Hi,
I have configure a SIP trunk between two asterisk 1.4.24.1
After a while, sometimes a day or two, sometimes only a few hours, the SIP
connection between the two servers is lost.
'sip show peer status' shows the peer is unreachable.
'sip reload' resolves the problem, but I'm wondering if
tried setting qualify in the sip.conf?
http://www.voip-info.org/wiki/view/Asterisk+sip+qualify
Ish
dan julius wrote:
Hi,
I have configure a SIP trunk between two asterisk 1.4.24.1
After a while, sometimes a day or two, sometimes only a few hours, the
SIP connection between the two
Hi,
I'm using Asterisk 1.4.24.1
Is it possible (and recommended) to have realtime peers that are not cleared
from memory when 'sip reload' is issued?
According to https://issues.asterisk.org/view.php?id=14196 I thought having
rtcachefriends=yes would be enough, but this does't seem to work.
?id=14196
Thanks,
Dan
On Tue, Jul 28, 2009 at 6:33 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
From my experience sip reload always clears the realtime cache, what
exactly are you doing? Wouldn't doing a 'sip prune realtime peer/user'
for single peers/users be of use to you?
Ish
dan julius
Hi,
checkout
http://svnbook.red-bean.com/en/1.4/svn.tour.importing.html#svn.tour.importing.layout
this explains about versioning
Dan
On Tue, Sep 2, 2008 at 3:56 PM, bilal ghayyad [EMAIL PROTECTED] wrote:
Hi List;
I see and hear about the Trunk version, and sometimes when I ask about
How can you configure Asterisk to forward the calls you don't want to answer
back on the 2nd PRI line?
Does this traffic increase the load on the asterisk server, or is it
completely dealt with by the 2 port card?
Thanks,
Dan
On Thu, Aug 28, 2008 at 3:46 AM, Paul Hales [EMAIL PROTECTED] wrote:
Hi,
I'm using asterisk 1.14.19
I'm making a video call between two SIP end-points, using h263p and iLBC. I
notice the video is jumpy and I believe the cause is due to RTP timestamps.
The sending device is working at 8fps and correctly increases the timestamp
by 11250 every frame. It appears
