simulate what will occur when AMI performs an
Originate action by using the 'channel originate' CLI command.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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:
ChanVariable(SIP/1234-0001): fu_callerid=foobar
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Matthew Jordan
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Check us out at: http://digium.com http://asterisk.org
. It appears as if you're running a modified version of Asterisk, in
which case all bets are off. This works fine on the Linux build agents,
which is what we use to build the tarballs on downloads.asterisk.org.
So, no, I don't think there's a bug in the shell script.
Matt
--
Matthew Jordan
Digium, Inc
:-)
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Matthew Jordan
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?
Instead of doing an explicit comparison, you could always use a regular
expression to perform the matching:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_REGEX
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us
. A pcap should show the changes in SSRC and
might illustrate what's occurring.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
.
Of course, the Asterisk open source developer community could take this
work on, which would be a welcome addition to Asterisk 12.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
the behaviour changed?
Just so I'm clear on the scenario, what are the channel technologies
involved? Is the transfer initiated via a protocol message or via a DTMF
feature?
Thanks,
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check
. Most
likely, the hangup handler has been attached to one half of the Local
channel as opposed to the channel you want it attached to. Can you
include the full dialplan that you're using?
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
resolution of 'FQDNz' probably took 3
seconds. You may want to consider a local DNS cache to help speed up
results.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
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Check us out at: http://digium.com http://asterisk.org
to it.
Thanks!
[1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
[2] https://issues.asterisk.org/jira
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Digium, Inc. | Engineering Manager
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probably use the stock Debian installation as a guide.
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to do dialplan shenanigans.
Matt
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://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_ConfbridgeTalking
Hope this helps,
Matt
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the unregistered peers to REGISTER in a device agnostic fashion?
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Matthew Jordan
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Check us out at: http://digium.com http://asterisk.org
On 01/29/2013 02:52 AM, Ishfaq Malik wrote:
On Wed, 2013-01-16 at 08:06 -0600, Matthew Jordan wrote:
On 01/16/2013 05:31 AM, Ishfaq Malik wrote:
On Thu, 2012-01-12 at 11:51 +, Ishfaq Malik wrote:
Hi Everyone
This issue has reared it's ugly head again for us. If a call comes
On 01/26/2013 07:26 AM, Andreas Sikkema wrote:
On 1/18/13 13:24 , Matthew Jordan wrote:
1) Contact your carrier and ask why they are rejecting the 200 OK.
2) Assuming they won't change their behaviour, find out what they want
in a response that declines an image media format. Without knowing
if you're using Asterisk 11 and res_xmpp.
res_jabber: yup, totally still a problem.
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Matthew Jordan
Digium, Inc. | Engineering Manager
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/digium02-0003
CallerIDNum: 657-5309
CallerIDName: digium01
ConnectedLineNum: unknown
ConnectedLineName: unknown
UniqueID: Asterisk-01-1359052866.2
DestUniqueID: Asterisk-01-1359052866.3
Dialstring: digium02
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL
.
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the exact same SDP in various
re-INVITE requests and ignored the negotiated offer isn't a good sign
that Asterisk would be able to do much to appease it.
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Matthew Jordan
Digium, Inc. | Engineering Manager
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be initialized. This was fixed in r374763
(Asterisk 10) under issue ASTERISK-20455. It was included in the 10.11.0
release.
Matt
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Check us out at: http://digium.com http://asterisk.org
so that it answers
back with whatever they told you they want. This will occur in
chan_sip's add_sdp method - in particular, look for the portion where
add_sdp adds *either* the m_modem/a_modem strings to the SDP *or* the
pre-formatted decline_m_line.
Matt
--
Matthew Jordan
Digium, Inc
. Asterisk 11 gets every fix for every bug that originates in
the 1.8 branch.
See the bug fix section of this page for more information on how the
merge process works:
https://wiki.asterisk.org/wiki/display/AST/Software+Configuration+Management+Policies
Matt
--
Matthew Jordan
Digium, Inc
, is the SDP answer you pasted the entire
SDP that Asterisk 11 responds with? Specifically, are there no format
attributes for the image stream in the SDP that Asterisk responds with?
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
of the inbound channel when it goes into the Queue?
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
by enabling reference count debugging in Asterisk
[1]. Note that this isn't available in menuselect, as it has to be
enabled in the specific modules you whose objects you want to track.
[1] https://wiki.asterisk.org/wiki/display/AST/Reference+Count+Debugging
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Matthew Jordan
Digium, Inc
?
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Is this on purpose, a fault on my side, or is this a bug?
No, that should work. What's the output of 'logger show channels'?
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
the behaviour may not be entirely desirable, this isn't so much a
bug as a limitation of the system.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
to Asterisk 11.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
Thanks you for your continued support of Asterisk!
