Hi Ethy,
Am 09.11.2016 um 17:13 schrieb Ethy H. Brito:
> How are these parameters available from dialplan?
>
> For instance, ${SIPURI} holds the internal "IP:port" if the client is behind
> NAT.
> I need the external IP:port
You can get the peer's signalling IP address from ${CHANNEL(recvip
Look at:
https://javiervalencia.net/2015/12/06/asterisk-en-realtime/
(Spanish)
Regards
El 09/11/2016 a las 17:06, Joshua Colp escribió:
On Wed, Nov 9, 2016, at 05:59 PM, Carlos Chavez wrote:
Is there some documentation for all the available sorcery.conf
mappings for realtime? Asteris
http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
On 2016-11-09 08:13 AM, Ethy H. Brito wrote:
Hi all
I'd like to log the client IP addr and port used for SIP and RTP *during* in a
call.
The IPs must be the real source IPs (internet accessible).
How are these parameters available
On Wed, Nov 9, 2016, at 05:59 PM, Carlos Chavez wrote:
> Is there some documentation for all the available sorcery.conf
> mappings for realtime? Asterisk already includes some tables in the
> database that are not enabled by default on the sorcery.conf like
> transports and outbound regist
Is there some documentation for all the available sorcery.conf
mappings for realtime? Asterisk already includes some tables in the
database that are not enabled by default on the sorcery.conf like
transports and outbound registrations. There are no examples in the
file on how to enable th
Does anyone know if Asterisk 13 will support T.38 Version 3?
?
Thanks
Bryant
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk
You, sir, are a genius. Thank you!
I spent ages staring at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Dial
but as soon as you gave than /n, everything is working again and I
found https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Optimization
which explains more.
Th
Hi all
I'd like to log the client IP addr and port used for SIP and RTP *during* in a
call.
The IPs must be the real source IPs (internet accessible).
How are these parameters available from dialplan?
For instance, ${SIPURI} holds the internal "IP:port" if the client is behind
NAT.
I need th
In article ,
Jonathan H wrote:
> Thank you - that makes sense. I've seen something about swapping and
> optimizing channels on the console, but I didn't realise "optimize"
> meant "not do what you wanted".
>
> OK, so here's why I'm dialling anything at all:
>
> The first dial is because I MUST l