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Hi,I have an issue with my sip phones configured on a 2911 router running CME Version 8.6. outgoing calls automatically gets droped after 79 sec see below configuration for voice service voip and voice register global as well as the sip DN'svoice ser...
Hi,,I am having a problem making outbound calls to my SIP trunk. Always having the Address incomplete message in my debugs. Below are my SIP comfiguration. (some commands i think are already redundant)**voice service voip**!voice service voip qsig ...
Hi,I would like to know if its possible to configure trunk hunting from a given range of nubers from an ITSP?I am using a CME. I have been given a number range e.g. 1231234500-1231234599What i want to accomplish is i will make the number 1231234500 ...
Hi,I this issue that when callers are trying to call an extension and dropped the line, the extension will keep on ringing forever until the the phone goes off-hook. If i configure the connection plar in my FXO the extension dedicated to the FXO beh...
Question:My voicemail setup is not working when i dial an extention it is not forwarding to the mailbox nuber instead i get a message that the extension is busy. Did some debugging and saw a "being refused message"Configuration:UC router!ephone-dn ...
hi suresh,I am curious by the source ip address media 192.168.1.1 i have forced this in the SIP peer interface i am not sure why it is still showingPlease see below debug ccsip calls:Feb 3 14:43:44: //224742/23D6DCD4BEB8/SIP/Call/sipSPICallInfo:Disc...
Hi,I cannot provide as of now the debugs as i took out the phones from my client. But i can recall the issue came up after configuring a SIP trunk on the CME which affected the SIP phones. SCCP phones are working jut fine. I suspect the translatio...
i cannot iput this command in the router.. even i only call the local internal extension call gets cut after 79 secs. this problem is only hapenning on the SIP phone sccp phones got no issues. this issue started when i configured a SIP trunk on the...
Hi Suresh,https://supportforums.cisco.com/thread/2241718 thi forum helped. So basically i changed the translation-profile for the !voice translation-rule 3 rule 1 /^.../ /114818300/! i just removed the 11 as the ITSP is only accepting 7 digits calli...