello All,

I am using Cisco Call Manager Express and CUBE feature on an ISR4331 Router. I have Cisco 7942 & and 8865 IP Phones registered with CUCME. 

I have configured the CUBE as listed below:

voice service voip
ip address trusted list
mode border-element 
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
signaling forward rawmsg
fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 0 fallback pass-through g711alaw
cause-code legacy
h323
emptycapability
no h225 h245-address facility
h225 id-passthru
h245 passthru tcsnonstd-passthru
sip
midcall-signaling passthru media-change
early-offer forced

And Dial-peer is configured as below for mobile calls:

dial-peer voice 16 voip
description *** outbound dial-peer for Mobile calls ***
destination-pattern 05........
session protocol sipv2
session target ipv4:ITSP_SIP_SERVER_IP
voice-class codec 1 
voice-class sip early-offer forced
voice-class sip bind control source-interface Gi0/0/1.2500 
voice-class sip bind media source-interface Gi0/0/1.2500 
dtmf-relay rtp-nte
no vad

SIP Phone is configured as below:

voice register dn 53
number 4885

voice register pool 53
busy-trigger-per-button 2
id mac 5C50.1544.2014
session-transport tcp
type 7942
number 1 dn 53
dtmf-relay rtp-nte
voice-class codec 10
username User53 password 1234
no vad

But when I dial any mobile number say 0512345678 is getting "No matching outgoing dial-peer" erro. And I found SIP invite send only 0 as below:

Received: 
INVITE sip:0@192.168.100.1;user=phone SIP/2.0

Your urgent support is greatly appreciated. 

Thanks

[License]

Hello All,

I am using Cisco Call Manager Express and CUBE feature on an ISR4331 Router. I have Cisco 7942 & and 8865 IP Phones registered with CUCME. 

I have configured the CUBE as listed below:

voice service voip
ip address trusted list
mode border-element 
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
signaling forward rawmsg
fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 0 fallback pass-through g711alaw
cause-code legacy
h323
emptycapability
no h225 h245-address facility
h225 id-passthru
h245 passthru tcsnonstd-passthru
sip
midcall-signaling passthru media-change
early-offer forced

And Dial-peer is configured as below for mobile calls:

dial-peer voice 16 voip
description *** outbound dial-peer for Mobile calls ***
destination-pattern 05........
session protocol sipv2
session target ipv4:ITSP_SIP_SERVER_IP
voice-class codec 1 
voice-class sip early-offer forced
voice-class sip bind control source-interface Gi0/0/1.2500 
voice-class sip bind media source-interface Gi0/0/1.2500 
dtmf-relay rtp-nte
no vad

SIP Phone is configured as below:

voice register dn 53
number 4885

voice register pool 53
busy-trigger-per-button 2
id mac 5C50.1544.2014
session-transport tcp
type 7942
number 1 dn 53
dtmf-relay rtp-nte
voice-class codec 10
username User53 password 1234
no vad

But when I dial any mobile number say 0512345678 is getting "No matching outgoing dial-peer" erro. And I found SIP invite send only 0 as below:

Received: 
INVITE sip:0@192.168.100.1;user=phone SIP/2.0

Your urgent support is greatly appreciated. 

Thanks

[Hilda Arteaga]

hi @License

thanks for extending your question. You can find the answer at the discussion page of this event https://community.cisco.com/t5/events-for-customers-and/ask-the-expert-cucm-configuration-amp-troubleshooting/td-p/3715341/highlight/false/page/4