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Hi all,I have downloaded v11 of the Cisco Web Communicator and the corresponding server files for the sample pages. When I open the pages, I allow the application to run in the background and then I get a blank command prompt running with a flashing ...
Hi all,
Just a quick question/thought; I was wondering if it would be at all possible to point the destination of a Jabber Guest URL to be that of Conductor rendezvous number?
If I set a route pattern on CUCM to say that any number begging 654X forwa...
Hi All,
I am having a few problems getting external calls from a SIP circuit reaching my CUCM server.
I have a range of DDIs (01539760560 through to 01539760574) and I will be setting up the extensions on CUCM to be between 5560 - 5574.
I have attac...
Hi all,I am having an issue connecting my SX10 to CUCM via SIP. I can connect other phones via SCCP but SIP seems to be having an issue. Registration = Unknown and IPv4 address = None however on the device I can see it has obtained an IP address succ...
Hello guys,I was wondering if someone would be able to help? I have encountered an issue where when I log in to the Jabber SDK web page I get the following error:"Unable to login: TFTP fetch error eTftpCouldNotConnect"I thought I had come across all ...
Sounds like a Windows issue whereby the output volume will automatically change if the operating system detects microphone activity and lowers the output from the speakers to cater for the microphone reducing feedback.Try altering this on your operat...
This is an excerpt of a debug I have ran:
Nov 13 13:59:45.693: //88/9E400A6C810D/SIP/Msg/ccsipDisplayMsg:Sent: SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 88.215.63.9:5060;branch=z9hG4bKfe3e3c6fbe7ebc01c6214db98252e5d6
From: <sip:07718572110@88.215.63.9>;t...
Hi Ayodeji,
I have attached the updated configuration as well as the output of the debugs you have asked for.
Just out of curiosity so that I can learn myself; what is it you typically look for in these debug outputs?
Thanks.
Hi,
I have updated the config; it now shows four call legs (two inbound, two outbound) as well as some sip-ua config too.
I have also attached debug voice ccapi inout and debug ccsip message outputs after applying the updated config.
Thanks