asterisk users - Oct 2006 - random one way audio and noise between SIP phoneson same LAN

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


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Klaus Darilion
Sent: Wednesday, October 18, 2006 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] random one way audio and noise between SIP
phoneson same LAN

Do you use canreinvite (sip.conf)?

Change the setting (setting canreinvite=yes may cause nat problems) nad 
verify if the problem still appears.

Using htis setting you can find out if the Audio problem occurs only 
when media is relayed via Asterisk (->the problem is caused by Asterisk) 
or in all cases (the problem is not caused by Asterisk)

regards
klaus

Giorgio Incantalupo wrote:
> Hi,
> sometimes I have one way calls and noise between sip phones connected to 
> the same LAN so no nat/firewall is involved. I tried with different sip 
> phone models soft phones and the result is the same. I searched inside 
> every log file but found nothing. I made different PBX with different 
> hardware but the result is always the same.
> 
> Is there anybody experiencing this terrible problem?
> Considering to monitor a remote PBX via ssh, which  text-only 
> application  could  I use?
> 
> TIA
> 
> Giorgio Incantalupo
> 
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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
>
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Klaus
Darilion
> Sent: Wednesday, October 18, 2006 8:48 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] random one way audio and noise between SIP
> phoneson same LAN
>
> Do you use canreinvite (sip.conf)?
>
> Change the setting (setting canreinvite=yes may cause nat problems) nad 
> verify if the problem still appears.
>
> Using htis setting you can find out if the Audio problem occurs only 
> when media is relayed via Asterisk (->the problem is caused by Asterisk)
> or in all cases (the problem is not caused by Asterisk)
>
> regards
> klaus
>
> Giorgio Incantalupo wrote:
>   
>> Hi,
>> sometimes I have one way calls and noise between sip phones connected
to
>> the same LAN so no nat/firewall is involved. I tried with different sip
>> phone models soft phones and the result is the same. I searched inside 
>> every log file but found nothing. I made different PBX with different 
>> hardware but the result is always the same.
>>
>> Is there anybody experiencing this terrible problem?
>> Considering to monitor a remote PBX via ssh, which  text-only 
>> application  could  I use?
>>
>> TIA
>>
>> Giorgio Incantalupo
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>     
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

Scott Scecina wrote:
> In all cases, the called party cannot hear the calling party.  

do you have the RTP ports open?




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