I think I have posted this before...but cannot locate it. We have MM5.2 and had CM5.2 and now are on CM6.0 - both phone system releases exhibit the same behavior - transfer a call into someone's voicemail and it cuts off the beginning of the greeting (and if someone has a rather short greeting, it just about elimiantes all of it and the caller only hears the beep). Is there a way to fix this other than having users re-record their greetings and leaving a pause at the beginning (not sure that works either). First thing they say is "our old voicemail system did not do that"...and they are right (old was Audix but they were also on CM3.1).
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Transfer callers to MM 5.2 - Not hearing beginning of greeting
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i assume that you are transferring by just blind transfer and not staying on the line? What type of ports are you using from the PBX to the VM ? T-1, E-1 set emulation , or Sip Gateway ?
Ken Means
"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)
Ken Means
"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)
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Correct and using SIP.
I'm being told (because I'm not using that phone/system yet) that when someone presses transfer/enters the transfer code, the complete button appears after the code is entered and by this time, the voicemail greeting is already playing.
I'm being told (because I'm not using that phone/system yet) that when someone presses transfer/enters the transfer code, the complete button appears after the code is entered and by this time, the voicemail greeting is already playing.
So they are using the transfer to VM code and not just transferring the call is what it sounds like and transfer to VM is a QSIG feature not a SIP feature. Is that what they are doing ?
And as far as the SIP goes is it native or with a Gateway ?
Ken Means
"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)
And as far as the SIP goes is it native or with a Gateway ?
Ken Means
"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)
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Yes - they use the code so it will associate with the called party's number/voicemail box. Otherwise, they would have to transfer/enter the voicemail bypass number/enter the extension of the called party - right?
SIP is through Session Mngr - does that answer your question?
SIP is through Session Mngr - does that answer your question?
yes it does. Bottom line is that the code (the SIP is the real culprit) is sending the call to fast for everything else to catchup so by the time the caller is connected the greeting is playing. I don't remember the exact setup of this but i will check to see if a pause can be inserted.
Ken Means
"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)
Ken Means
"I find that the harder I work, the more luck I seem to have."
- Thomas Jefferson (1743-1826)
In the VMSC, check your "Call Setup Delay for Media Re-direct (ms)" is set to (in the PBX section), the default is 500 I believe. Not sure if this will help but try to set it to a longer value like 1000 or longer if it will let you. The value says it doesn't require service restarts but I don't always believe that.
If changing that option doesn't help your situation, then you might have to call Avaya to see if they will give you a registry key that would delay MM from answering calls by a certain number of ms or seconds. This was done back in the day (for different integration reasons) but I'm not sure if the registry value is supported today by Avaya.
If changing that option doesn't help your situation, then you might have to call Avaya to see if they will give you a registry key that would delay MM from answering calls by a certain number of ms or seconds. This was done back in the day (for different integration reasons) but I'm not sure if the registry value is supported today by Avaya.
BTW, I know you said you were using SIP, but you didn't specify if you were using a SIP gateway, SES, Session Manager or direct to CM. The transfer to voicemail FAC/feature is not supported by Avaya through a SIP gateway - as noted in the configuration note 88011.
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