I need to run this by everyone and see if someone has any idea's. I have a asterisk server setup and currently am receiving the inbound calling number where the name should be. My setup is....
One pri terminating into a Cisco 2431 router
Sip messages from the Cisco get sent to a asterisk server
linksys ata's a each remote end.
I can receive the calling name if the call originates from another extension on the asterisk server, I also can "make" the Cisco send out a generic name to the asterisk sip server and I see the name I statically assign in the Cisco appears on the terminating end (linksys ata)
I use the command in the Cisco under sip-ua
calling-info pstn-to-sip from name set name
timers buffer-invite 5000
I have also tried to add the commands...
-voice service voip,sip
Basically I need to take the field Remote-party id and place it in the sip message "From"
Here is some debug from the sip messages in the Cisco... this is an example of "no caller id Name"
ATTACHED IS A MY CONFIG AND SOME SIP DEBUG MESSAGES
SIP traces provide key information in troubleshooting SIP Trunks, SIP
endpoints and other SIP related issues. Even though these traces are in
clear text, these texts can be gibberish unless you understand fully
what they mean. This document attempts to br...
Please find the attached HTML document, download and open it on your PC.
This provides an easy to use form where you simply answer a few
questions and it will render the proper jabber-config.xml file for you
to copy/paste. There is built in logic to verif...
CUCM Database Replication is an area in which Cisco customers and
partners have asked for more in-depth training in being able to properly
assess a replication problem and potentially resolve an issue without
involving TAC. This document discusses the bas...