Re: [asterisk-users] SIP and RTP port and IP addresses

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

Re: [asterisk-users] sorcery.conf mappings

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?

Re: [asterisk-users] SIP and RTP port and IP addresses

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

Re: [asterisk-users] sorcery.conf mappings

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

[asterisk-users] sorcery.conf mappings

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

Re: [asterisk-users] Asterisk 13 T.38 Version 3?

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:

Re: [asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)

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.

[asterisk-users] SIP and RTP port and IP addresses

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

Re: [asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)

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".