Welcome to the Cisco Networking Professionals Ask the Expert conversation. This is an opportunity get an update on Cisco Unified Communications Express with Cisco expert Tony Huynh. Tony is a technical marketing engineer for Cisco Callmanager Express (CME) at Cisco Systems, Inc. He is a (CCIE # 11056) Cisco Certified Internetwork Expert in routing, switching and voice with eight years of design and support experience in the information technology industry. Over the years, he has worked for various corporations including several Fortune 500 companies. Tony is an expert in documenting, designing, and implementing various communication systems. His areas of expertise include technologies such as routing & switching as well as IP telephony.
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We have a CUCME 7.1 working on Cisco 2811 router.A sip trunk is been configured with a service provider at london.
Is there an option to change the calling name based on the called number.This facility is required for us as the inbound calls are answered by a single support team having different DID numbers for different support.
As i had seen through the older posts in the forum and found a tcl script to match the calling number and name.This option i would not be able to use as the callers are from different parts of the world.
Kindly let me know if there is an option available
I am always facing the disconnection problem on FXO ports when i integrate the cucme with analog PBXs or with TELCO lines.
I tried all ways described on the internet (frequency, cadence, battery-reversal...) with no luck. Sometimes, the line disconnects after 9 seconds.
I asked many experts, and they told me that it is a headache and till now there is no permanent solution by cisco. Even when i open a case with TAC engineers to solve it but with no luck.
Is there any common problem that cisco engineers are working on to solve it???
Your help is really appreciated
What type of signaling are you using on the fxo ports? I know that if analog lines are not terminated properly (eg. 4 wire instead of 2 wire, polarity reversed, etc) then this essentially causes problems on the connections. Could you provide more info?
The SNR feature on CME is available for free without any additional software, hardware or license requirements.
Can CME interoperate with CUPC to provide presence and soft phone capabilities?
I have an issue making 2 or more calls within a short space of time between two phones, one native IP behind a CME running 220.127.116.11 and the other behind a 2811 router with a basic H323 voice gateway configuration.
The (edited) configs for each end are attached. The first call from one phone to the other (or using csim) goes through fine. However, once the call is ended, if a subsequent call is made within a minute or two afterwards to the same phone (or a phone via the same destination gateway), the call fails and I get the following debug output from h225/h245 (from CME).
*Jul 16 09:01:49.052: h323chan_chn_process_read_socket: fd=2 of type CONNECTED has data
*Jul 16 09:01:49.052: h323chan_recvdata: recv failure on fd=2: errno=254 errstr=Connection reset by peerh323chan_chn_close: Calls Exist on socketfd=2 Owner
*Jul 16 09:01:49.052: h323chan_close: TCP connection from fd=2 closed
If the calls are initiated from the Head Office end, the h225 output contained in the 'output from voice gateway' file is generated.
CME 7.1 with CUE 7.0(3).
Can we setup message waiting idicator on/off for 2nd/3rd button's voice mails as the primary line?