Hey guys,I'm having yet another strange problem. I've recently set canreinvite=yes, allowing the RTP streams to avoid our * server. Now, a few people are experience one way audio drops on internal calls. External calls are working fine (they re-invite directly to a Cisco router). Sometimes, if you
- Original Message -
From: Gary Richardson
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 13:54:04 -0300
Subject: [asterisk-users] canreinvite=yes
and RTP dropping in and out
Hey guys,
I'm having
My next attempt at this is going to be putting a hub in between the path to the switch. I'm hoping to be able to sniff the packets to see what's going on.Also, using the network status page on the hard phones, the transmit and receive counters for the direction of the channel slows way down as if
- Original Message -
From: Gary Richardson
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 14:34:31 -0300
Subject: Re: [asterisk-users]
canreinvite=yes and RTP dropping in and out
My next attempt
