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

[Giorgio Incantalupo]

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: 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
>>
>>
>>     
>
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