On 5/3/06, Goke Aruna [EMAIL PROTECTED] wrote:
...
#0 ast_var_name (var=0x1) at chanvars.c:71
#1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46
OUTBOUND_GROUP) at pbx.c:5904
#2 0xf5bbe1e4 in dial_exec_full (chan=0xa281820, data=0x0,
peerflags=0xf469fee8) at app_dial.c:964
On 4/30/06, Hatami Nugraha [EMAIL PROTECTED] wrote:
Hi all,
I always get this error message after I hangup a call, what does it mean ?
WARNING[8957]: file.c:583 ast_readaudio_callback: Failed to write frame
This means you hungup while asterisk was trying to play a file to you.
It should be
Please Ignore if you cannot edit the code.
You will have to modify app_dial.c in apps directory.
Look for code that calls ast_dtmf_stream(chan, ..., timeout)
The last parameter is the inter digit timeout, it can be set to as low
as 1 (1 millisec) a value of 0 it will default to 100millisecs.
The
Use:
modprobe wct4xxp vpmsupport=0
On 3/1/06, Aaron Daniel [EMAIL PROTECTED] wrote:
Does anyone know how to disable the VPM in software rather than removing
the card altogether? The canceler isn't working as well as the software
cancelers were.
Aaron
I have experienced similar problems using gmail.
Gmail certainly had some problems with emails from asterisk lists.
I donot know if it was only restricted to asterisk lists.
As not all emails were being delayed (or dropped), some of you might be under
the impression that theres no problem.
Please
here is a little explanation:
End user (You) - Your Telco -- Carrier 1 ---
Carrier 2 Carrier 3 --- Carrier 4(PTT)
--- Far End User
So basically, the Echo cancelling work backwards usually cancellation
for you would be done by
On 2/7/06, Imran Ahmed [EMAIL PROTECTED] wrote:
here is a little explanation:
End user (You) - Your Telco -- Carrier 1 ---
Carrier 2 Carrier 3 --- Carrier 4(PTT)
--- Far End User
So basically, the Echo cancelling work
AFAIK asterisk does not drop the packets, it just turns them into
silence if it detects a dtmf.
On 2/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello Friends,
I am experiencing a problem. The rtp packets which detect dtmf from inband
are being dropped. I have tried a priority ip
Step 3 The Iax client heve to send some other DTMF to the IVR.
How is the IVR still involved if the call has been transferred into a
conference room?
The IVR records the conversation between the other partecipant to the
conference and wait '#' to stop recording and a '1' to save the
AFAIK there's no DTMF option in IAX2...
IAX always sends DTMF inline, eliminating the confusion often found with
SIP.
http://www.voip-info.org/wiki-IAX
Even though no IAX client supports inband dtmf, An IAX client can send
inband dtmf which would have corrected your problem.
The problem here
On 2/1/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Imran Ahmed wrote:
Even though no IAX client supports inband dtmf, An IAX client can send
inband dtmf which would have corrected your problem.
No, it won't. No IAX2 client will start a DSP to listen for inband DTMF,
because IAX2
Here is my problem, at this point the IVR doesn't hear the dtmf sended
by the iax client, even if it can hear the dtmf sended by the first zap
channel.
I donot know if IaxComm has inband dtmf mode available, if so enable
it and see if it works.
___
You donot need multiple asterisk boxes for a single t1. A single p4
box should be helpful, you can use digiums te110p pci card for a
single pri line into the box. The same box could also be on another
network dealing with SIP.
On 1/9/06, Carlos Alperin [EMAIL PROTECTED] wrote:
All that you
I think the broken pipe issue is related with the mpg123 player,
try disabling moh and see if it behaves the same way
On 12/19/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote:
I have the same problem !!
:-(
2005/12/18, Mohammad Shokuie [EMAIL PROTECTED]:
Hi there,
Any one
Hi All,
I want to know what is the maximum number of users allowed in a single
meetme conference. How far is this number practically feasible
Thanks
Imran
___
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