[asterisk-users] Sporadic One Way Audio

[Brent Davidson]

I don't think it's a snom specific issue as I have 5 branch offices all 
with identical configurations and only one of the 5 is experiencing this 
problem.  I have checked and re-checked the config files until they're 
practically burned into my brain and everything appears looks OK.  Any 
place there is a config option for the server address it matches the 
server's physical address.  The phones and all computers on the network 
are statically assigned (these are very small 1-person or 2 at the most 
offices) so I'm absolutely sure that there is no chance of an IP 
conflict.  The phones and the Asterisk server are plugged in to the same 
ethernet switch.

I have tried tried using both sip debug and Wireshark to try to catch 
anything relating to this, but the calls seem to happen very rarely and, 
as usually happens when I'm trying to troubleshoot some sporadic 
problem, the two people in that office can't ever reliably reproduce 
this when I happen to be in the office or remotely logged in to the server.

To make things even more difficult, the Snom phones have a switch built 
in to them and the PC's are in the pass-through port.  This means that 
any traffic generated by the phones is in the PC's blind spot.  Anyway, 
I've got about 1000 other projects going right now.  Once I get a couple 
of the more pressing ones knocked out I'll try to spend a couple of days 
at the problematic office and see if I can't come up with some way to 
resolve the problem.

Thanks for the suggestions,
Brent

OCG Technical Support wrote:
> Well, if this is snom specific I can't offer more insight.  It really sounds
> like misconfigured iptables and/or sip helper (conntrack/nat/etc).
>
> Are you sure your IP address is right in your sip.conf? If you don;t have
> NAT set to yes for these phones, they will trust the sip header for IP
> address and may misroute.
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brent Davidson
> Sent: October 24, 2008 7:36 PM
> To: Asterisk Users List
> Subject: Re: [asterisk-users] Sporadic One Way Audio
> Importance: High
>
> The asterisk server is connected to the PSTN via a Rhino R4FXO-EC card.
> The lost RTP would have be between the Asterisk server and the phones.
> There are only 2 phones in the building, 2 lines coming in to the
> asterisk server and the server is on the same ethernet switch as the
> phones.  The phones are SIP phones.  This is a simple PBX system that
> picks up calls from the analog lines and routes them to the appropriate
> phone, although it will eventually be linked to a larger system once all
> the minor bugs are resolved.
>
>
>
> OCG Technical Support wrote:
>   
>> How is your asterisk server connected to the PSTN?  SIP/IAX out...ISDN/T1
>> out? Etc...
>>
>> Are you looking for lost RTP between * and internal phones or * and
>>     
> external
>   
>> provider?
>>
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Brent
>>     
> Davidson
>   
>> Sent: October 24, 2008 5:55 PM
>> To: Asterisk Users List
>> Subject: [asterisk-users] Sporadic One Way Audio
>>
>> I'm having an unusual problem at one of my branch offices.  Every now
>> and then they will make a call and the person they call is unable to
>> hear them, but they are able to hear the person.  The Asterisk server
>> has only one ethernet interface and is on the same physical network as
>> the 2 snom 300 phones and is connected to the PSTN lines with a  Rhino
>> R4FXO-EC card.  Usually hanging up and calling back solves the problem,
>> but it is still aggravating to the customer that has been called.
>> Normally I'd suspect that something was only passing packets in one
>> direction, but there is no firewall between the asterisk server and the
>> phones and no iptables or anything like that running on the Asterisk
>> server and sifting through sip debug logs to try to find one call out of
>> maybe 50 has so far proven fruitless.
>>
>> Are there any common issues that might cause this?
>>
>> Thanks,
>> Brent Davidson
>>
>>
>>     
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081027/2b6018d4/attachment.htm