Sorry there is a mistake, actually I don't redirect executing channel:
exten = s,n,ChannelRedirect(${CHANNEL_TO_REDIRECT},context_2,AMD,1)
I can't use Goto because the ${CHANNEL_TO_REDIRECT} is active channel and
Goto drops (hangs-up) it.
So is there a way to pass variable to the target context
And I'm pretty sure if you look at any of those peers that have a
non-5060 port, the routers in front of them will rewrite packets
destined for ports 53277, 4121, 47822 etc. to the proper corresponding
internal IP:port where something is listening. The router of my
provider won't. It
Am 21.02.2014 15:12, schrieb Andres:
Wow, if this is the case then I would be changing VM providers
immediately.You would have problems not only with Asterisk but with
most other services you wanted to host on it. There are many VM
providers out there that work just fine with Asterisk even
Hi,
Here is my scenario.
I have a SIP call between two SIP endpoints. A calls B.
During the ringing, B disconnects (network cable is unplugged).
But A continue ringing forever (until the dial timeout) even if asterisk
detects that B is disconnected with the qualify.
Is there any setup or
Dear Mr. Newton
Thank you for your response. I red the wiki and sip.conf sample file and I
know about directmedia option. Actually these options are for times that
you know about your connected networks (you know which clients are behind
NAT and which are not). But my configuration is different. I
On Fri, Feb 21, 2014 at 11:13 AM, Gholamreza Sabery gr.sab...@gmail.com wrote:
Dear Mr. Newton
Thank you for your response. I red the wiki and sip.conf sample file and I
know about directmedia option. Actually these options are for times that you
know about your connected networks (you know
Anyway, thank you so much. ;-)
On Fri, Feb 21, 2014 at 9:32 PM, Rusty Newton rnew...@digium.com wrote:
On Fri, Feb 21, 2014 at 11:13 AM, Gholamreza Sabery gr.sab...@gmail.com
wrote:
Dear Mr. Newton
Thank you for your response. I red the wiki and sip.conf sample file and
I
know about
