[asterisk-users] random one way audio and noise betweenSIP phoneson same LAN

[Scott Scecina]

Giorgio,

I'll answer in reverse order:

I've not had reports of "noise" from my users.  However, when I went down to
get the s/w version from the phone that seems to be acting up the most, the
user reported that earlier they were actually on a call that was ok then
spontaneously dropped the audio. Per my instructions (based on another
similar report I read on Digium's site), my user hit a digit on the phone
which brought back the caller's audio.  I've also had them attempt to put the
call on hold, and then resume, but that did not bring the audio back.

As far as the S/W versions:

One of the phones that acts up (and they all should match):

Polycom 501
BootRom: 3.1.3.0131
BootBlock: 2.5.0
SIP: 1.6.6.0036

My phone, on which I've never experienced the problem:

Polycom 601
BootRom: 3.1.3.0131
BootBlock: 2.6.0
SIP: 1.6.6.0036

- Scott


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Wednesday, October 18, 2006 11:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] random one way audio and noise betweenSIP
phoneson same LAN

Hi Scott,
seems that we have the same problem...I have canreinvite=no and polycom 
phones.
I do not have cisco switch and qualify=yes but I think that is not the 
problem.

I've got 2 questions:
1) my polycom firmware is:
sip.ver: 1.6.5.0043
bootrom.ver: 2_6_2

what are yours?
2) have you got one way calls only or noise on sip calls conversations too?

TIA


Giorgio Incantalupo

P.S.: for configuration/monitoring apps  I'm still on it...I hope to 
find useful tools asap. In case, I'll let you know.


Scott Scecina wrote:
> I'm having the same "random" problem.
>
> I have "canreinvite=no" on all extensions.  I have "qualify => yes" on all
> non-NAT extensions. I do have several NAT extensions, but I've not had
> reports of problems from those.  95% of my extensions (all polycom 501/601)
> are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches.
>
> In all cases, the called party cannot hear the calling party.  The calling
> party has the "still ringing" icon on their phone, but can hear the called
> party talking.  I've got call monitoring turned on, and asterisk is
recording
> both sides of the conversation.   
>
> The problem occurs on SIP->SIP and Zap->SIP calls. 
>
> I've tried enabling sip debug on a particular extension that seemed to be
> experiencing the problem more than others.  However the problem did not
occur
> when the debugging was on.
>
> Sip debug generates so much noise I've been hesitant to turn it on
> system-wide.  Is there a way I can turn on sip debug and have all that
> logging go to a specific file (and not in the asterisk console)?
>
> Also, are there any other configuration/logging tricks I can try?
>
> Thank you,
>
> Scott Scecina
>
>