[asterisk-users] Sporadic One Way Audio

[OCG Technical Support]

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
>
>

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