Hi,
>> deny=0.0.0.0/0.0.0.0
>> permit=XXX.XXX.X.X/29
>> permit=192.168.1.0/24
>
> Are you sure your provider *always* sends data from this /29?
I'm sure, yes. Its a MPLS net, only voice inside.
Regards,
Georg
--
_
-- Bandwidt
> deny=0.0.0.0/0.0.0.0
> permit=XXX.XXX.X.X/29
> permit=192.168.1.0/24
Are you sure your provider *always* sends data from this /29?
Maybe you have this in your iptables as well and sometimes audio is
received from outside this /29 and therefore blocked?
--
Andreas Sikkema
--
2009/10/16 Ishfaq Malik :
>
> Brent Davidson wrote:
>> We have several offices running Asterisk version 1.4.20.1, and OSLEC
>> with Rhino R4FXO-EC and one running a Digium TDM800P card for interface
>> to analog lines. All offices are running Snom 300 phones. Phones all
>> have static addresses a
Hi
For the sporadic one way audio, check that the codec list in the snom
phones is the same as set by the server. The codec list is in the RTP
tab of the identities.
Hope that helps
Ish
Brent Davidson wrote:
> We have several offices running Asterisk version 1.4.20.1, and OSLEC
> with Rhino
ight 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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: October 24, 2008 7:36 PM
To: Asterisk Users List
Subj
We have seen cases where an IP address conflict caused something like this.
You can take Wireshark traces on the PC (possibly run them in a loop so that
you have a pretty long context) and if you have one-way audio be quick to log
on to the web interface of the phone and also take a wireshark (P
dress and may misroute.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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
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 ph
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: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brent Davidson
Sent: October 24, 2