voice quality degradation when transferring to another extension

Posted on 2013-08-30
Medium Priority
Last Modified: 2014-04-09
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?
Question by:kavlins
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LVL 15

Expert Comment

ID: 39453916
Are the phones all using the same audio codec?

Author Comment

ID: 39454002
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.
LVL 15

Accepted Solution

Perarduaadastra earned 550 total points
ID: 39454025
Try specifying the lowest-bandwidth codec supported by all the handsets, and see if this makes call quality better. The fact that this happens only when the call is transferred to another phone suggests that the problem is possibly a centralised one; perhaps your PBX configuration needs tweaking a bit.

There must be Asterix experts who can help you out with the details; my knowledge of these systems is limited and somewhat generic.
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Assisted Solution

nauliv earned 450 total points
ID: 39454335
kavlins: a key question is do you allow phone-to-phone voice traffic ?
it's in sip.conf; prior 1.6 "canreinvite=" , and later "directmedia="
whatever the current configuration is; try the opposite.
also is there routing going on between the asterisk pbx and the phones; or are they in the same network ?
ideally if you can post your sip.conf; it'll help us a lot :)

Author Comment

ID: 39454351
Will post shortly, sip.conf maynot have much since freepbx GUI is installed will include sip_additional.conf

Author Comment

ID: 39454383
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..

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