Hi
After going from AST_DATA (RES_DATA) to realtime with mysql-driver my
console is jam'ed with
SIP Seeding peers from Astdb '000b8201' at
[EMAIL PROTECTED]:35273 for 120
I got arround 4000 sipclients registered at that server and all the
sip-client re-register every 120 sec. so the consol
Hi
Anyone have a hint how to get callers IP-Address from a php-agi script ?
/HH
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Hi Adam,
Sory to say it, bu it still interupt the mouse if you have microsoft
wireless mouse/keayboard.
The mouse jumps around on the screen. Any news on this ?
/HHA
Adam Hart wrote:
As always, I'm happy to announce a new version of Firefly.
Firefly 1.9.8 has more of what you want and less of wha
Hi
I got a problem with asterisk 1.0.2 - it drops the calls, both
sip<-->sip, and zap<-->sip.
The conntions can stay for seconds to several minuttes, and then the
connection just cut off.
I can't see anything in the logfiles. (or dont know what to look at.)
It drops several connections at a tim
Hi
I'm running safe_asterisk, but get core-files in /tmp - how do I debug
them ?
I know gdb asterisk core.12370
and bt full
But it didn't show anything usefull for me.
Can anyone help me ?
(Running asterisk 1.0.2 with ast_data
/Hans-Henrik
-
Last from bt full:
priority=200, ca
Hi
In voicemail.conf I have
attach=yes (tried with =1 and = thrue)
but I cant get asterisk to attach the voicemail.
Any clue ??? (using ast_data)
/HHA
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Seems pretty strange. Do you get any debug messages?
No, not in debug, or message-log :(
Anyway in the meantime, make sure you are using safe_asterisk to start
asterisk with (it will restart the server if anything goes wrong). You
Nice - I'll do that
also have to bear in mind that this will ki
Hi,
Now, whe I do asterisk -ccvvr
Unable to connect to remote asterisk
/HH
Olson, Dana wrote:
I'm reasonably sure that if you do the restart now that asterisk does restart in the background, but you will be kicked out of the CLI. You'll have to reconnect to it. I am assuming that you're run
Hi,
Sometimes my asterisk server stops. (after a day or two)
Last output from CLI is:
-- Registered SIP '000b82017eb7' at 213.237.12.125 port 11620 expires 120
-- Channel 0/26, span 1 got hangup
-- Hungup 'Zap/26-1'
voip1*CLI>
Disconnected from Asterisk se
Hi,
Is there a easy way to disable callwaiting ?
I had tried incominglimit in sip.conf, but it seems not to work.
/hhandresen
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To
Non Global SIP elements,
Non Global IAX elements,
Are there a way go get asterisk to wirte it's own hostname in the
sip/iax-friends table when a sip/iax-client connects ?
(When it update the ipaddress, client name eg.)
I have several *-servers using the same db, and the user can connect to
a ra
Hi,
I want to monitor how much load I got on my channels (mostly zap)
any tools sugestions ?
running 1.0.2
/HH
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Hi after patching with
http://bugs.digium.com/bug_view_page.php?bug_id=0002613
I got this error when I start Asterisk - any clue ?
[res_config_mysql.so]Nov 27 13:22:46 WARNING[15909]: loader.c:318
ast_load_resource: No load_module in module
/usr/lib/asterisk/modules/res_config_mysql.so
Nov 27
RealTime info would be greate
/HH
Matthew Boehm wrote:
Which version of Asterisk? It depends...if CVS, then you need RealTime info,
if Stable, then you need other info.
Matthew
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Hi
I mean change in sip.conf after asterisk has started and I want it to
have effect immediately.
/HH
Damian Minkov wrote:
What does it mean "RealTime configuration" is it just configuring from
database when * starts , or if i change something in database
it will immediately affect the * config
OT:
But where did you get the 1.0.5.18 firmware ?
/HH
PS - My Grandstream phones (BT100) with 1.0.5.18,
and Send-Flash-Event-as-DTMF=No,
now are doing Attended transfer just fine!
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11-10-2004 there was a subject:
Re: Where did USE_SIP_MYSQL_FRIENDS go?:
on asterisk.user list.
>All db specific code has been removed from the code in favor of the
>currently-in-development "RealTime" method of configuration from
>database.
>You are most likely not using the 1.0 stable branch.
>
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