IP Telephony and Phones

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New Voice Gateway

Hello,   I have been tasked with setting up a Cisco 4431 to replace our outdated 3825.  Instead of setting it up MGCP as the 3825 is, management wants it set up SIP.    This is not my strongest area of expertise and I'm having a hard time wrapping my...

Tony C by Beginner
  • 448 Views
  • 4 replies
  • 0 Helpful votes

Resolved! I want to rebuild my Subscriber

HelloI got this problem for my cucm version System version: 7.0.2.20000-5Product Part Number: MCS7825I3-K9-CMC1. I want to rebuild my Subscriber , my publisher is working fine , only the subscriber has an issue and I m thinking in rebuild it .I have ...

mohd355355 by Beginner
  • 5674 Views
  • 8 replies
  • 0 Helpful votes

Resolved! CUCM NTP will not synchronise

In a lab, I have a CSR that is configured as NTP master. The CSR is showing as synchronised: R1#show ntp statusClock is synchronized, stratum 3, reference is 185.53.93.157 nominal freq is 250.0000 Hz, actual freq is 249.8750 Hz, precision is 2**10ntp...

pedrop by Beginner
  • 354 Views
  • 1 replies
  • 0 Helpful votes

Resolved! debug ccsip messages not showing output

Hi all,   Im facing something strange, when im doing " debug ccsip messages" and after that term mon I don't seeing any output coming by in the cube. This has been worked before, i was checking the configurution ive noticed the command logging buffer...

monasir by Beginner
  • 4033 Views
  • 9 replies
  • 0 Helpful votes

Resolved! Call Forwarding

Hello, I would like to get some clarity on the TRAI regulations in India. We know that, we cannot forward a call originating on PSTN in India to a user outside India using VoIP. My question is: can we forward a call originating in countries like US o...

mailgunj by Beginner
  • 670 Views
  • 6 replies
  • 0 Helpful votes

call routing

 Hi,I have the following conditions.Have sip trunk which is connected to remote non cisco pbxAnd route group with two mgcp gateway on premise .  I need to achieve the following . If the user calls calls must  go through  sip trunk . If sip trunk fail...

elite2010 by Participant
  • 304 Views
  • 3 replies
  • 0 Helpful votes