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
the time your issue is in Triage:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
If you are experiencing a crash, you may also want to read up on how to
get a backtrace:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
Thanks!
--
Matthew Jordan
Digium, Inc
://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
Thanks!
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
for all
future versions. I don't believe either is true in this situation.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
have purchased phones from Digium - call
technical support. They should be able to help you resolve this issue.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
On 10/12/2012 10:35 AM, Christopher Harrington wrote:
On Fri, Oct 12, 2012 at 9:59 AM, Matthew Jordan mjor...@digium.com wrote:
Is type=peer strictly necessary? I don't know how they're currently
being specified from users.conf, is that possible to specify in
users.conf? I was under
branch. You can either test with a checkout from
1.8, or - since this was fixed in 1.8.18.0-rc1 - test with the release
candidate.
If you find that you still have this problem after that, please let us
know and we'll reopen ASTERISK-20337.
Thanks!
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Matthew Jordan
Digium, Inc. | Engineering
. The script
responsible has been sternly reprimanded.
Sorry for any confusion!
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
the channel in extension 700?
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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)
same = n,Hangup()
include = parkedcalls
include = catch_all
[catch_all]
exten = _X.,1,Noop(22 - Running in ${CONTEXT} at ${EXTEN})
exten = _X.,n,Set(AGISIGHUP=no)
exten = _X.,n,AGI(/var/lib/asterisk/hash3/bin/exten2.py)
exten = _X.,n,HangUp
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Matthew Jordan
Digium, Inc. | Engineering Manager
445
can't overstate how much I agree with this. A configuration option to
'tweak' the behavior in pbx.c is much more likely to introduce problems than
solve them. If a clear consensus cannot be reached, I'd err on the side
of doing nothing than put in yet another configuration option.
--
Matthew
application (ConfBridge) that, in performance tests, performed much better
than MeetMe. A big limitation of MeetMe is its reliance on DAHDI for
mixing. ConfBridge removed this limitation, and typically can mix more
participants at a faster rate.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan
something here, but - unless you have a business case
that requires some functionality MeetMe provides - it sounds as if Page
would be more appropriate than MeetMe. Its certainly a lighter weight
approach to sending audio from a single source to multiple devices.
--
Matthew Jordan
Digium, Inc
that
started with '*', we prevented the latter scenario.
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Matthew Jordan
Digium, Inc. | Engineering Manager
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globally. If you execute
sip show settings, what is the value of the Q.850 Reason header?
If you enable 'sip set debug on', what is the actual CANCEL request
sent to the UA?
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
they
would be failing on your machine.)
The Asterisk Test Suite is a tool to aid in Asterisk development and test.
If you don't feel comfortable debugging problems in Asterisk, then it might
not be the tool for you.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW
of order DTMF transmission.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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, as well as maintain
the handling of out of order packets. We certainly try to not have a new
version of Asterisk break any existing devices, even if those devices are
behaving oddly.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
a pcap of the RTP stream and a DEBUG log with RTP debug
enabled, using 'rtp set debug on'.
Thanks,
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
:11 PM, Matthew Jordan wrote:
- Original Message -
From: Vladimir Mikhelson v...@mikhelson.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, September 14, 2012 9:24:41 PM
Subject: Re: [asterisk-users] DTMF digits falsely
with
the official RPMs.
No, that is not a known bug. Please open an issue in the issue tracker.
https://issues.asterisk.org/jira
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
the TEST_FRAMEWORK flag should resolve that problem.
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Matthew Jordan
Digium, Inc. | Engineering Manager
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will not cause inbound XMPP processing to
resume working until you completely restart Asterisk.
Thank you,
Noah Engelberth
MetaLINK Technologies
Yeah, that doesn't sound right at all.
Do you mind filing a bug in the issue tracker and attaching a DEBUG log?
Thanks!
--
Matthew Jordan
Digium
a finger towards the
culprit.
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Matthew Jordan
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Check us out at: http://digium.com http://asterisk.org
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sorry you don't care for JIRA,
but its the system we use now and its highly unlikely that we will migrate
away from it anytime soon.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
to stream your sound from a source other than MOH, using a
Local channel may be appropriate. I'm not sure how a Queue would help here.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
- Original Message -
From: Markus unive...@truemetal.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: Matthew Jordan mjor...@digium.com
Sent: Monday, August 27, 2012 12:48:53 PM
Subject: Re: [asterisk-users] One leg in a conference
- Original Message -
From: Markus unive...@truemetal.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: Matthew Jordan mjor...@digium.com
Sent: Monday, August 27, 2012 1:55:08 PM
Subject: Re: [asterisk-users] One leg in a conference
when you have more than 10 priorities
in an extension seems sub-optimal.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
mechanisms available for third parties to get at
that information.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
- Original Message -
From: Matthew Jordan mjor...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, August 23, 2012 4:37:53 PM
Subject: Re: [asterisk-users] GotoIf redirection to label not working
correctly
of authorization
is required. The 403 response matches what would be sent if the
username was valid but an invalid password/hash was provided. This
response should be sent regardless if the username was actually
valid.
