SIP phones in remote site with CME cannot call to Main office ip phones registered in CUCM

[filiberto.aguirre]

Hi,

 we have the following scenario:

- Remote site

CME 10.5
SIP phones / SCCP phones

- Main Office

CUCM 11.5
SIP phones / SCCP phones

- Remote office connect to Main office by VPN

- In the Main office there was a CME but was replace by the current CUCM 11.5 last weekend.

- With CME in  main office there was no problem, but after the installation of CUCM this issue was detected.

The calls from remote site with SCCP phones to Main Office work fine and from Main office to remote side also.

The calls from remote site with SIP phones to Main Office cannot be completed but calls from main office to remote site with SIP phones work well.

Currently this is the dial-peer set in remote office (CME).With this we can dial from remote site to main office just with SCCP phones.

dial-peer voice 198 voip
description *** to Main office ****
destination-pattern 15..
session target ipv4:X.X.X.X   (CUCM)
voice-class codec 1
dtmf-relay h245-alphanumeric

We also tried with SIP option but for this case did not work for SCCP and SIP phones
!
dial-peer voice 1181 voip
description *** to Main office ****
destination-pattern 10..
session protocol sipv2
session target ipv4:X.X.X.X   (cucm)
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip pass-thru content sdp
dtmf-relay rtp-nte
no vad

any suggestion, recommendation ?

[vhilario]

How did you integreate CME with CUCM?  What´s the exact call flow for the problem?

Phone-----CME----SIP-----CUCM---Phone?       Have you checked that the CSS of the SIP Trunk (or what ever protocol are you using) has the partition of the phones on the HQ? 

Have you gathered any debugs or cucm traces to see why the call cannot be completed?

What error are you getting? The call cannot be completed as dialed, or are you getting a fast busy

[filiberto.aguirre]

Hi Hilario,

I attach  the traces for ext 1548 in CUCM , and a debug in CME for remote site for ext 5122.

The behaivor is... when the user in remote site  (CME  sip phone extension 5122)  dial  to main office (CUCM  ext 1548)

the user (remote site)  listen  ring back tone  but when the user in main office off-hook  a busy tone is listenin remote site.

regards

[vhilario]

Hi,

It sounds to me like a códec issue, let me check the traces and Will get back to you as son as I can

[vhilario]

Hi Filiberto,

Do you have the callmanager trace files? I need them to check the call, I need timestamp of the call, calling and called number.