Vladimir Mikhelson 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 (AST_CONTROL_PV
>
> 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
--
_
-
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
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 ud
x27;s awsome scripts) fails.
I can work on Asterisk integration by myself, i'd be happy to know of
such engine(s) at all.
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Andrey Utkin
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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 ca
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 chann
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 the
where this can be seen.
Mera SS is 10.150.16.4. 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.
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 all,
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 => 107,1,Dial(SI
trunk, but
we need to put Asterisk in a different geographical location from the PBX and
need to explore other options.
Thanks.
Andrey
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of the softphones I've tried end up with data
> elements appearing in weird 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
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Sincerely Yours,
Andrey Loginov
Insource LLC.
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SEP000AF4BB7D59.cnf.xml exists.
I'll be very thankful for any your help.
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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
> yo
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 an
> Immediately strides to ATA, rips off cover... woohoo, EEPROM is
> socketed well maybe I'll just copy the contents of a working
> ATA into the programmer, and reflash the locked one, taking care
> to change the MAC address and serial number..
As far as i know most of Cisco equipment built
on can
help me.
But I don't know how.
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Pavel Zheltouhov, Comlink ISP, Voronezh, Russia
phone/fax +7(0732) 727172, http://www.comlink.ru
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Sincerely yours,
Andrey Katkov.
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d
username=2299
secret=lordwhorfin
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=2299
nat=1
But, demo voicemail doesn't accept dtmf dialing.
I've changed string "dtmfmode" to "inband" and demo start work ...
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Andrey Katkov.
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27;ve used guide from http://www.loligo.com/asterisk/Cisco/ATA-186-
guide.v20030628.txt
I did all step by step. And one time it worked. When I reload Asterisk it
don't work again (and without message about RFC). How can I trace what happen?
--
le rtp.c, Line 221 (process_rfc3389): RFC3389 support
incomplete. Turn off on client if possible.
Where am I wrong?
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Andrey Katkov.
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his is the problem of dialgate, but
when i use it w/o asterisk it works w/o this problems.
Thanks in advance,
Andrey
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