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?
LVL 6
kavlinsAsked:
Who is Participating?
I wear a lot of hats...

"The solutions and answers provided on Experts Exchange have been extremely helpful to me over the last few years. I wear a lot of hats - Developer, Database Administrator, Help Desk, etc., so I know a lot of things but not a lot about one thing. Experts Exchange gives me answers from people who do know a lot about one thing, in a easy to use platform." -Todd S.

PerarduaadastraCommented:
Are the phones all using the same audio codec?
0
kavlinsAuthor Commented:
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.
0
PerarduaadastraCommented:
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.
0

Experts Exchange Solution brought to you by

Your issues matter to us.

Facing a tech roadblock? Get the help and guidance you need from experienced professionals who care. Ask your question anytime, anywhere, with no hassle.

Start your 7-day free trial
Powerful Yet Easy-to-Use Network Monitoring

Identify excessive bandwidth utilization or unexpected application traffic with SolarWinds Bandwidth Analyzer Pack.

naulivCommented:
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 :)
0
kavlinsAuthor Commented:
Will post shortly, sip.conf maynot have much since freepbx GUI is installed will include sip_additional.conf
0
kavlinsAuthor Commented:
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
0
It's more than this solution.Get answers and train to solve all your tech problems - anytime, anywhere.Try it for free Edge Out The Competitionfor your dream job with proven skills and certifications.Get started today Stand Outas the employee with proven skills.Start learning today for free Move Your Career Forwardwith certification training in the latest technologies.Start your trial today
Voice Over IP

From novice to tech pro — start learning today.