* You are viewing the archive for August, 2011

phone + video

Hi all,

I know that a lot of people have negative experiences with
grandstream-2000, but personally. i’d only the repace one poweradapter
after three years…

So, can anybody give some comment on one of their recent models,
the GXV-3175 (the one with the 7″ display)
I’m looking for a phone with video capabilities, as i don;t want to
limit my self to testing with softphone..

HtH, Hans

Asterisk 1.8.6.0 Now Available

The Asterisk Development Team announces the release of Asterisk 1.8.6.0.
This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.6.0 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:

* Fix an issue with Music on Hold classes losing files in playlist when
realtime
is used.
(Closes issue ASTERISK-17875. Reported by David Cunningham. Patched
by Igor
Goncharovsky)

* Resolve a potential crash in chan_sip when utilizing auth= and
performing a
‘sip reload’ from the console.
(Closes issue ASTERISK-17939. Reported by wdoekes. Patched by Richard
Mudgett)

* Address some improper sql statements in res_odbc that would cause an
update
to fail on realtime peers due to trying to set as “(NULL)” rather than an
actual NULL.
(Closes issue ASTERISK-17791. Reported by marcelloceschia. Patched by
Tilghman
Lesher)

* Resolve issue where 403 Forbidden would always be sent maximum number
of times
regardless to receipt of ACK.
(Patched by Richard Mudgett)

* Resolve issue where if a call to MeetMe includes both the dynamic(D) and
always request PIN(P) options, MeetMe will ask for the PIN two times:
once for
creating the conference and once for entering the conference.
(Patched by Kinsey Moore)

* Fix New Zealand indications profile based on
http://www.telepermit.co.nz/TNA102.pdf
(Closes issue ASTERISK-16263. Reported, Patched by richardf)

* Segfault in shell_helper in func_shell.c
(Closes issue ASTERISK-18109. Reported by Michael Myles, patched by
Richard
Mudgett)

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

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.6.0

Thank you for your continued support of Asterisk!

cli command show codecs

Core show channels verbose is probably your best bet. I think the answer also depends on your * version.

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of RSCL Mumbai
Sent: Wednesday, August 31, 2011 10:44 AM
To: Asterisk Users Mailing List – Non-Commercial Discussion
Subject: [asterisk-users] cli command show codecs

Hi,

Is there a CLI command which will tell me the codec used for active calls and if transcoding is happening ?

Thx
Sans

8th September 2011

Update: the location is fixed now, it happens in the Unibräu in the
Alten AKH Campus. We start at 7pm.

Please find details at
http://sip-router.org/10-years-ser-vienna/

If you plan to attend please send me an e-mail to make sure that we have
enough seats.

Transfer to VoiceMail Asterisk 1.6

Hello,
I’m using Asterisk 1.6 with Polycom SoundPoint 650, everything is running
fine except that I can’t program a button on Polycom to transfer inbound
call to Voicemail directly.

I have the following in my extension.conf

exten => _547xx,1,Voicemail(${EXTEN:1}@default,u)

Reception can transfer directly to VoiceMail when dialing digit 5 I want to
make a softkey on Polycom 650 does anybody know how to accomplish tranfering
directly to VoiceMail?

Thanks,
Motty

MOH making calls appear hung up

I noticed the CLI shows that the music on hold actually stops for some reason?

Here’s the output of my CLI:
Connected to Asterisk 1.6.2.19 currently running on localhost (pid = 6363)
Verbosity is at least 28