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missing speech issue when calling callpilot from sip phone off g450 2

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monkey101

MIS
Apr 24, 2009
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I have a 1000e, 1000m and callpilot with SM,SMGR, and some g450 all tied together. The problem I am having is when a RED SIP phone calls into callpilot to retrieve there messages the begining of the system greeting is missing. For example McFly voice mail system is what is played when called with a Blue phone. When you call with a Red phone you get Fly voicemail system. you are missing the Mc. Also when you dial an internal Red or Blue extension from a Red phone and it get transferred to voicemail you get the pers at extension blah blah instead of the person at extension blah blah. It is the same thing from every Red phone. All the Blue phones work fine. Any one have any ideas on what I can change to correct this?
 
sorry forgot to mention that both the cs1000 and CM are the latest releases.
 
WOW no offense but we need a delorian packet sniffer , what red kit do you have ?? this may well be public/private unknown numbering ....but unfortunately with the info you have provide the doc is going to be gunned down by terrorists in a mini van and we are all going to end up in the 18th century

APSS (SME)
ACSS (SME)
ACIS (UC)
 
the red side is cm\sm\smgr 6.3 with 96xx sip phones on G450's. I am a blue guy so i need more help to what exactly on the red side info wise you are looking for.
 
Maybe turn off shuffling in system-parameters features, last pages. That, or toggle early media in the SIP trunks to SM for the phones or to SM for CS1k.

Maybe you're not getting that beginning part when you're on DSPs before you shuffle off of them and go direct IP. Between your SM trunks from CM for the phones and CS1k, reconciling stuff there is maybe your problem. Or at least having an exact idea of the call flow. Please excuse the crudity of my explanation. :)
 
thanks I will look into that when back on site tomorrow.
 
it was the shuffling that was causing my issue. soon as i turned it off i was golden....thanks again
 
We had the same problem and turning off shuffling fixed it.. however this is not the best solution if you have more than 400 phones or so.
Does anyone have another idea?
 
how come it is not the best solution for alot of phones? what does this do?
 
Shuffling" is the ability to set up a call path between two IP endpoints by rerouting the voice channel away from the usual TDM bus connection and creating a direct IP-to-IP connection. Shuffling saves resources like TDM bus time slots and media channels and improve voice quality by eliminating unnecessary codec conversions. So by NOT enabling shuffling, you are continuing to use DSPs in the G450 for the duration of the call which will limit the number of simultaneous calls which can be made.
 
Glad to hear it. Now, that setting was a test - to turn off shuffling system wide. I would recommend doing it at a more granular level - like on the sig group form or the network region interconnectivity such that you can shuffle system wide except when CP is involved.

And, list measurements Dsp to see what your usage is and make an informed decision about how you're impacting the overall availability of those DSPs.

Also, check into your cs1k sip programming and see if you can set up sip early media. I think your problem was that you weren't hearing the call on thedsp before it shuffled. To say, you were always using the Dsp to start the call, and a second in, you went direct from the 9600 to probably an mgc card in cs1k. You can either go phone to mgc card all the way - probably with sip early media from the get go, or by not shuffling and going avaya phone to avaya Dsp to mgc the whole way.
 
would turning off shuffling cause one way speech from the 9611 sip phones but not one-x clients?
 
It certainly can. Status the stations and get their speech path details one way and the other and you'll get some more details over where the problem might be.
 
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