Heads Up :
The post you are writing will appear in a public forum. Please ensure all content is appropriate for public consumption. Review the employee guidelines for the community here.
I am having a SIP issue on UCM with call forwarding and caller ID. If I enable privacy header under voice service voip/sip, I no longer receive caller ID on incoming calls.However, if I remove this command caller ID will work but DN call forwarding f...
I did not setup single sign on for WebEx, this was done by someone who is no longer with our company.I see in the admin portal that my Cisco (SP) cert is expiring soon, but the cert usage shows "none". We use Azure IdP which is valid for another few ...
Running an 11.5 HA pair of UC. Primary server is at our datacenter. Secondary is at our HQ. When the primary box answers calls, the voice quality is very garbled. When I stop taking calls and force calls to be answered by the secondary box, the ca...
I have a couple dozen stand alone conference units (SX10/20, RoomKit etc). Currently, I have an HTML file that I import as the Favorites for the contact list. As we keep adding units, it is getting time consuming to update all the Favorites each ti...
Last night, we ported over our DID's from legacy ISDN/PRI to SIP trunks from PSTN. After doing so, our conference bridge from UCM to Lync will not connect. I suspect an incompatibility with the SIP trunks or profiles but cannot determine the cause. I...
Here is my SIP settings under voice service voip. Do you recommend removing asserted-ID PAI? sipsession transport tcprel1xx disableheader-passingerror-passthruregistrar server expires max 1200 min 60asserted-id paiprivacy headerearly-offer forcedmid...
Sorry for the late reply. Weather issues here so no power over the last week. No issues to the PSTN. I have a phone at the DC and no voice quality issues between DC phone and HQ phone, although I do not have a phone on the server VLAN so slight dif...
After much head banging and web sluething, I Was able to get this working finally by changing a setting in the SIP profile for the trunk. I set the "SIP Rel1xx Options" to "Send PRACK if 1xx Contains SDP"
I know this is an old thread but I had the same issue and was able to resolve by going to the web interface of the ATA, Line 2 and select enable. It was set to "no" for the line enable option.
