Tilghman Lesher wrote:
On Tuesday 27 December 2005 14:15, James Sizemore wrote:
I think I found what is munging up the peer lookup:
This call from another Asterisk box starts:
-- SIP read from 192.168.69.254:5060:
The peer lookup that fail reads:
-- SIP read from 192.168.7.250:52141
Doubling the value to 500 did not seem to effect the length of the
tone played at allhm. Back to the drawing board for me.
Anyone know what this value is supposed to effect?
James Sizemore wrote:
I did a bit of searching around and found this class in chan_sip.c:
I am going to test
I have a gateway using a Digium card to convert a PRI
call to a sip call then I transport the sip call to a Cisco
IAD where it is converted back to a PRI. This all works
well except DTMF is sent with a duration of .25sec.
PRI specs says this should be .25sec to .5sec so this
is with in spec,
(req, Content-Length, clen);
add_line(req, tmp);
return 0;
}
==
James Sizemore wrote:
I have a gateway using a Digium card to convert a PRI
call to a sip call then I transport the sip call to a Cisco
IAD where it is converted back to a PRI. This all works
well
If I had to guess you also log cdr's to a database and your database
server is slow for some reason, Asterisk will not hang up a call till
the database query finished, the telco will only wait so long for an
acknowledgment from a hang up and disconnects it's end and tried to use
the same
A clean checkout of CVS asterisk and zaptel are now failing to build:
gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-12/18/03-04:09:32\
I'm assuming you did something similar to this:
if (!join_queue(queuename, qe)) {
/* Start music on hold */
if (strcmp(qe.moh,RING))
{
ast_indicate(chan, AST_CONTROL_RINGING);
}
else
ast_moh_start(chan,
I would love a copy.
RAD Development wrote:
We have developed a RADIUS acc module for Asterisk. If someone is
interested, just drop a mail.
cheers
_
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Have you made a bug report on this? What is the number?
John Todd wrote:
Did you find any solutions to this? I've recently discovered the same
problem with my server here. I had 499 sip show channels entries,
and then Asterisk refused to start up any new sessions due to Too
many files
