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Question about how traffic passes from phones

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

duplicate keys change from zaptel to dahdi 2.4.0

I just updated from zaptel to dahdi 2.4.0
I dont recall missing keys or duplicating keys with zaptel.
With dahdi 2.4.0 I tried serveral calls and I was trying to enter 204
and I got 2204.

Is there another or new parameter I need to tweek?

Thanks,

jerry

TLS/SRTP calls go to circuit busy.

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

Failover Routing

Look like you should work with channel status variable. If channel not
answer then jump on 5xx

Asterisk 1.4.40 Now Available

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 presence.
(Closes issue #18075. Reported by mdu113. Patched by twilson)

* Resolve issue in res_odbc where it may crash when a query fails.
(Closes issue #18243. Reported, patched by ks3)

* Fix CPU spike when pressing DTMF after agent login.
(Closes issue #18130. Reported by rgj. Patched by jpeeler)

* Fix cross-compiling issue.
(Closes issue #18301. Reported, patched by abelbeck)

* This version of Asterisk includes the new Compiler Flags option
BETTER_BACKTRACES which uses libbfd to search for better symbol information
within both the Asterisk binary, as well as loaded modules, to assist when
using inline backtraces to track down problems.
(Patched by tilghman)

* Resolve several issues with DTMF based attended transfers.
(Closes issues #17999, #17096, #18395, #17273. Reported by iskatel, gelo,
shihchaun, grecco. Patched by rmudgett).
NOTE: Be sure to read the ChangeLog for more information about these changes.

* Fix regression that changed behavior of queues when ringing a queue member.
(Closes issue #18747, #18733. Reported by vrban. Patched by qwell.)

Additionally, this release has the changes related to security bulletin
AST-2011-002 which can be found at
http://downloads.asterisk.org/pub/security/AST-2011-002.pdf

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.40

Thank you for your continued support of Asterisk!

CEL and PGSQL

Hi,

Would someone know where I can download the CEL schema for (create commands) for PostgreSQL please ?