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,The calls are getting disconnected at Voice gateway. Q931 debug logs says cause code 80bfservice option not available. Please let me know wheher it is an issue with Service provider or I need to modify my VG configuration.ThanksKrishna Mohan
Hi,i have 2921 gateway with PVDM3-64 and 12 FXO ports. I am using the gateway as SRST and for PSTN calls. I have a confusion about the DSP calculation for e-phones (are they utilizing any DSP resorce and how much?) note that i have max-ephone 25 and ...
Hi, there.My name is carlos and I'm having some issues with customization. Is there any possibility to change the subject on the email template.The default is:subject: Message NotificationThe router is running a Unity express version 3.2.1
Hi ,I have configured B-ACD on CME 7.1, I attach the sh run file with debug file.When I dial 800 from any ip phone I expect to listen welcome prompt (already loaded on flash memory also app-b-acd-3.0.0.2.tcl and app-b-acd-aa-3.0.0.2.tcl files) but ...
Hi,i have a centralized call processing cluster of CUCM servers version 6.0.1 (publisher and subscriber) in main site and i have remote site a gateway with PSTN lines for the remote branch and it will function s SRST in case of WAN failure.all remote...
Hi, I am having a problem with caller id. I am receiving something like 55E6E6D in the caller id. Is there any translation rule I can use to strip all the letters or replace them with digits? I am using Cisco AS5400. Thanks.
I have the problem of not being able then updated the phone 7971, I configured the following:tftp-server flash: apps70.8-4-1-23.sbntftp-server flash: cnu70.8-4-1-23.sbntftp-server flash: cvm70sccp.8-4-1-23.sbntftp-server flash: dsp70.8-4-1-23.sbntftp...
I have a problem with voice quality, the problem began 3 days ago.if user of CME calls to PSTN , PSTN party listen CME's user voice choppyHi,I have already checked CME , the only thing that keep my attention was the info displayed by the following c...
Hello everyone--i've got a client with a SIP trunk, running through CUBE. After a painful process of getting the AA to work properly after the migration, they are having MoH issues now. What's happening is internal users hear ringback, MoH, etc. fi...
When I try to import a user into Unity Connection, I get an error that says"Friday, September 10, 2010 1:31:58 PM MDT ERROR importing user (ljavalera) with extension = 1310 : A User with the specified extension already exists in the Partition."yet wh...
HI,Imagine I have this scenario:CLients&IP-Phones(non Cisco)---Layer2Switch-----Router2821------WANLink----Provider(IP Centrex Solution, Call Control hosted in the cloud service provider).Let's say WAN Link goes down.Question:Can IP phones still make...
I have a CUCM 4.2 Pub/Sub setup and my client had an employee leave and they hired a replacement at the same extension.I made all the appropriate changes to the system but now if someone tries to dial by last name the new username is not found. Dial...
Guys:CUCM scheduled backup fails with exit code:2.. Here is the log file=====================================================Server : NJHPSUCXN01Feature : CONNECTION_MESSAGES_UNITYMBXDB1Component : MESSAGESTime Completed: 2010-07-06-...
All, Im racking my brain on this one and need some Presence help. Ive got everything working except two things. When I into the CUPS client, and go to server health, I have two things that show up as not working:1. Voicemail: It says "failed to c...
Hi all,I'm trying to sort out if I could send a number to for my phone to call over IP. The idea is to simple take a number from my PC and by sending it over IP it'll start calling that number on my phone.Any thoughts?Thanks,John
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