Technology: Voice - Communications Manager
Subtechnology: Communications Manager PSTN/PBX Inbound/outbound Call Failure across Cisco Gateway (H.323, MGCP, SIP)
Problem Code: Configuration Assistance
Software Version: Cisco Unified Communications Manager v.18.104.22.16800-2
Problem Details: Hi team.
I need support in the integration of a Communications Manager v.8.6 with a PBX Nortel Meridian 1.
The SIP Trunk was configured successfully between both devices. The calls on CUCM->PBX-Nortel direction are established perfectly.
The calls on PBX-Nortel->CUCM direction fail to establish and generates busy tone to caller.
The SIP Trunk configuration was performed according to the steps in the following document:
The last step is to apply a recommended SIP Normalization Script and set the one in the paper.
With this configuration, the call behavior on the PBX-Nortel->CUCM is the same.
This SIP Normalization Script understand that it is for the CUCM can modify the "phone-context" and the parameters that are before or after the @ (as defined).
In our case [according the captures (.cap) attached] calls coming from the Nortel PBX adds the extension you are calling, followed by the @ + domain.
Obviously the CUCM not know that domain and does not allow the call is established. We think these instructions should be given in the SIP Normalization Script. I need support to learn if the script used is correct.
On PBX-Nortel -> CUCM direction (Ext. 23264 to Ext. 22595), the call reaches the CUCM as follows:
sip:22595; firstname.lastname@example.org:5060 ... that "phone-context" and "domain" after the @ not know how the CUCM can interpret it and remove it. We believe it is through the SIP Normalization Script that CUCM can "clean up" the number of the receiver and the call can be established.
im having the same issue on the same kind of integration (Nortel - CUCM), how did you fixed it?
can you please share?
I have a similar issue. In my case I have a CUCM 9.1.2 and a Nortel CS1000E v.7.x
Did you find a solution for this ?
hi, Jerome, i have done a piece of script as follows
to = to:gsub(";phone%-context=[^;]*;([^@]*)@domain.com", ";%email@example.com")
to = to:gsub(";phone%-context=[^;]*;([^@]*)", ";%firstname.lastname@example.org")
to = to:gsub(";phone%-context=[^@domain.com]*@", "@")
to = to:gsub(";user=phone", "")
get to read it from the following link
http://www.lua.org/manual/5.2/manual.html#6.4, but at the end the nortel guy modified the parameters and removed the piece of information i needed.
I have the same issue for sip trunk between Call Manager 9.1 and Nortel but for Call manager return Unable to find a device handler for the request received after get SIP invite from Nortel.
I don;t know how to solving this issue.
Can you explain your script for me?
The issue you describe does not seem to be related with the script.
Have you made sure that the Incoming CSS contains the right partition ? I remember having faced the same issue some time ago and the CSS/Partition was the reason.
Other possibility is the Organization Top Level Domain in the enterprise parameters that is not set.
incoming CSS is correct, Organization Top Level Domain already configure but problem still occur. So if we used voice gateway between CUCM and Nortel is work fine.
CUCM -- [sip] -- Nortel = not work
CUCM -- [sip] -- CiscoVG -- [sip] -- Nortel = work
have you configured all the possible IP addresses for the SIP trunk? Perhaps the Nortel will send from multiple IP addresses. For example I have a virtual IP to send my requests to, but in total there are 4 addresses that may reply.
Can you check the CDR logs and see whether your calls fall into the right partition ?