06.05.2015 10:58, s m пишет:
Hello!
I'm not h323 expert, may be somebody else can understand from this log
what is happening, but I can't :-(
Could you, please, provide log with
tracelevel=6
in ooh323.conf ?
Thank you!
hello Dmitry
thank you for your reply. Ok, you are right. i want to
On 05/05/15 17:52, Olivier wrote:
2. From personal experience, would you rate an IMAP migration as an
easy or as a difficult task ?
By IMAP migration, I mean changing from one IMAP software to another,
on the same or on an other box.
There is software called 'imapsync' which will sync mail fro
Hi Guys
We have a strange issue whereby one phone has delayed rtp
So what happens is when the lady answers the phone for the 1st 1 second
they can not hear her and then everything is fine
I am running asterisk 1.8.28.0
Has anyone seen this before?
--
_
Hi,
I've got a system on which Asterisk 11 was first installed.
Then Asterisk 13 was tentatively installed.
Then I'm trying to re-install Asterisk 11.
Every install was made from Asterisk source.
Now Asterisk 11 seems correctly installed but basic "dialplan reload"
command fails with:
dialplan r
imapsync seems a very interesting tool.
Thanks for sharing this.
Now I'm still curious to learn if moving from local file storage to cloud
IMAP storage is still resilient to short network outages (without IMAP
replication).
Regards
2015-05-06 10:18 GMT+02:00 Gareth Blades :
> On 05/05/15 17:52,
With regard to your question about the imap store I highly suspect that
if it is unavailable when the message is saved then the message will be
lost. There are configuration options for the timeout and it might retry
a couple of times but Asterisk has no scheduling so I cannot see a way
for it
Hey;
It seems to me that for what you want to do, it would be easier just to
email the user the voicemail audio file as an attachment. I believe that when
you choose to store voicemails using IMAP, it applies to all of your users
which may not be what you want to do.
Regards;
John
Fro
Hello,
I am running Asterisk 11 on CentOS 6.4 with about 150 local SIP clients on a
LAN. The SIP clients are a mixture of Yealink phones (e.g SIP-T32G, SIP-T42G,
etc). I have configured the system as follows:
sip.conf:
[169]
secret=11
dtmfmode=rfc2833
directmedia=no
directrtpsetup=yes
canrein
hello
thanks Dmitry for your useful hints. i enable debug and solve my problem:).
it was codec compatibility problem. but it is so strange; if i set codec
g711alaw in cisco router and asterisk, i have the mentioned problem but if
i set codec to transparent in cisco router, every thing will be ok.
07.05.2015 08:17, s m пишет:
hello
thanks Dmitry for your useful hints. i enable debug and solve my
problem:). it was codec compatibility problem. but it is so strange;
if i set codec g711alaw in cisco router and asterisk, i have the
mentioned problem but if i set codec to transparent in cisc
10 matches
Mail list logo