Hi Scott,
so it seems that are polycom phones not working well...
have you tried with other IP phones or only with polycom?
Giorgio Incantalupo
Scott Scecina wrote:
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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
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