I am New Member, i am working on Voice Project with Call Manager Express(very new for me) , I have lot of queries can someone guide me through it
1)I have 2 Cisco 2951 routers can they have HA(publisher,subscriber) like Call Manager ?
A)Cisco 2951 rtr with CME:-8.6 and CUE:-8.6
2)I have Call Center set up which i need to Implement wherein all users will get call via Toll Free Number but the users wont have DID numbers as extensioni have 2 PRI and both should be active and by pressing 9(TATA PRI) and 0(REL PRI) how to configure this set up
3)How the users will make outbound call and take board line number
4)I need to configure common mailbox if the agents are busy customer can leave Voice mail for them ?(I JUST HAVE 6 PORTS IN WHICH IVR AND MAIL BOXES SHOULD WORK)
Can somebody help me with configuration ? or any notes
Thanks in Advance for help,
Recently , I had to configure a SIP trunk to operate with Microsoft Exchange as voicemail system but as the integrator of the exchange doesn’t know a lot about his system , he was unable to configure the voicemail boxes with the correct number so it gave me some mistake as Exhange was unable to resolve the called number . So after some searches , I found that we can use the SIP Normalization as in the CUBE but this time directly in the CUCM .
The SIP normalization for SIP trunk on CUCM is based on lua ( an old application language) but even if you have no complex structure , you can easily modify the SIP stack in the inbound and outbound direction.
So you need to read the cisco documentation to master but after an half day you can already play with it to have something.
Lua is really something that you need to try when you need to update a little your sip fields/option and which is really powerful . I will come on later on this post to put a short example.
I ran the issue that I was unable to import any of my tar files created on OSX stations . So after some digging , I found this article of William Bell who gave me the right solution for this problem :
In any case , don’t hesitate to read this good blog 🙂
I know that it is a while that I have not posted anything but now the time has come to put something back on the website .
I will be nearly two years that I have been certified and it is time to go back for study , so a little revision of some concepts and new SNRD is well welcome !!!
For the rest , I will try to populate the blog with some interesting UC stuff may be not in regards with the CCIE certification but well in regards with the Unified Communication stuffs that I want to share with you and that you can may be use in your CCIE Labs .
So keep an eye on the blog because I will come back in the coming days for an interesting subject but I ‘ll let you discover 🙂
It can seem dumb , but I’ve just played/discovered the ATA 187 ( which is working in SIP) . while I was trying to enter in the IVR menu , I was prompted for a password that we have not in the past.So after one or two google line , I finally discovered that the default password is cisco ( yeah I know , they are very hard 🙂 ) , so in numeric it is giving you 24726.
numb advise of the day 🙂 lol
Hi, I am very new to this voice domain.. I have a requirement. One of my customer is having Call Manager 8.X in his network. He wants his GM to receive all the internal calls directly, where as the calls from PSTN should reach the Secretary first and the secretary should forward the calls to GM. The last four digit of the DID is used as the extension number and both GM and secretary are using single line each (not a shared line). GM’s Phone model is 9951 and secretary’s 7965. Please help on this requirement.