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voice quality degradation when transferring to another extension

We are experiencing voice quality degradation when transferring call to another extension within office. PBX systems in all offices are Asterisk or different flavors of it (Elastix,Trixbox,AsteriskNOW). IP phones are Polycom ranging IP 335, 450, 560..And we do have separate VLAN for data and voice. When in house receptionist answers the call regardless of the call was transferred thru PSTN or SIP line , quality is acceptable but when transferred to another phone , volume cuts to half, choppy... what do you think , the cause would be?
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Perarduaadastra
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Are the phones all using the same audio codec?
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kavlins
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ASKER

I haven't place codecs in extension settings (except on my phone Polycom ip 450) , G722,ulaw,alaw.. Other extensions are left blank.. Does this matter? I thought it would take the default codec G711 if nothing is placed.
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Perarduaadastra
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nauliv
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kavlins
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Will post shortly, sip.conf maynot have much since freepbx GUI is installed will include sip_additional.conf
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kavlins
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Please see attached sip.additional. fyi- i took out hundreds of other extensions from it , extension settings are same so..  Asterisk ipbx and phones are on same VLAN (vlan 200). Though the Cisco switchports are configured for both data and voice vlan. vlan 11/ 200. PCs are plugged into Polycom phone's PC ports..
sip-additional.txt