CUCM and CME integration Problems

[luqman_khalid]

hello everyone!

i am facing problem in integration of CME with CUCM. whenever i try to call from one sip phone(registered on CUCM) to  another which is registered on CME. phone rings but as i picked up the call is dropped.

i have formed SIP trunk on CUCM and all routing is fine. but SIP session(i think) is not establishing between two phones. two c2921 routers are integrated on gigports and one is CME enabled and other is CUCM integrated.following is configuration for CME.

voice service voip
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
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711ulaw
sip
registrar server expires max 600 min 60
!
voice class codec 1
codec preference 1 g711ulaw

dial-peer voice 1 voip
description Incoming from CUCM
session protocol sipv2
session target sip-server
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad

regards

[Aeby Vinod]

Is the region relationship between the IP Phone registered on the CUCM and SIP trunk allowed for G711.ulaw codec, if not can you configure that to be, and apply the voice class codec 1 you configured on the dial-peer and test ?

Please rate if you find this helpful.

Regards,
Aeby

[luqman_khalid]

yes i have tried out to assign g77ulaw codec on cucm and sip trunk but problem remains the same

[Aeby Vinod]

Ok, can you run the below two debugs after pasting in this config

dial-peer voice 1 voip
voice-class codec 1

Debugs to run:
debug ccsip messages
debug voice ccapi inout

Once debug is enabled make a test call, if unsuccessful please attach the output of show run along with the debugs gathered.

Also please include the calling and called number.

Please rate if you find this helpful.

Regards,
Aeby

[luqman_khalid]

i have attached router's configuration(CME enabled,remote site).

after i gave codec 77ulaw call is not dropping when calling from HQ (CUCM) but i cannot hear from SIP phone on CUCM and it can hear my voice from Phone on CME and also i can not make call from my CME enabled router(remote site) to CUCM(HQ). and ping is successful from CUCM to every SIP phone.

number on CUCM(1000004)

SIP Phone on CME (3020005)

both routers are connected on gi0/0 via optical fiber

(SIP Phone)1000004<---->CUCM(HQ)<--->c2921rtr<---->c2921(CME router)<---->3020005(SIP Phone)

[R0g22 · Accepted solution]

Your configuration is missing SIP bindings. The following seems like your voice SVI -

interface GigabitEthernet1/0.2
description RS 12 MOR COMMS rtr Voice
encapsulation dot1Q 2
ip address 172.17.33.1 255.255.255.0

Bind your CUCM facing dial-peers with the correct interface/IP.

[luqman_khalid]

what is purpose of SIP binding in this scenario ?

[R0g22]

Read the benefits section here -
https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-sip-bind.html