I'm running 10.7.1 (yes, I know it's old, but this may be a problem in
later versions too) and had a conference being recorded via:
Set(CONFBRIDGE(bridge,record_conference)=yes)
The bridge started out at 8KHz despite one HD device. But when the
second came in (G.722), it switched to
Hi
Is there any way to change the preferred audio playback format in asterisk
(I'm using 1.8.25.0)
i.e. first check for gsm, if doesn't exits then check for slin?
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e:
On 23/01/14 13:38, Ishfaq Malik wrote:
Hi
Is there any way to change the preferred audio playback format in
asterisk (I'm using 1.8.25.0)
i.e. first check for gsm, if doesn't exits then check for slin?
It should pick whichever source format requires the least cpu to
transcode into the
Hope basically depends on the codec Asterisk will playback the file
automatically
On 23 Jan 2014 19:25, Gareth Blades mailinglist+aster...@dns99.co.uk
wrote:
On 23/01/14 13:38, Ishfaq Malik wrote:
Hi
Is there any way to change the preferred audio playback format in
asterisk (I'm using
On Thu, Jan 23, 2014 at 3:25 AM, Michelle Dupuis mdup...@ocg.ca wrote:
Hi
I'm creating an AMI client and I only want to get newchannel events (as
well as responses to any actions I initiate). What would I set the
eventmask to to only get the newchannel events?
Are you talking about the
Hello,
How to move 2 of 3 users in the MeetMe conference to the newly created
MeetMe conference? Dialplan example is welcome.
Best,
Igor
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New to
Thanks A. J.
*José Pablo Méndez *
On Wed, Jan 22, 2014 at 3:22 AM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
On Wednesday 22 January 2014, José Pablo Méndez Soto wrote:
Hello,
Is there anyway to encrypt or scramble a bit the secret used to register
with a provider? Im talking
On Thu, Jan 23, 2014 at 3:06 PM, Michelle Dupuis mdup...@ocg.ca wrote:
That's an interesting link - I didn't know you could set a per user
eventfilter in the conf file
However, I'm hoping to do this in the AMI connection for more
flexibility. Upon login, you can specify the event mask to
hi,
which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor
can i record to Opus?
--
---
Marek Cervenka
===
--
That's an interesting link - I didn't know you could set a per user eventfilter
in the conf file
However, I'm hoping to do this in the AMI connection for more flexibility.
Upon login, you can specify the event mask to restrict the type of events sent
over the AMI connection. Looking through
On 23/01/14 15:21, Marek Cervenka wrote:
hi,
which file extensios are supported in mixmonitor application?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor
can i record to Opus?
core show file formats will give you a list of formats your system
supports
Thanks Richard and Andres.
I had come to the same conclusion, however the provider was fairly snarky
in saying is was my equipment.
We were able to replace the Cisco 2800 with a Cisco 2900 series and the
problem appears to have been resolved.
Thanks again, I always appreciate another set of
Thanks - I've been through that doc before and couldn't find the info needed,
which is why I went to the source code eventually.
All events are grouped, and each group is given a name/flag like 'system',
'call', etc. The docs just don't say which events are in which group/flag.
Perhaps
When you use a product which version number is 11 or even 12, you might go
with the assumption all big bugs are fixed and then you find there is a
huge, important, expensive bug still running in the code we are relaying
upon...
The problem is simple. If you transfer a call, that dialing will be
On Thu, Jan 23, 2014 at 12:44 PM, Leandro Dardini ldard...@gmail.com wrote:
When you use a product which version number is 11 or even 12, you might go
with the assumption all big bugs are fixed and then you find there is a
huge, important, expensive bug still running in the code we are relaying
On Thu, Jan 23, 2014 at 9:31 AM, Michelle Dupuis mdup...@ocg.ca wrote:
Thanks - I've been through that doc before and couldn't find the info
needed, which is why I went to the source code eventually.
All events are grouped, and each group is given a name/flag like 'system',
'call', etc. The
On Thu, Jan 23, 2014 at 8:46 PM, Matthew Jordan mjor...@digium.com wrote:
On Thu, Jan 23, 2014 at 9:31 AM, Michelle Dupuis mdup...@ocg.ca wrote:
Thanks - I've been through that doc before and couldn't find the info
needed, which is why I went to the source code eventually.
All events are
2014/1/23 Matthew Jordan mjor...@digium.com
On Thu, Jan 23, 2014 at 12:44 PM, Leandro Dardini ldard...@gmail.com
wrote:
When you use a product which version number is 11 or even 12, you might
go
with the assumption all big bugs are fixed and then you find there is a
huge, important,
can someone confirm that mp3 is unsupported? is patch available?
what about patch for Opus?
uncle google doesnt know
Dne 23.1.2014 16:31, Gareth Blades napsal(a):
On 23/01/14 15:21, Marek Cervenka wrote:
hi,
which file extensios are supported in mixmonitor application?
On 24-01-14 00:37, Marek Cervenka wrote:
can someone confirm that mp3 is unsupported? is patch available?
Iirc mp3 was in asterisk-addons in asterisk 1.4 and iirc in later
versions of asterisk you can enable format_mp3 in make menuselect.
what about patch for Opus?
uncle google doesnt
On Wed, Jan 22, 2014 at 5:56 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I'm looking at setting type=peer vs type=user (in both IAX and SIP conf
entries), and I found a comment attributed to digium
(http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that
type=user is
On Thu, Jan 23, 2014 at 7:01 PM, Rusty Newton rnew...@digium.com wrote:
snip
the 1.8,11, or 12 branches. That being said, 12 is rather new and has
many significant changes that should be considered.[3]
I meant to reference link [1] of course. :)
--
Rusty Newton
Digium, Inc. | Community
On Wed, Jan 22, 2014 at 9:11 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
snip
So how do I get the Digium IP-phone to use the md5 digest authentication ??
For the benefit of the archives and those reading the list, but not
the forums - this was answered here
On Thu, Jan 23, 2014 at 8:09 AM, Igor Dvorzhak idm...@gmail.com wrote:
snip
How to move 2 of 3 users in the MeetMe conference to the newly created
MeetMe conference? Dialplan example is welcome.
Maybe something like an AMI redirect?
in the sip.conf i specified
[general]
sendrpid=rpid
trustrpid=yes
language=de
videosupport=yes
callevents=yes
caninvite=yes
There is a typo in the last line above. Should be canreinvite. AFAIK it's
obsoleted in favor of directmedia. BTW, try to set it to NO.
BTW, what is the codec order?
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