Hello, We are currently using a Be6000 system and when we schedule a call forwarding on our phones (model 7945) to external numbers, it works perfectly when the original call comes from an internal position, but when it is an external call then this ...
Hi, I am new to cisco and this forum. Currently we have a 2821 router connected to 16 ATA186 devices. This is being deplyed on a client site and we have to add it to their existing rack system. They are looking to have both data and voice on site. Th...
I have a UC560 with PRI setup at the primary site and like to setup another UC at a remote site. What's the easiest way to change the default IP address of CUCM and CUE for the remote UC?The 2 sites are connected via MPLS. Can the remote site have a ...
Hello everyone,I got a new 2960s L3 PoE switch and need to enable it for voip on our network. The phone vendor is coming in to enable the phoen system for voip and I need to know how to configure the switch for the new IP phones. I need to know how t...
Hi all, hope everyone is well !!!I am using 3750 switches at the wiring closet and I sometimes lost power to the wiring closet and the switch got rebooted. But somehow not all the phones come up after the switch reboot and back online. Some phones go...
Hi,I have a customer with a cme 8 that needs to send the special code #00 that connects to another system in a sip trunk.I did a dial-peer with this destinattion-pattern like this:dial-peer voice 30 voip description ** #00 call ** destination-pattern...
Hello All. I am trying to figure out how SRST mode works in conjunction with my CUCM BE 5000 server. I am currently using MGCP from call manager to my gateway, which is also my SRST router. To my understanding when the call manager box goes down the ...
Hello,I did an upgrade yesterday to CUCM 8.6su1 and accidently used the unrestricted version. It is installed on VM. Is there any way to revert these changes, I have been told that it is permanent. Thanks,Ayaz
I have two dial-peers1. dial-peer voice 513 potstrunkgroup LOCAL-PSTNtranslation-profile outgoing drop_onepreference 1 destination 9513.......progress_ind alert enable 8forward-digits 72. dial-peer voie 1999 potstrunkgroup LD-PSTNtranslation-profile ...
Hi team,I have a environment working properly. I can choose 2 options after to hear the new messages: - Press 2 to delete the message- Press 3 to do a call sender How delete the message from "new messages" after to do a call the sender. thanks.
Hi voice team,I have a environment with Cisco Unity connection 8.5 working properly. At the menu "after message" i have 2 options: - Press 1 to save the new message;- Press 2 to delete...- PRESS 3 TO CALL SENDERMy problem is when I choose the option ...
All,The scenario is centralized CM 7.x at HQ with remote sites A and B. An MPLS network connects all sites. Call #1 from Site A to Site B completes but there is no audio. Hang up and make call #2 and the call completes with Audio working. Any ide...
Hello,could you help me, when i call from SIP phone 9951 to AVAYA over H323 there is no voice, but call is established.Thanks in advancevoice register globalmode cmesource-address 192.168.4.192 port 5060max-dn 5max-pool 5load 9951 sip9951.9-2-3-27aut...
Hello FolksI was wondering if anyone can answer the following;I need to use lots of FXS-E vics on 2911 gen2 routers.Now I can populate the 4 EHWIC slots giving me 8 ports. Next I thought I could use an NM-HD-2V to give me another two slots (I was und...
Hi,I am configuring the CME ver 8.6 for FXO ports.My customer has 4 FXO Ports and 1 T1/E1 card.The requirement is..once the PRI goes down then all the calls coming to recpetion should land on the FXO and it should ring on reception phone.My current r...
Hello guys! Im in the phase of integrating a CUCM 8.5 with a H323 Voice Gateway (CISCO2921-V/K9). The Cisco 2921 will be connected through interface Gi0/1 to the CUCM 8.5 and through interface Gi0/0 to the PSTN Service Provider. There will be ...
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