You can check whether logrotate sees/understands the asterisk's file in
/etc/logrotate.d volume by looking through
/var/log/logstatus
You should see names of the files you plan to rotate
Actually you should do
logrotate -s /var/log/logstatus /etc/logrotate.conf to see these files
hope this
Hello;
Facebook and Whatsapp sort-of support XMPP, so we can use Jabber to communicate
with them. But, how much jabber channel in asterisk is stable and updated?
Regards
Bilal
--
_
-- Bandwidth and Colocation Provided by
Hello;
There is no free channel to be used to have integration between asterisk and
skype? What is the software that I can use to send and receive chat messages on
skype network?
Regards
Bilal
--
_
-- Bandwidth and Colocation
On Thursday 23 May 2013, bilal ghayyad wrote:
Hello;
There is no free channel to be used to have integration between asterisk
and skype?
No; because by design, Skype is proprietary and closed, and takes various
extreme measures to block out all attempts to interoperate with it, or even to
On 23 May 2013, at 10:49, bilal ghayyad wrote:
Facebook and Whatsapp sort-of support XMPP, so we can use Jabber to
communicate with them. But, how much jabber channel in asterisk is stable and
updated?
You can find out the support status from menuselect
On 23.05.2013, at 12:57, bilal ghayyad bilmar...@yahoo.com wrote:
There is no free channel to be used to have integration between asterisk and
skype? What is the software that I can use to send and receive chat messages
on skype network?
For voice calls, you could try Skype Connect, which
I am not aware of any alive project on this field.
My client is interested in it. He told me that Google Translate web
service used to provide pronunciation of hebrew texts, but now speech
button is disabled for Hebrew, and also direct requesting for speech
generation (by Lefteris Zafiris's
Am 23.05.2013 15:14, schrieb Marie Fischer:
For voice calls, you could try Skype Connect, which is SIP - but needs a
business account, so not free. http://www.skype.com/en/features/skype-connect/
For voice, you can use SipToSis. Works flawlessly with Asterisk and the
best part, it's free.
Hi,
Actually i would like to get the input from the user and he should not try
more than 3 times, he can try more than 3 times, if yes it will get routed
to the next priority and if not it goes to the loopback again from the
beginning.
And following is the one I created, I just want to know
For voice, you can use SipToSis. Works flawlessly with Asterisk and the
best part, it's free. :)
www.mhspot.com/sts/
(site is down right now)
And that's related to the problem with it: it hasn't been maintained for
quite a while.
--
Hello Everyone,
I have bumped into the thralling penguin page on linux vs solaris for
asterisk. Does the benchmark still hold with the newer versions of
kernels? Curious to know of your thoughts. Also, they mentioned
running it on Sun Fire x2100, but no benchmarks were given for that.
Can
Hi,
Maybe you have not allowed T.38 as acceptable codec ;-)
You can try with allow=all in your sip.conf.
Am 22.05.2013 16:39, schrieb Andrew Colin:
Hi guys,
Any idea why I am getting this error when someone tries to send me a T38
Fax?
--
As long as you're dialing a screened registered voter list and don't call
.gov or .edu, you're fine.
On Wed, May 22, 2013 at 5:54 AM, Don Kelly d...@donkelly.biz wrote:
Calls on behalf of political candidates are generally legal--even to people
on the do not call lists. It doesn't seem to be
Am 08.05.2013 01:12, schrieb James Cloos:
SN == Sebastian Niehaus nieh...@web.de writes:
SN Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the
SN same maschine: Hylafax fax server. I want hylafax to use t38modem (a
SN virtual T.38 modem) for sending faxes. t38modem schould
One of the tricks used in Canada was to call the other party's
supporters and pretend that you are from their favorite party and piss
them off.
You can also give them misleading information such as a phony change in
voting location.
It appears to work since the guys who did this won the
Am 23.05.2013 16:04, schrieb Richard Kenner:
For voice, you can use SipToSis. Works flawlessly with Asterisk and the
best part, it's free. :)
www.mhspot.com/sts/
(site is down right now)
And that's related to the problem with it: it hasn't been maintained for
quite a while.
True, but it's
I just want to make some increment... to 3 and yes go to the desired option
not to one more option.
On Thu, May 23, 2013 at 7:19 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi,
Actually i would like to get the input from the user and he should not try
more than 3 times, he
488 not acceptable is due to codec error. Make sure you have right codec in
place between the end points.
On Fri, May 24, 2013 at 12:18 AM, Maximilian Grobecker
m.grobec...@portunity.de wrote:
Hi,
Maybe you have not allowed T.38 as acceptable codec ;-)
You can try with allow=all in your
18 matches
Mail list logo
