Displaying 20 results from an estimated 5000 matches similar to: "Asterisk 302 Moved Temporarily"
2019 May 31
1
[mail-crypt-plugin] Password Query for Folder Keys questions
> Can you try
>
> doveadm -o plugin/mail_crypt_private_password=desired_password mailbox > cryptokey generate -u user -UR
>
> Aki
I tried that and got the following:
user at host:~$ doveadm -o plugin/mail_crypt_private_password=desired_password mailbox > cryptokey generate -u user -UR
Folder Public ID
user at host:~$
Then I sent a new email to the mail server, and I
2016 Jul 04
3
Updating release docs and minimal tools
Folks,
I'm in the process of updating this page:
http://llvm.org/docs/HowToReleaseLLVM.html
which had its last review in the middle ages.
In particular, I'm removing "dragonegg" from the list of requirements,
changing the "build" instructions to use the test-release script (by
pointing it to the right doc [1]), and updating the list of
requirements.
CMake is now
2012 Feb 07
10
Ruby Developer position
Please let me know your interest in following.
Location: Columbia, SC
Duration: 12 months+
Rate: $65/hr 1099/c2c
Required Skills:
RUBY, RAILS, GIT, MYSQL, CUCUMBER, RSPEC, JQUERY, EXCELLENT ORAL AND WRITTEN
COMMUNICATION SKILLS, TEST-DRIVEN DEVELOPMENT, LINUX, OS X, JSON, COMMAND
LINE, SQL, SSH, HAML, SCSS
Thanks
Sandeep
Sandeep Jain
Software People Inc.
www.softwarepeople.us
2019 May 31
1
[mail-crypt-plugin] Password Query for Folder Keys questions
So when I tried this way I got the following output:
user'@'host:~$
doveadm -o plugin/mail_crypt_private_password=desired_password mailbox > cryptokey generate -u user -UR
user'@'host:~$
And when I tried this way I got the following output:
user'@'host:~$doveadm -o plugin/mail_crypt_private_password=desired_password mailbox cryptokey generate -u user -UR
Folder
2006 Aug 01
1
Subscribe
2010 Jun 07
0
No subject
When I load the codec as an Audio Codec the call to AudioCodecInitialize appears to work but if I then get the IsInitialized property that says it is not initialized - so looks like that is my problem, and ties up with the crash in vorbis_synthesis_start(). Looks like vorbis_synthesis_init() has not been called, perhaps.
There's sample code out there that suggests people have got this
2010 Sep 17
0
5-7 second delay in connecting outgoing FXO calls
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2017 Nov 06
3
ORC JIT and multithreading
2006 Feb 13
4
DRM and Ogg Vorbis ??
Dear all,
I would like to know whether it is possible to use any DRM scheme
with Ogg-Vorbis ?
My idea would be to distribute music commercially and be able
guarantee to artists that they won't be copied freely.
Thanks for your advice.
Regards,
Ulrich
ulrich@bbtest.roxr.com
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2005 Feb 28
1
Weird behaviour on incoming DIDs
Folks,
I have a problem here. I have 2 DIDs, one a 415 number
and the other a 650 number. I have my extensions.conf
set up to handle both of them exactly the same way,
passing them to an internal context. When _I_ dial
either DID, I get exactly the same behaviour that I
have specified (the call is answered, and then I play
my own welcome mesage, then handle any extension
dialed).
However, when
2015 Nov 22
0
可能您的账户已经被盗用。被不法分子利用发送不良信息！
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<HTML><HEAD>
<META content="text/html; charset=utf-8" http-equiv=Content-Type>
<META name=GENERATOR content="MSHTML 8.00.7601.18667"></HEAD>
<BODY>
<TABLE
style="WIDOWS: 1; TEXT-TRANSFORM: none; BACKGROUND-COLOR: rgb(255,255,255); TEXT-INDENT: 0px;
2004 Dec 06
1
SIP response 302 "Moved Temporarily "
Does Asterisk 1.0.2 support 302 redirects? With 1.0.1 I get:
Got SIP response 302 "Moved Temporarily"
When forwarding the call to other SIP server.
