Hi Matt,
I also tested the directmedia=yes over 3g connection ie with a public ip
but I am getting only one way audio
am I doing anything wrong?
On Wed, Jul 9, 2014 at 6:54 PM, Sameer Rathod sam...@hostnsoft.com wrote:
Hi Matt,
Thank you so much for explaining me this concept
One more
sorry I forgot to add the conf
sip.conf
[101]
type=friend
username=101
secret=101
host=dynamic
context=mkg
;nat=force_rport,comedia
;dtmfmode=rfc2833
;canreinvite=no
directmedia=yes
;directrtpsetup=yes
;avpf=yes
;encryption=yes
;disallow=all
;allow=ulaw
;icesupport=yes
[102]
type=friend
Hi there,
In one of my asterisk installation, there is a Digium E1 pri card connected.
The asterisk and card are working properly.
The problem we have is that when a storm occurs in the area, the card stops working, and E1 lines connected not rise, even restart the machine.
If however if you turn
On Thu, Jul 10, 2014 at 4:28 AM, Sameer Rathod sam...@hostnsoft.com wrote:
Hi Matt,
I also tested the directmedia=yes over 3g connection ie with a public ip but
I am getting only one way audio
am I doing anything wrong?
If you are getting one way audio when direct media is enabled, then
one
On Thursday 10 Jul 2014, Ismael Gil wrote:
Hi there,
In one of my asterisk installation, there is a Digium E1 pri card
connected. The asterisk and card are working properly.
The problem we have is that when a storm occurs in the area, the card
stops working, and E1 lines connected not
Hi guys.
Does somebody knows how to get the concurrent calls from the dial plan?
Or.
How can i control not to run more than n simultaneus agi applications?
Thanks in advance.
rv
--
_
-- Bandwidth and Colocation Provided by
you can use GROUP and GROUP_COUNT
n,Set(GROUP()=aname)
n,GotoIf($[${GROUP_COUNT(aname)} 8]?${EXTEN},200)
200,Hangup
On Thu, Jul 10, 2014 at 3:24 PM, Rafael Visser visser.raf...@gmail.com
wrote:
Hi guys.
Does somebody knows how to get the concurrent calls from the dial plan?
Or.
How can
Works fine..
Thanks Asghar!
rv
2014-07-10 9:35 GMT-04:00 Asghar Mohammad asghar...@gmail.com:
you can use GROUP and GROUP_COUNT
n,Set(GROUP()=aname)
n,GotoIf($[${GROUP_COUNT(aname)} 8]?${EXTEN},200)
200,Hangup
On Thu, Jul 10, 2014 at 3:24 PM, Rafael Visser visser.raf...@gmail.com
I would have to read the source code to know for sure. Is it too much trouble
to try my suggestion?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Wednesday, July 09, 2014 8:58 PM
To: Asterisk Users Mailing List
I cannot wait for the regular bug-patch process to play out. I am
considering hiring a developer to fix bug 24015, and of course submit the
patch for the bug. The issue is simple, the app Transfer does not transfer
when using PJSIP.. I called Digium and they said that they do not do this
kind of
Hi
I'm using a macro to dial in a AEL dialplan. The problem is the macro do
not set the field CDR(dst), showing only ~~s~~.
I tried various configurations, but without solutions.
This is the macro:
macro dial-out(destno,dialstring,route_descr,interno) {
__TRANSFER_CONTEXT=ipbx;
if(${interno} =
The Asterisk Development Team has announced the release of Asterisk 1.8.29.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.29.0 resolves several issues reported by the
community and would have not been
The Asterisk Development Team has announced the release of Asterisk 12.4.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.4.0 resolves several issues reported by the
community and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 11.11.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.11.0 resolves several issues reported by the
community and would have not been possible
On Thu, Jul 10, 2014 at 9:08 AM, CDR vene...@gmail.com wrote:
I cannot wait for the regular bug-patch process to play out. I am
considering hiring a developer to fix bug 24015, and of course submit the
patch for the bug. The issue is simple, the app Transfer does not transfer
when using
Dear All,
I have different Asterisk Servers most of them are version 1.8 - I have
recently upgrade to Asterisk version 11 on 2 servers.
I have Jabber ( chan_gtalk ) configured on 1.8 version and it is working
within all 1.8 version servers.
I have XMPP ( chan_motif ) configured on 11 version
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