I have a SIP PSTN connection to my CME box. I'm able to use SIP DID into CME and forward over to Unity 4.2 for Voicemail or Auto Attendant greeting. I am unable to enter an extension and be transfer back to a CME extension. Internally on a 7961sccp Unity call transfer does work. How can I fix this? It seems on Callmanager a MTP is required but CME doesn't support this.
Cisco Unity Express supports IP telephones using Skinny Client Control Protocol (SCCP) or analog telephones behind an SCCP gateway (such as the Cisco VG 248 or the Cisco ATA). Media Gateway Control Protocol (MGCP) IP telephones, analog FXS telephones on the Cisco Unified CME router, and soft telephones are not supported.
Only the owner of a personal mailbox can delete messages in the mailbox. Members of a GDM can delete messages in the mailbox. The administrator cannot delete messages or display the length of time for which messages are stored in the system. When the mailbox owner logs in to the voice mailbox, the application notifies the owner of any expired messages.
If the mandatory message expiry feature is enabled, the owner must delete the expired messages. If the mandatory message expiry feature is disabled, the owner can delete or save each message.
If a message is saved from the expired messages menu, the expiry timer is restarted for that message.
Mailboxes can have different storage sizes. Consider the purpose of the mailbox when assigning a smaller or larger size than the default. The aggregate of all mailboxes cannot exceed the maximum storage allowed on your system. See "Software Licenses and Factory-Set Limits" for the mailbox storage capacity for your system, and use the show voicemail usage command to display the amount of storage
Are you replying to my post or did you click on the wrong post? My post has nothing to do with mailbox sizes or leaving messages. I asking about Auto Attendant in Unity 4.x and transferring out to extension. Internally this function works. Externally it does not. My external calls come in on a sip connection and then get forwarded to voice mail. When I enter an extension to transfer out of the auto attendant it states please wait while I transfer your call and then drops. Again this feature works internally but not when coming in on the sip connection. Any ideas would greatly be appreciated.
Did you check that unity has the ability to dial back to the operator extension? There are rules in Unity that limit dialing xxxx or 9xxx, etc. If the operator is not in the Unity subscriber system, and you enter xxxx, unity may not allow the transfer to happen because of the COS or dialing rules.
Yes. I am able to dial extension from Unity internally. I have the dialing rules and operator setup. It is only from the external SIP connection that it cannot transfer the calls from Unity. Looking at the port monitor I see one port in use by the current call and when it tries to transfer a second port shows in use for a split second and then the call drops. I've tried the full consult option as well.
I have tried both. I also installed the lastest TSP as well to see if that would fix the issue. It must be something to with SIP but I have no idea what to do from here?
Just for s and giggles, try this:
Create a new CallHandler
attach an extension to it (something you can call from the outside in.
In CallManager, CTI route point this to Unity so when it calls in, CCM routes it to Unity first.
In the CallHander, create a Caller Input to be transferred to your desk or some number OTHER than the operator.
"Thanks for call company xyz, press 1 for Nelson" Link this to your Subscriber Nelson, or a system transfer to input the number.
If it works, then it must be something with the extension at the operator.
If you have full blown Unity, you can see the call being transfered in Call View in Unity Tools or in the Port monitor.
Check this doc.... it would be similar to what you are trying to do with the CTI