Peter Fern wrote:
I've had the same problem with all boxen running the same version. We
ditched IAX2 for SIP and it has been working fine since.
Well, upgrading my remote site to 1.2.5 appears to have fixed my issues.
-Barry
Doug Lytle wrote:
Barry Flanagan wrote:
Hi,
I have a
Hi,
I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to
connect to a 1.2.5 box for PSTN. There are 15 users on the remote
server, all connecting via SIP softphones.
For some reason, there is an increasing number of calls where the callee
does not get any audio although the
-users-
[EMAIL PROTECTED] On Behalf Of Barry Flanagan
Sent: Monday, March 20, 2006 2:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Problem with intermittent one-way audio
Hi,
I have a 1.2.4 asterisk box at a remote location, which is using IAX2
Barry Flanagan wrote:
Hi,
I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to
connect to a 1.2.5 box for PSTN. There are 15 users on the remote
server, all connecting via SIP softphones.
For some reason, there is an increasing number of calls where the callee
does not get
Doug Lytle wrote:
Barry Flanagan wrote:
Hi,
I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to
connect to a 1.2.5 box for PSTN. There are 15 users on the remote
server, all connecting via SIP softphones.
For some reason, there is an increasing number of calls where
I've had the same problem with all boxen running the same version. We
ditched IAX2 for SIP and it has been working fine since.
Doug Lytle wrote:
Barry Flanagan wrote:
Hi,
I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to
connect to a 1.2.5 box for PSTN. There are 15
