asterisk users - Nov 2008 - upgrade from 1.2 to 1.4 and now half channel audio

Hi all,

I upgraded from asterisk 1.2.23 and zaptel 1.2.19
to asterisk 1.4.18 and zaptel 1.4.12.1
I use polycom 501 phones internally.

Everything seems fine. I can pick up the phone and call out,
calls coming in work just fine.

The issue I see is when the system first calls me,
then calls someone else. This works if its polycom to polycom. I hear 
audio full channel.
If I do  polycom to external line like a cell I only get HALF channel audio.
At this time they can hear me but I cannot hear them.

What might this be???

Jerry

You could trying changing this in sip.cfg
<AES voice.aes.hs.enable="0" 
To
<AES voice.aes.hs.enable="1" 

It's at line 324 in mine.  Results not guaranteed.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jerry Geis
Sent: Friday, November 21, 2008 10:28 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] upgrade from 1.2 to 1.4 and now half channel audio

Hi all,

I upgraded from asterisk 1.2.23 and zaptel 1.2.19
to asterisk 1.4.18 and zaptel 1.4.12.1
I use polycom 501 phones internally.

Everything seems fine. I can pick up the phone and call out,
calls coming in work just fine.

The issue I see is when the system first calls me,
then calls someone else. This works if its polycom to polycom. I hear 
audio full channel.
If I do  polycom to external line like a cell I only get HALF channel audio.
At this time they can hear me but I cannot hear them.

What might this be???

Jerry

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>
> You could trying changing this in sip.cfg
> <AES voice.aes.hs.enable="0" 
> To
> <AES voice.aes.hs.enable="1" 
>
>   
Just tried that - rebooted my polycom and still half audio.
Thanks,

Jerry

You could try un-commenting "duplex=2" in rpt.conf and changing it to
duplex=3.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jerry Geis
Sent: Friday, November 21, 2008 11:05 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] upgrade from 1.2 to 1.4 and now half
channelaudio
>
> You could trying changing this in sip.cfg
> <AES voice.aes.hs.enable="0" 
> To
> <AES voice.aes.hs.enable="1" 
>
>   
Just tried that - rebooted my polycom and still half audio.
Thanks,

Jerry

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I am using an AGI to setup the call to the first person,
then jumping into the dialplan with some Variables set.

Is the AGI messing up my channel???

My dialplan at that point looks like:

exten => 
call_cont,1,Dial(${CONT_CALLAT},${CONT_DIAL_TIMEOUT},${CONT_ONHOLD}tT)

CONT_CALLAT is Zap/1/506XXXX where X is my number
CONT_DIAL_TIMEOUT is 60
CONT_ONHOLD is tT

Seems like this should still be working also.
How do I tell where/how my audio is getting blocked. Internal polycom to 
polycom works fine with this AGI,
the old 1.2 worked fine with this AGI, its just polycom to external 
world with the AGI is giving me a half channel.

Jerry