Hello,
My problem is, outgoing calls (from asterisk to CCM) work fine but incoming
(from CCM to Asterisk) does not work because of CCM is trying to use g729
over SIP trunk. I have found that link after a quick search. Problem is the
same as in link below (However my Asterisk version is 1.8.13)
Hello,
I have a question about directmedia or canreinvite, I have experience that
whatever I set directmedia=yes or no. After I run sip show settings.
all settings looks the same.
My question is how I could make sure from sip show settings that my
directmedia configuration is applied.
Thanks
Hi,
If you use rfc2833 and set directmedia=yes, diect media will not work. You
must set other value like SIP info in order to make diectmedia work ...
Regard/chui kingh man
寄件人︰ Shitian Long longst...@gmail.com
收件人︰ asterisk-users@lists.digium.com
傳送日期︰
Hi,
Let say we have a call center from which agents get calls from both
on-queues and off-queues calls (ie calls passing through queues or
direct calls non passing through queues).
Regulation here specify prior consent before recording call.
How can I best enforce this compliance ?
What would
Hello,
To my surprise, with asterisk 1.8 (I've not tried with other versions), it
seems you cannot set CDR's userfield from within a dialplan macro called by
dynamic features.
See :
testfeature = *321,self/callee,Macro,toto
[macro-toto]
exten = s,1,Verbose(0,Into macro-toto with CDR(src) set
Possibly switch to using subroutines instead of Macros. Macros are being
deprecated in place of subroutines.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Olivier oza_4...@yahoo.fr
To: Asterisk Users Mailing List - Non-Commercial Discussion
I'm not using the DHCP server configuration and IP addresses assigned in
the network are manual and there are no clashes found in the network.
The version of Asterisk I'm using is 10.4.2. I think there might be some
issues in this version perhaps I may try to upgrade to 10.12.
UDPTL
2013/1/17 Onur Cem Çelebi occel...@gmail.com
Hello,
My problem is, outgoing calls (from asterisk to CCM) work fine but
incoming (from CCM to Asterisk) does not work because of CCM is trying to
use g729 over SIP trunk. I have found that link after a quick search.
Problem is the same as in
Hey all.
RE: Conf Bridge.
I am looking into a project that would need 8 to 10 thousand parties in a
single conference.
Most would be on mute but 5 to 6 would be presenters.
Is the new conf bridge solid enough to handle this kind of load?
Any ideas on hardware projections?
If not 8 to 10
On Thu, Jan 17, 2013 at 3:02 PM, Bryant Zimmerman brya...@zktech.com wrote:
Hey all.
RE: Conf Bridge.
I am looking into a project that would need 8 to 10 thousand parties in a
single conference.
Most would be on mute but 5 to 6 would be presenters.
Is the new conf bridge solid enough to
Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a
2u server for our small business's phones system.
We are using some Polycom Soundpoint IP phones. The whole thing came crashing
down over the Holidays and as of right now that's about
all we have working right now
On Thu, Jan 17, 2013 at 3:05 PM, Joe Ruffolo j...@mrkgroup.com wrote:
Hi all! In need of some serious help. We currently run Trixbox and Cent Os
on a 2u server for our small business’s phones system.
We are using some Polycom Soundpoint IP phones. The whole thing came
crashing down over
On 01/17/2013 09:05 PM, Joe Ruffolo wrote:
Hi all! In need of some serious help. We currently run Trixbox and Cent
Os on a 2u server for our small business’s phones system.
Afaik Trixbox is no longer maintained and their forum are hardly active
anymore so it may be a bit of a challenge to get
Dears;
I am using asterisk for call center and I used also VICIDIAL. But they are fine
for voice, I need the agents to be able to handle email and web chat messages
as long with the voice calls, in addition to be integrated with the social
media like Facebook and twitter.
Where I can find
Hi Joe
On 18/01/2013, at 9:05 AM, Joe Ruffolo j...@mrkgroup.com wrote:
Hi all! In need of some serious help. We currently run Trixbox and Cent Os on
a 2u server for our small business’s phones system.
We are using some Polycom Soundpoint IP phones. The whole thing came crashing
down over
From: Andrew Latham lath...@gmail.com
Sent: Thursday, January 17, 2013 3:04 PM
To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Conf Bridge
On Thu, Jan 17,
Hey all
For some reason the mailing list is sending all messages from the sending
party.
This makes it less than ideal when responding; as selecting reply goes to
the person and not the list.
Can we have it set back to the old way please?
Thanks Andrew for pointing this out to me.
Bryant
On Thu, Jan 17, 2013 at 6:32 PM, Bryant Zimmerman brya...@zktech.com wrote:
Hey all
For some reason the mailing list is sending all messages from the sending
party.
This makes it less than ideal when responding; as selecting reply goes to
the person and not the list.
Can we have it set back
On 18/01/2013, at 12:37 PM, Andrew Latham lath...@gmail.com wrote:
On Thu, Jan 17, 2013 at 6:32 PM, Bryant Zimmerman brya...@zktech.com wrote:
For some reason the mailing list is sending all messages from the sending
party.
This makes it less than ideal when responding; as selecting reply
I get direct replies when people reply to my posts. I thought that was just
'cause they wanted to make sure I saw their replies!
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pete Mundy
Sent:
On 18/01/2013, at 4:28 PM, Jim Boykin boykin...@gmail.com wrote:
Hi,
We are looking for the web based console for our asterisk system. We
came across AsteriskNow but it's kind of bundle and hence not usable
for us. What we need is a separate GUI package which we can add to our
existing
Alright for anyone who ever runs into this in the future, the problem seems
to be resolved by
a) removing the lines Set(Channel(language)=) before the Dial and
possibly
b) using the flags 'dI' with followme app
I guess when using Followme, just don't try and set any another variables
that
Hi,
i am using elastix 2.3 and created some dahdi extensions,now i dialing
between the extensions i.e like 2000 to 2001 , but there is delay of 3 to 4
second before it ring the destination. so cany anyone know how fix it so
that after dialing the digits the destination should ring . without any
If you dial 2001# does it complete the call immediately?
Your dial plan may be ambiguous about numbers starting with 2, so it waits
a few seconds to see if you're going to dial a longer number.
--Don
From: asterisk-users-boun...@lists.digium.com
Hi,
yes if i press # then immediately ring , i configured all these by GUI
only so how should i fix this issue??
--
Upendra
On Fri, Jan 18, 2013 at 11:06 AM, Don Kelly d...@donkelly.biz wrote:
If you dial 2001# does it complete the call immediately?
** **
Your dial plan may be
Thanks for reply Leandro.
We have installed g279 codec in Asterisk box.Even if not so, there is no
problem outgoing (from Asterisk to CCM) calls. But after i searched the
issue, i figured out that CCM 4.x does not let g729 codec to pass through
over SIP trunk. This is limited only in CCM. If we
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