On Mon, 2012-05-07 at 02:49 +0530, Mitul Limbani wrote:
For 100% High Availibility and Hot Failover, I would recommend one of
those Red-fone Fonebridges.
Also getting 800 Phones all register on single server is crazy, add a
SIP proxy to distribute load evenly between 2 Ast boxes.
For
Dear Kevin,
Thanks for your answer.
At least in this case, only TOP DOGS must be encrypted End-To-End while
they are talking between them.. so Asterisk should be right solution,
they would not take advantage of .. some Asterisk Features while they
are talking between them, but all other
Hello,
when a call comes in and is answered by colleague A, this colleague A
sees the CallerID of the external calling number.
When colleague A transfers the call to colleague B, attended or
unattended, then colleague B sees the number of colleague A on his
screen while talking to the
According to the best information I have access to, blind transfer is the
only way to do this pre Asterisk 10.X
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Tuesday, May 08, 2012 7:14 AM
To: Asterisk Users
- Original Message -
From: Sebastian scg...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, May 7, 2012 5:26:14 PM
Subject: Re: [asterisk-users] 1.8 busypatterns
Can you point me to the commit to see if i can
- Original Message -
From: Jonas Kellens jonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, May 8, 2012 7:13:30 AM
Subject: [asterisk-users] Asterisk 1.8 Transfer CallerID
Hello,
when a call
On 05/08/2012 08:50 AM, Jonathan Rose wrote:
- Original Message -
From: Jonas Kellensjonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Sent: Tuesday, May 8, 2012 7:13:30 AM
Subject: [asterisk-users] Asterisk 1.8 Transfer
On 05/08/2012 04:24 PM, Kevin P. Fleming wrote:
On 05/08/2012 08:50 AM, Jonathan Rose wrote:
- Original Message -
From: Jonas Kellensjonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Sent: Tuesday, May 8, 2012 7:13:30
Hi,
Am Dienstag, den 08.05.2012, 14:13 +0200 schrieb Jonas Kellens:
Hello,
when a call comes in and is answered by colleague A, this colleague A
sees the CallerID of the external calling number.
When colleague A transfers the call to colleague B, attended or
unattended, then colleague B
On 05/08/2012 04:32 PM, Karsten Wemheuer wrote:
Hi,
Am Dienstag, den 08.05.2012, 14:13 +0200 schrieb Jonas Kellens:
Hello,
when a call comes in and is answered by colleague A, this colleague A
sees the CallerID of the external calling number.
When colleague A transfers the call to colleague
On Mon, May 07, 2012 at 10:10:24AM -0600, Greg Woods wrote:
On Mon, 2012-05-07 at 10:26 -0500, Russ Meyerriecks wrote:
On Sun, May 06, 2012 at 10:42:16AM -0600, Greg Woods wrote:
I have a Digium TDM400P card that appears to have died. The first noted
symptoms were that dahdi would fail to
On Mon, 2012-05-07 at 10:26 -0500, Russ Meyerriecks wrote:
There were a few compatability issues with the tdm400's pci interface chip
and certain motherboards
Interesting idea, but it's been running in this same server with the
same motherboard for at least two years now, and this problem
Hi All,
I am posting this thread with the hope that someone in UK (or elsewhere)
had a similar issue:
Our issue is simple, we cannnot use our ISDN line, when watching asterisk
console it gives a bunch of ISDN errors where the following is probably the
most relevant:
Span: 4 TEI=0 MDL-ERROR (J):
Hello Marcus,
Had the same problem, looked in the internet and found your question.
Since I now have an answer
I will put it here! :)
In the dialplan I had this:
exten = 4000,1,Dial(Dahdi/4)
exten = 4000,n,hangup()
exten = _4,1,Dial(IAX2/PBX/${EXTEN:1})
exten =
I am learning how to use AMI and I am having 1 problem.. When I make a call
to my mobile phone and when I answer it - it get disconnected/hangup right
away.
Why is that? What is the solution to stop that?
For example:
ACTION: Originate
Channel: SIP/447XXX@vpsprovider
Exten: 210
Priority: 1
It is likely the 60 second timeout you are providing. Or it could be the
hangup() command in the 210 context.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
Sent: Tuesday, May 08, 2012 3:11 PM
To:
No, that 'timeout' option is when I don't answer the call.
My problem is when I DO answer the call, it get disconnected right away.
Yes hangup() get executed right away when I answer the call.
On Tue, May 8, 2012 at 9:17 PM, Danny Nicholas da...@debsinc.com wrote:
It is likely the 60 second
Since you are Originating the call, the hangup command isn't needed. Remove
and reload.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
Sent: Tuesday, May 08, 2012 3:20 PM
To: Asterisk Users Mailing List - Non-Commercial
I have tried that and that did not fixed the problem,
However, I have added this in the dialplan:
exten = 210,n,Wait(60)
That will hangup the call after 60 seconds...
That is fine by me but now Monitor() dont even work now... it does not
record a call...?
On Tue, May 8, 2012 at 9:22 PM,
Try MixMonitor instead of Monitor. I think it is more robust.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shahid H
Sent: Tuesday, May 08, 2012 3:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
We are using snom 821's and it works as described with sendrpid and
trustrpid both set. We are using realtime for sip peers and users.
Version 1.8.10.0
Cheers
Stephen
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On Mon, 07 May 2012 06:40:46 -0600
Nunya Biznatch aster...@ihearbanjos.com wrote:
I need to plan to use FreePBX on all Asterisk Servers, but I don't
intend to install it until I'm in regular MAC maintenance mode.
It is ashame you are going this far with your setup to rely on
FreePBX.
Don't blame me guys, but if you are a new to VOIP and aint't using asterisk
for a while, checkout freeswitch (freeswitch.org). Also do not forget at
least about this
http://www.opensips.org/html/docs/modules/1.6.x/load_balancer.html for
loadbalance to your pbx boxes. Just my 2 cents.
2012/5/9
Hi Shahid,
I am in favor of asterisk for what it is doing to your call. When you send
an AMI event like the one you wrote it sends the A party invite right away
w/o going into any context/extension. As soon as the A-party answers the
call Asterisk manager connects/lands A-channel to the test
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