IP Telephony and Phones

I noted what seems to be excessive multicast traffic coming from a CME 2821 and was a-wonderin what would be considered normal.I am seeing about 55 packets per second (3300 per minute) of the following:Source: CME RouterDestination: 239.0.0.1Protocol...

Hello All,I have a UCM version 8.02 utilizing an internal SIP Trunk to an IVR (server).  The SIP Trunk functionality works well to the IVR, but if I initiate a transfer from the IVR to an outside number the call fails.  The call from the IVR utilizin...

Don't ask why, but I have some 800#'s to point to a local extension. Depending on time of day these numbers are answered locally.  But after hours, get forwarded off to another phone number that is NOT part of my CUCM7 cluster.My setup is using CUCM ...

Hello All,I am trying to convert and upload a ringtone into CUCM v8.0(2) with no luck at present - I have a few issues:I first converted a .wav to a .raw, uploaded it to the TFTP server, modified the Ringlist.xml and restarted TFTP clusterwide. Once ...

I have 1 third party IVR connected to an H323 gateway via a Digital T1 E and M winkstart port.  I can make calls to the IVR but when a person tries to zero out, it tries to transfer call but it is not working.The vendors says that I need to provide d...

I cant find any notes on what this package actually does on the router.Thanks!

Is there a way to restrict access to a VM box if someone else is already in it?This is a mailbox that several nurses check, depending on who is available. Since they are located in different areas, they aren’t aware when somebody else is accessing th...

We migrated from some older 7925 servers awhile ago but had not removed them from the Servers list  (System-->Servers).  When I removed them it seems this broke the attendant console program??? - I get the error "Initialization of Call Control Failed...

Hey everyone, I do not have a CUCM in front of me right now, but does anyone is there is an interdigit timeout for MGCP controlled trunks? (QSIG, PRI, CAS, etc).Im having a interdigit timeout issue I believe or I need switch my MTP sources from CUCM ...

Hi everybody.I have IP trunk between CUCM 7.0 and asterisk. Sometimes i got calls from asterisk like this:INVITE sip:7211388@xx.xx.xx.xx SIP/2.0Via: SIP/2.0/UDP yy.yy.yy.yy:5060;branch=z9hG4bK30e69283;rportMax-Forwards: 70From: "Kubly" <sip:Kubly@yy....

I'd like to know if Linksys phones configured in SIP and registred with CallManager 7.1 are able to support the SRST featureThanks