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 ...
Why do some of my calls get sent to Cisco Unity messaging system default mailbox instead of my personal mailbox?
Dears.I Have Avaya K175 configured as SIP on Cisco Call Manager.the phone keeps disconnecting every 2- 3 minutes.I can make calls from Avaya phone to Cisco phones. but Avaya is not receiving from Cisco " gives busy tone".I checked Cisco & Avaya logs ...
There may not be a "Best" strategy for adding a 9 when dialing out.I want to set up a fax machine so that any time the user sends a fax, they don't have to dial a "9"I thought I read that PLAR could do this in a Translation Pattern.I'm having trouble...
I’m trying to deploy a DECT IP network using Cisco Unified Communications Manager (CUCM).“CUCM” is essentially just a Cisco PBX and “DECT” is just a protocol for handsets to communicate with base stations.I think I found a limitation in CUCM.Basicall...
Dear Members,I have a Vodafone UK Landline; I was able to successfully configure it on Grand Stream HT812. When I use an analogue phone, everything works perfectly. Inbound & Outbound calling works, and also the caller ID is working perfectly. Now, W...
Hello, guys. I have a problem on my 4321 router. When I try to configure a sip ua, i've got a message: %Parse_RC-4_PRC-NON_COMPLIANCE:credentialsXXXXXXX username YYYYYYY realm ZZZZZ so, I can't see any outbound traffic to my SIP ISP and there is n...
We have SolarWinds monitoring solution in our environment. I was trying to configure SIP trunk session monitoring on SolarWinds VNQM, however I did not see any SIP trunks populated in SolarWinds. I opened a ticket with SolarWinds and they did postman...
My setup is as follows: CUCM -> CUBE -> ( Cisco Meraki MX400) -> ITSP. CUBE is behind NAT. Meraki does not support SIP NAT ALG. ITSP sees private IP in headers as the IP to send audio to despite 1:1 NAT mapping on firewall. How do I get the CUBE ...
Please is there a way I can register a 6851 3pcc to cucm We use cucm and have more than 200 6900 8800 and 9971 registered to it But then we also have a lot of 6851 3pcc which were mistakenly bought some years ago Is there a way we can register them t...
Hi, we configured for our customer some 8961 ip phones with 3cx. We created the required templates and the calls are working fine. The problem occur when someone wants to forward a call. The phone sends the following message: REFER sip:x.x.x.x SIP/2....
Hello !I've recently setup a few 8811 and 8851 in our client office with an Asterisk-based stack but could not find a few settings asked by end-users.The biggest problem come from simultaneous call not ringing. If someone has a second call while alre...
Are there any known interoperability issues or more-specific causes of very high Jitter numbers in the call stats, such as the attached? Should the attached example be just considered invalid, or indicative of severe network issues?In this particular...
I have a SIP phone registered to CME on 12.4 Ive tried everything to keep the phone from trying to register to the ITSP I have tried: voice register dn, no-reg. Still trying to register to ITSP.I tried: voice register global, max-dn 15 no-reg both re...
Hi All, I have a 2900 using c2900-universalk9-mz.SPA.157_3.M8.bin IOS. I am trying to configure TLS to and ITSP but some of the commands are not available. I am loosely following the guide - Cisco Unified Border Element Configuration Guide Through Ci...
Is it possible to use free pbx or astrix as a voicemail service instead of cisco unity or unity express
