how can I configure chan_dahdi.conf?
Must I create Channels?
Can you help me please?
Thank you very much.
_
AMICALEMENT...
and in calling...
Please help me...
Thank you in advance
AMICALEMENT…
__
Manu
- E-Mail : et...@manu-dpk.net
- TimeLine Twitter : @manudpk
- Skype : manu-dpk
Découvrez mes compositions musicales personnelle
...
Thank you for your help...
AMICALEMENT…
__
Manu
- E-Mail : et...@manu-dpk.net
- TimeLine Twitter : @manudpk
- Skype : manu-dpk
Découvrez mes compositions musicales personnelles ainsi que mes podcasts :
http
at...
I'm blind, I use asterisk with app-directory() to make a personal directory,
and to make a audio lookcallid anon cement.
I'd like build asterisk on osx to use voiceOver voice, are better quality.
That's why...
AMICALEMENT…
___
EMENT…
__
Manu
- E-Mail : et...@manu-dpk.net
- TimeLine Twitter : @manudpk
- Skype : manu-dpk
Découvrez mes compositions musicales personnelles ainsi que mes podcasts :
http://www.manu-dpk.net
___
Respectons l’environn
7; is now Reachable. (2ms / 2000ms)
[Apr 12 18:31:45] [1;33mNOTICE[0m[12303]:
[1;37mchan_sip.c[0m:[1;37m23569[0m [1;37mhandle_response_peerpoke[0m:
Peer 'ippi_outgoing' is now Reachable. (15ms / 2000ms)
[0m
Please...Do somebody help me ?
AMICALEMENT…
transfert on my mac mini server.
AMICALEMENT…
__
Manu
- E-Mail : et...@manu-dpk.net
- TimeLine Twitter : @manudpk
- Skype : manu-dpk
Découvrez mes compositions musicales personnelles ainsi que mes podcasts :
Yes i tried it and i am not happy of the result
Envoyé de mon iPhone!
AMICALEMENT…
>__
>
> Manu
>
>- E-Mail : et...@manu-dpk.net
>- TimeLine Twitter : @manudpk
>- Skype : manu-dpk
>
> Découvrez mes compositions music
Up,
Does any body have solutions ?
1 Asterisk 12 is compiled,
2. No sound for ivr, voicemail, out calls in calls.
3. I can't use TTS from google, I want use "say" function on OsX.
4. Cli error
Privilege escalation protection disabled!
See https://wiki.asterisk.org/wiki/x/1gKfAQ for more details.
All,
I am trying PSTN Termination using Asterisk and SER.
FXS--->SER--->--->FXO--->PSTN
I could configure FXS and SER. Do i need to register SER with Asterisk to forward call
to Asterisk.
Also, how to terminate there calls to respective FXO gateways. For achieving this, do
i need to register
All,
I am trying PSTN Termination using Asterisk and SER.
FXS--->SER--->--->FXO--->PSTN
I could configure FXS and SER. Do i need to register SER with Asterisk to forward call
to Asterisk.
Also, how to terminate there calls to respective FXO gateways. For achieving this, do
i need to register
Hi,
You can use function SIP_HEADER instead. See
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
Manu Mehta
A R I C E N T
Plot-17, Sector 18, Gurgaon 122015,
Haryana, India
Main +91.124.4095888 x3274
Fax +91.124.4095912
"Jean-Marc Salsa&quo
n Asterisk console that
"res_features.c:1415 ast_bridge_call: Bridge failed on channels
SIP/ua1-ac750040 and AsyncGoto/Local/[EMAIL PROTECTED],1"
Here is my dial plan:
[manu]
exten => ca1,1,Dial(SIP/ca1,,wWtTkKrR)
[nway-conf]
exten => _.,1,Answer
exten => _.,n,Set(CON
existing bridged channel
when a subsequent call is received, i can then redirect that channel to
play tone.
The problem is how can i get hold of the bridged channel in the first
place? Also is there a better way of accomplish call waiting.
TIA,
Manu Mehta
A R I C E N T
Plot-17, Sector 18
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