Vladimir Mikhelson vlad at mikhelson.com writes:
Hi,
Here is the reply from the developer as to what can be done
immediately to remove the offending logging.
He can just ignore these messages, they say that chan_ooh323 don't
known indication signal 33
exten = 123,1,Set(TIMEOUT(absolute)=3600)
exten = 123,n,MeetMe(blah,d)
if you are using freepbx and you want to set timeout for all conference rooms
go here -http://dn.forceit.ru/asterisk-conference-timeout
--
_
--
Thank you for reply, Larry!
On 06/03/2013 05:14 AM, Larry Moore wrote:
1) On SPA112 set FAX T38 Redundancy = 3
I have tried to change this value with no effect.
2) Add t38pt_usertpsource=yes in [mtt] section
This option take no positive effect for me, asterisk continues to use
ports from
On 06/03/2013 05:03 PM, Larry Moore wrote:
Have you checked the installed version of firmware against the latest
available from Cisco?
Oh! I didn't guess to check. The firmware was not fresh, but upgrading
doesn't help.
Looking at your SIP information when your ITSP initiated a T.38
session it
awsome scripts) fails.
I can work on Asterisk integration by myself, i'd be happy to know of
such engine(s) at all.
--
Andrey Utkin
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Mark Michelson wrote:
Caution: One shortcoming of queue member penalties is that they are not
taken into account if a queue member of a low penalty does not answer a
call. Say for instance that the queue application determines that there
are two members available to answer an incoming call.
Try to upgrade to the latest version. This was an issue (bug) but now
it's resolved.
Gary Chen:
I am using a2billing as calling card platform with asterisk 1.2.17.
After running for several days, if I issue 'sip show channels'
command, I got a lot of dead sip channels although 'show
I am using Local channel instead of callback agents and it works not as
good as I expected. If you add /n option then after the transfer queue
assumes that agent is still busy because asterisk doesn't hangup such
channels after tranfer. If you don't use /n then queue doesn't have info
about
. We see that asterisk replies to INVITEs after 4
seconds. That's wierd. Server is not heavily loaded - about 10
simultanious calls.
I've downgraded to 1.2.13 and problem has gone away. I guess there is
something wrong with asterisk.
Regards.
Andrey Solovjov.
Edoardo Serra:
Same to me
Hi
The same is for me. BLF doesn't work with 1.4. I've added notifyringing
= yes and this doesn't help.
Show hints doesn't show any status changes so asterisk doesn't send any
NOTIFY messages to grandstream. Message is only sent when extension
unregisters.
Andrew.
Ricardo Carvalho:
Dear
Chris Mason (Lists) пишет:
Tzafrir Cohen wrote:
Do you rotate Asterisk's logs with the logger or with logrotate?
I have never addressed this before and never seen this problem before.
The issue is causing thousand of log files to be written to the
/var/log/asterisk directory, so many that I
Hello all
I asked similar question some time ago but didn't get answer... Maybe
this should asked in asterisk-dev or bugs.digium.com?
For example, I have 3 sip phones defined in sip.conf - 101, 103, 109 and
this simple dialplan:
[local-ext]
exten = 101,1,Dial(SIP/101,,t)
exten =
trunk, but
we need to put Asterisk in a different geographical location from the PBX and
need to explore other options.
Thanks.
Andrey
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
places (or not visibile at all) with the larger
font size.
Try to use SJphone. It's free and easy to use.
http://sjlabs.com
--
Sincerely Yours,
Andrey Loginov
Insource LLC.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
and SEP000AF4BB7D59.cnf.xml exists.
I'll be very thankful for any your help.
--
WBR, Krasavin Andrey
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options
Hi,
Try this one:
Dual Tone Multifrequency Relay for SIP calls Using Named Telephone Events
http://www.cisco.com/en/US/products/sw/iosswrel/ps1839/products_feature_guide09186a0080087edb.html
Andrey.
This is a password-protected document (CCO account required.) Can
you refer to a non
http://www.cisco.com/en/US/customer/products/sw/iosswrel/ps1839/products_feature_guide09186a0080087edb.html
Andrey.
We're having an issue with connecting a Cisco ITS installation to * such
that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or
to any of the interfaces
): RFC3389 support
incomplete. Turn off on client if possible.
Where am I wrong?
--
Sincerely yours,
Andrey Katkov.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
=1
But, demo voicemail doesn't accept dtmf dialing.
I've changed string dtmfmode to inband and demo start work ...
--
Sincerely yours,
Andrey Katkov.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk
19 matches
Mail list logo
