asterisk users - Jun 2006 - Bad call quality using a certain channel.

Hi,

I am fairly new at working with Asterisk.

I am having a call quality issue that I really need to get ironed out before
we go to rollout the system in a week.

Any help would be greatly appreciated!!! Even if it is just pointing me in
the right direction.

 

My current setup:

I have Asterisk Setup on a Dell Server. It has 2 T100P cards. One will be
for out T1 PRI from the Phone Company (We don't have this installed yet)

The other T100P connects to a VINA T1 IAD (Channel Bank)

I also have a Cisco 7960 SIP Phone attached and registered.

The Server is connected to a broadband connection.

 

 

My issue:

When I call the IAX Demo from the SIP Phone, the call is perfect. Asterisk
Voice is 100%, and the Voice from the Digium Test server is almost 100% (an
occasional stutter)..but very usable.

 

When I call the IAX Demo from a Phone connected to the VINA Channel Bank,
the Asterisk Voice is 100%, but once it connects to the test server it is
extremely choppy. You can kind of understand what is being said, but it is
very very poor quality and quite unusable.

 

When I between the Channel Bank and the SIP phone, the quality is 100% no
problems at all.

 

 

So..why does the VINA Channel Bank connection not seem to like the IAX side
of things, When I know that the IAX side is functioning great when used from
a SIP Phone?

 

I don't know what details would be pertinent to this, but here is what the
Asterisk Console Displays:

    -- Executing Playback("SIP/200-ad26", "demo-abouttotry")
in new stack

    -- Playing 'demo-abouttotry' (language 'en')

    -- Executing Dial("SIP/200-ad26",
"IAX2/guest@misery.digium.com/s@default") in new stack

    -- Called guest@misery.digium.com/s@default

    -- Call accepted by 216.207.245.8 (format gsm)

    -- Format for call is gsm

    -- IAX2/216.207.245.8:4569-1 is ringing

    -- IAX2/216.207.245.8:4569-1 answered SIP/200-ad26

    -- Hungup 'IAX2/216.207.245.8:4569-1'

  == Spawn extension (from-sip, 861, 2) exited non-zero on
'SIP/200-ad26'

asterisk1*CLI>

asterisk1*CLI>

    -- Starting simple switch on 'Zap/25-1'

    -- Executing Playback("Zap/25-1", "demo-abouttotry") in
new stack

    -- Playing 'demo-abouttotry' (language 'en')

    -- Executing Dial("Zap/25-1",
"IAX2/guest@misery.digium.com/s@default")
in new stack

    -- Called guest@misery.digium.com/s@default

    -- Call accepted by 216.207.245.8 (format gsm)

    -- Format for call is gsm

    -- IAX2/216.207.245.8:4569-2 is ringing

    -- IAX2/216.207.245.8:4569-2 answered Zap/25-1
<---------- THIS IS WHERE THE AUDIO BECOMES ALL CHOPPED UP.

    -- Hungup 'IAX2/216.207.245.8:4569-2'

  == Spawn extension (chan_bank, 861, 2) exited non-zero on 'Zap/25-1'

    -- Hungup 'Zap/25-1'

asterisk1*CLI>

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