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victdang
‎08-30-2017

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Member since ‎11-04-2010
21
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40
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RTP Traffic not seen with debug IP packet

Created by in Voice over IP
11-23-2011
11-23-2011
Have you tried the suggested PCM capture procedure from Cisco below:# config t# voice hpi capture bu...

No busy signal when calling from PSTN

Created by in Unified Communications Infrastructure
11-23-2011
11-23-2011
Are you using SIP or H323 between the 2921 to CUCM? Also beside not hearing busy tone, do you hear a...

Translation rule problem with SIP trunk

Created by in Other Collaboration Subjects
11-23-2011
11-23-2011
Do you have a specific device pool assigned to the SIP trunk?My general practice I used to do is set...

CCM SIP 503 SERVICE_UNAVAILABLE

Created by in IP Telephony and Phones
11-07-2011
11-07-2011
There in insufficient information to troubleshoot your issue. Is this your own gateway or a 3rd part...

5300 problem when user dial digits

Created by in IP Telephony and Phones
11-07-2011
11-07-2011
Is your AS5300 providing the IVR application itself, or are you forwarding via VoIP to a 3rd party I...

How to configure FXO on a CME router

Created by in IP Telephony and Phones
10-27-2011
10-27-2011
The easiest way is you can create several separate outbound POTS dial-peers, one pointing to the PRI...

Meeting Place : Web Interface

Created by in Other Collaboration Subjects
09-12-2011
09-12-2011
As mentioned, for MP8, you need to integrate with WebEx. To keep meeting internal, you need to imple...

Meeting Place Express 2.1.1 CLI Username/Password

Created by in Unified Communications Infrastructure
08-24-2011
08-24-2011
Hello Hasnain,Have you tried the recovery procedures below:http://docwiki.cisco.com/wiki/Cisco_Unifi...

MeetingPlace 8.x and WebEx Integration

Created by in Unified Communications Infrastructure
08-24-2011
08-24-2011
Hello Mark,Just to respond to some of your questions:1. I intend to set up a test cluster as a proof...

Re: CUE/CME MWI calling phone's DN

Created by in IP Telephony and Phones
03-21-2011
03-21-2011
Hello Michael,A few things you can check:1. Have you created an inbound voip dial-peer for the MWI n...

Re: My VIC cards not getting enough DSP resources

Created by in IP Telephony and Phones
03-19-2011
03-19-2011
What codec complexity are you using on your voice-card?You can try this:voice-card 0 codec complexit...

Re: Transform the dialed number with translation rule

Created by in IP Telephony and Phones
03-18-2011
03-18-2011
You can try this...!voice translation-rule 100 rule 1 /\(.*\)#/ /\1#/ rule 2 /\(.*\)/ /\1#/!voice tr...

Re: Transform the dialed number with translation rule

Created by in IP Telephony and Phones
03-18-2011
03-18-2011
I dont think the carrier will expect you to specifically put the # symbol, it is an optional setting...

Re: CUCM 7.1 and CME sip trunk issue

Created by in IP Telephony and Phones
03-18-2011
03-18-2011
I see this in your log:SIP/2.0 503 Service UnavailableDate: Fri, 18 Mar 2011 06:18:07 GMTWarning: 39...

Re: CUCM 7.1 and CME sip trunk issue

Created by in IP Telephony and Phones
03-17-2011
03-17-2011
The configuration looks fine... I simulate similar setup on my lab here (although I didnt put in as ...
Public Statistics
Date Registered ‎11-04-2010 06:03 PM
Date Last Visited ‎08-30-2017 08:43 AM
Total Messages Posted 21
Total Helpful Votes Received 40
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