Issues with this configuration. The document above works well for me for most calls with a couple exceptions.
Call Quality is excellent for Voice Calls until my system transfers a call to the unity express module. Then all I get is really garbled speech. (Anyone seen this before?)
The xml config file has been updated so the screen shots in the above instructions need to be modified a bit.
Ted - thanks for the feedback. Glad to hear all works well for the most part
1. CUE garbled voice - do not see that in my setup. Does CUE work well internally? If so not sure if there is a transcoding issue - would confirm the transcoder is registered - show sdspfarm unit status
My issue was transcoders. I discovered that if you have a preexisting SIP config in CCA, changing the profile does not put all the CLI in properly. There should be a way to remove a SIP config in CCA before applying a new one.
My new issue is this. When A call is redirected to voicemail from an IPphone Skype DID starting with 9905........... the number translation does not work properly. Unity Express says that there is no number for that extension.
Did you set the SIP Trunk provider to None before you applied the Skype for SIP profile? The steps to change a SIP Trunk SP on CCA would be to set the SIP trunk to None and apply config, then set SIP trunk to new SP and apply.
On the new issue - would need some more specifics on the version on the UC500 / CUE, the config and a SIP troubleshooting log from CCA for a failed call. You can private message this to me.
I just re set up Skype for SIP both in CCA 2.1 and with the Beta of 2.2 using the Beta software pack. I had the same result with both configurations and figured out the fix.
To get a Skype call to successfully redirect to voicemail I went to the web GUI of Unity Express and had to define an E.164 number for the user. The E.164 number that worked was the 9905.......... number.
Well you know how you can fix one issue and cause another? I now do not have DTMF relay working on calls inbound from SIP after a redirect to voicemail with the default configuration. Any ideas? (For all of you reading - SIP for Skype is still in beta - Finding bugs is what beta is about!)
Not sure how DTMF fails after your last change as they are not related as far as I can see. Can you point the DID straight to the AA on CUE and see if you can press any digits. If not would need to collect logs per section 5.3 along with "deb voip rtp session named"
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