For those having the same problem, my solution was to upgrade to the
newest firmware on the Linksys WAG160.
It seemed a NAT-problem because NAT-ting was not correctly handled by
the firmware.
Jonas.
On 05/21/2010 02:41 PM, Jonas Kellens wrote:
Hello list,
I am confronted with the following p
Hello list,
I am confronted with the following problem :
making a call only leasts for about 30 seconds, then the call is ended.
The CLI shows :
[May 21 14:31:50] WARNING[25345]: chan_sip.c:1980 retrans_pkt: Maximum
retries exceeded on transmission 954539948-506...@192.168.1.100 for
seqno 1
[asterisk-users] Hanging up a call by DTMF
Hello,
Is it possible to hangup an active call by simply sending a DTMF code to
Asterisk for example # code.
If yes, What function to use in the dialplan.
Thanks.
___
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Hello,
Is it possible to hangup an active call by simply sending a DTMF code to
Asterisk for example # code.
If yes, What function to use in the dialplan.
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Hello, never posted to a mailing list before. I've been trying to work out this
problem for quite awhile now. I have a PHP script which is run whenever an
emergency situation happens. The script connects to the AMI and originates
calls to previously defined "emergency" extensions. I'm looking fo
at (3) i need to know if this is a normal hangup (A or B has hung up) or
if it is a result of the transfer.
It's normal, who start's the conference can't hangup.
On 6/20/07, Julian Lyndon-Smith <[EMAIL PROTECTED]> wrote:
Is there anyway on knowing in the "h" extension if a call has been end
Is there anyway on knowing in the "h" extension if a call has been ended
as a result of a transfer ?
i.e.
1) A calls B.
2) B transfers A to C.
3) B gets hung up.
4) A talks to C
at (3) i need to know if this is a normal hangup (A or B has hung up) or
if it is a result of the transfer.
Julian
Lex Lethol wrote:
As far as I know when I setup a 3-way on something like a cisco will
disconnect everyone when the middle (person who setup the conference)
hangs up.
The problem I describe happens on ATAs and the like that uses flash to
put on hold while setting up the second call.
I am not su
As far as I know when I setup a 3-way on something like a cisco will
disconnect everyone when the middle (person who setup the conference)
hangs up.
The problem I describe happens on ATAs and the like that uses flash to
put on hold while setting up the second call.
I am not sure about other phon
How does this compare to using the conference features on a SIP phone,
say a Snom? I have used a Snom many times for an ad-hoc conference,
without any troubles...
PaulH
On Sun, 2007-01-07 at 18:12 -0600, Lex Lethol wrote:
> Apparently asterisk's default way to a 3-way conference lets the user
>
Apparently asterisk's default way to a 3-way conference lets the user
in the middle hangup and the other parties stay on the conversation.
This is great some times but it creates quite a bit of problems when
trunks dont have disconnect supervision or when trying to do
accounting and billing on the
hello!
i'm trying to implement a callback feature. to accomplish this, i've
written a python script(callback.agi) that starts another script as a
independent process(with spawnl), without asterisk waiting for the
other script (callback_dead.sh) to finish before it goes to the next
extension. runni
On Tue, 2005-10-04 at 11:12 -0700, Jesse Keating wrote:
> I've got an issue w/ 1.2.0beta1, where if I call VoiceMailMain from a
> sip phone, and then either put in incorrect passwords or just hang up, I
> never get a Spawn Extension that hangs up the call, and my sip phone is
> not capable of makin
I've got an issue w/ 1.2.0beta1, where if I call VoiceMailMain from a
sip phone, and then either put in incorrect passwords or just hang up, I
never get a Spawn Extension that hangs up the call, and my sip phone is
not capable of making any more calls until I restart the daemon. Can
anybody help m
I would like to do the following:
"A" calls "B"
"C" calls "A"
"A" hears call waiting beep and flashes the line to
talk to "C"
::Here's where I run into a problem::
"A" hangs up on "C" and immediately returns to a
conversation with "B"
The only way I have got this to work is if "C" hangs
up. Then
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