Hopefully this is a simple question. How does a non-secure phone that is on a PBX connected to an asterisk over a SIP trunk communicate with a secure phone connected to the Asterisk server? I like to think that the secure call terminates on the Asterisk and the non-secure call is somehow joined to that call on the Asterisk. Same with the voice traffic, I'm once again thinking that the Asterisk proxies the calls. Just not sure. Thanks, Mitch
I'm in the process of testing a TLS/SRTP install. My experience is
improving with each new challenge, but this one is a great test of my 2
month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error
message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the
extensions and sip.conf files. *CLI> == Using SIP RTP CoS mark 5
The Asterisk Development Team has announced the release of Asterisk 1.4.40. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.4.40 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you! The following is a sample of the issues resolved in this release: * Correct issue where res_config_odbc could populate fields with invalid data.
(Closes issue #18251, #18279. Reported by bcnit, zerohalo. Tested by trev,
jthurman, elguero, zerohalo. Patched by tilghman) * Resolve issue where re-transmissions of SUBSCRIBE could break…
Thanks a lot!
Rodrigo Lang. 2011/2/28 Borin
> try this pls
> it did help to me
> On Thu, Feb 24, 2011 at 9:31 PM, Rodrigo Lang <
> email@example.com> wrote:
>> Hi to all!
>> I'm trying to create a context for integration with extensions.lua and
>> libsql.mysql, but I'm not getting to run. When I reload the module
>> pbx_lua.so the following error appears:
The problem I have been experiencing since last month is that some of my
customers are getting calls with "Asterisk
them in the middle of the night. And my asterisk server has no record of
these calls. The customers were getting irritated as you can imagine. I
guessed the only way to receive incoming calls by by-passing the
registration server is thru sip-uri calls directly to customers. I have
updated the customers atas to not accept any calls from sources other than
the registration server.…
Hi All; I would like to have two Asterisk machines to have redundancy between them, so if first machine failed then we can depend on the second machine. Because of this, I would like to know (if someone can advise me): 1) If I did modification on the configuration, how this will be applied to the other machine? 2) I am going to use E1s, so what is the method to be able to let the calls go for other Asterisk machine if first one is down (due to maintenance or upgrade)? Is there a switch or something can help…
I've tried researching this, and so far, have struggled to find any contemporary information on the issue, so I do apologize if asking this irritates people who have answered this before. I have managed to set up Asterisk 1.8 on the web server here. I have two softphones (Ekiga) able to communicate with it. So far so good. I'm now curious to see if I can link it with the PSTN phone line here. The web server in question is an Intel Atom system with a Mini-ITX motherboard