This is a "bug":
http://lists.digium.com/pipermail/asterisk-users/2004-May/045774.html
---
Jan Baggen - jbaggen@ip2.nl
IP2 Internet BV / http://www.ip2.nl
2010 Sep 06
1
Dial timeout and SIP 302 Moved Temporarily
Hi,
With a 1.4.35 or 1.6.1.19, I'm facing this behaviour :
- extension 7002 is a SIP hard phone currently configured to forward
incoming calls to extension 7003, when a call is unanswered within a 10s
time frame
- when extension 7001 is calling extension 7002 with a Dial(SIP/7002,20)
statement and no one answers, then :
- after 10s, Asterisk receives "SIP 302 Moved temporarily"
2007 Mar 30
0
unconditionally redirecting incoming calls by 302 Moved Temporarily messages doing right accounting
Dear all,
In my Asterisk 1.2.17 architecture different levels of permissions are
established using different contexts that hierarchically include more
permissive contexts until default context is reached.
In default context there are only local extensions, only in more
restricted contexts there are the PSTN access.
So, if some user dials some number, Asterisk looks which context that
user
2014 Mar 16
0
302 Moved Temporarily and channel variable
When a call is transferred to another extension using a blind transfer,
asterisk keeps traces of who is transferring in the BLINDTRANSFER variable.
If instead the call is "forwarded" using most phone call forward feature, a
302 Moved Temporarily is sent back to asterisk
-- Called SIP/104-DEVEL
-- Got SIP response 302 "Moved Temporarily" back from
83.211.***.***:5063
2013 May 07
1
passing '302 moved temporarily' back to the SIP provider
Hello,
I 'm looking for a way to pass the '302 moved temporarily' received from the SIP device
back to the SIP provider.
Here is the setup:
Some SIP phones are connected to an Asterisk System version 1.8.
External connection to the public network is also done via SIP to a VoIP provider.
Phone A has a CFW all calls to a phone number in public network (Mobile Phone)
incoming call to
2004 Dec 18
2
Problem with 302 "Moved Temporarily" Do not disturb
I have some Cisco 7905 phones with the SIP load 1.02.00(040406A).
When the phone is off-hook but no call has been placed, or when the Do
Not Disturb is activated, the phone returns a 302 "Moved Temporarily"
message back to asterisk as follows:
-----------
-- Executing Dial("SIP/5060-0811bb00", "SIP/9871234|20|Ttr") in new stack
-- Called 9871234
-- Got SIP response
2008 Jan 04
1
Unable to forward call on SIP channel after SIP response 302 Moved Temporarily
Hi,
I have the following problem that when asterisk receives SIP response 302 it
cannot forward the call
I get such debug:
[Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel
type registered for 'Local'
[Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to
create local channel for call forward to 'Local/poczta at routing-sip' (cause =
66)
2012 Dec 12
1
Polycom phones and ring no answer/302 Moved Temporarily
I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8.
Setting forwarding for "Always" works as expected; the phone issues a 302 Moved Temporarily, and Asterisk shifts the call to the new location.
Setting forwarding to "No Answer" means a 302 never gets issued. It just rings and eventually goes to voicemail. Watching with Wireshark, I never see a
2016 Sep 02
3
Trouble getting peer variable (sip username) on 302 Moved Temporarily
Hello
when setting a local forward (in this case to extension 23) on a SIP
phone, I see the following on the Asterisk CLI :
[Aug 31 14:59:34] -- Got SIP response 302 "Moved Temporarily" back
from 11.22.33.44:40670
[Aug 31 14:59:34] -- Now forwarding
Local/myaccount184 at CallFromQueue-000007f4;2 to 'Local/23 at from-internal'
(thanks to SIP/myaccount184-00003729)