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Greetings,
In my lab I am playing with CUCM, IM&P and Unity.
In user profile configuration "Home Cluster" and "Enable User for Unified IM and Presence" > are enabled.
I already configured in IM&P Server the CUCM SIP trunk, presence gateway, TFTP Serv...
Hi, I have a question regarding using of Route Group and Route List to manage fail over.When we have two SIP trunks (Trunk1 and Trunk2) and want to implement fail over solution for them and keep a trunk (for example Trunk1) as main trunk, we could co...
Hi everyone,
I can export phones and edit the Directory Number Forward All for the lines and import it to CUCM. It works fine and lines get the Call Forward All. I mea I did this as pilot for one CTI Remote Device which has shared line, without any p...
Hi Team,
we are facing one-way audio issue on hold and resume or conferencing scenarios.
When CUBE on one call leg gets an OK with only one code (Ex. g711alaw), it offers all the codecs configured on an outbound leg instead of offering only the filte...
Hi All,when I check the status of SIP connections I could see that I have a high number of "No. of inactive conn. ageouts". Base of documentations this value should be zero."Indicates the number of times that the connections were ended or timed out b...
Hi, try to build it in this way: { "service": "tomcat", "certificates": ["' + <tomcat_signed_certificate_as_pem> + '", "' + <Certificate_Chain_Trust> + '"], "description": "Uploaded by API"}tomcat signed certificate should be in this format:-----...
I've just check the bugs, it seems we are hitting following bug.
https://bst.cisco.com/bugsearch/bug/CSCvz36330
I couldn't find a link for remove/delete my own post, so answer it.
An old topic but still helpful.If you see ""no sip message to display", there are following possibilities:SIP traces have not been enabled in CUCM.Regarding the number of calls that you had, the traces had been overwritten.Someone deleted the traces,...
Hi Matthieu , after some years I faced to the same issue with one of our customers. I could solve the issue and want to share with others: Workaround:To solve the issue, add a dummy device such as CTI Port or CSF Device with a reletaed Device Pool th...
Hi Frank, you may try followings: Soultion 1:you can extend the match string in your sip profile and prove the call behavior:rule 4 request INVITE peer-header sip HISTORY-INFO copy "sip:(.*)@testdomain.grp;user=phone?" u07rule 6 request INVITE sip-he...