Based on your provided SIP traffic, that appears to be what happened.
--
Matthew
on a call (that
don't allow multiple calls), when another call comes in, the hint will flash,
but cannot be picked up, as there isn't a ringing extension.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http
in asterisk 10. Looks like I need to move to
app_confbridge.
ConfBridge is the preferred conference application in Asterisk 10+. While
MeetMe is currently deprecated, you can still enable it and run it in
Asterisk 10+.
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW
is based on, that sounds like a great idea - but it would
be a non trivial task. We'd certainly welcome assistance from the community
in performing that migration.
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Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http
If the result here is no, then the ODBC library installed on the system
Asterisk is running on was not detected to have support for Unicode types.
You can either update your database schema to use the non-Unicode column
types, or you could explore updating the ODBC library on your system.
--
Matthew Jordan
- Original Message -
From: Mike Diehl mdi...@dominion.diehlnet.com
To: asterisk-users@lists.digium.com
Cc: Matthew Jordan mjor...@digium.com
Sent: Thursday, August 9, 2012 12:08:01 PM
Subject: Re: [asterisk-users] No CDR after upgrade (1.6.x - 10.2.1)
On Thursday 09 August 2012 7
://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
Note that Macro is deprecated in more recent versions of Asterisk, and the 'b'
option will only be available in Asterisk 11.
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out
- Original Message -
From: Dmitry Melekhov d...@belkam.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, July 29, 2012 10:44:42 PM
Subject: Re: [asterisk-users] Video conferencing?
27.07.2012 19:25, Matthew Jordan пишет
-2578;2'
Looking for s in default (domain 127.0.0.1)
So: in your particular dialplan, you should be able to escape
the semicolon and rerun. You should see a SIP MESSAGE request
sent to SIP provider.
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806
- Original Message -
From: Mike Diehl mdi...@dominion.diehlnet.com
To: asterisk-users@lists.digium.com
Cc: Matthew Jordan mjor...@digium.com
Sent: Sunday, July 29, 2012 3:39:05 PM
Subject: Re: [asterisk-users] No audio playing back voicemail from odbc
On Saturday 28 July 2012 6:45
the
audio, which is probably not what you want.
File storage is the only mechanism to have video voicemail (with audio)
at this time.
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
For older versions of Asterisk:
https://wiki.asterisk.org/wiki/display/AST/Old+Calling+using+Google
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
an idea)
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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- Original Message -
From: Tiago Vasconcelos tiago.o.vasconce...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Saturday, July 28, 2012 1:12:57 PM
Subject: Re: [asterisk-users] How to send a SIP MESSAGE outside a call
On 28-07-2012 18:23, Matthew Jordan wrote:
Thank you so
- Original Message -
From: Support mdi...@diehlnet.com
To: asterisk-users@lists.digium.com
Cc: Matthew Jordan mjor...@digium.com
Sent: Saturday, July 28, 2012 2:38:09 PM
Subject: Re: [asterisk-users] No audio playing back voicemail from odbc
CLI core show translation paths slin
/2012/astridevcon.aspx
Barring that, if this is a feature you would love to have, then you can
either write it and submit the contribution to Asterisk, or you could
work with developers in the Open Source community to write this feature.
I hope this clarifies the refusal.
Thanks!
--
Matthew Jordan
that you may need to re-compile Asterisk with the appropriate
build options to get a usable backtrace.
Once you have a backtrace, please file an issue on the issue tracker.
https://issues.asterisk.org/jira/
Thanks!
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville
is documented
here:
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
Does anybody have any experience in making this work?
Thank you!
Alex
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com
Would I be better off asking this question of the dev community?
Nope, as this isn't an Asterisk development question.
Thanks
Ish
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
what to look for?
sean
You'll need to provide a backtrace using the instructions below:
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
As soon as you have the information, please open an issue in JIRA.
Thanks!
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis
for their hard work and effort,
but its still freely available if you choose not to do so)
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
media)
; on outgoing calls to a peer. Calls will fail with
HANGUPCAUSE=58 if
; the peer does not support SRTP. Defaults to no.
In the DEVICE CONFIGURATION section, encryption is explicitly listed as a
supported setting for devices.
--
Matthew Jordan
Digium, Inc
or the setting field in the freepbx
that can resolve this (the voice mail message to be maximum
for 30 or 40 second, after that to hangup even without
pressing #). From where?
Regards
Bilal
--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806
seems to be helping.
Since you have a number of different servers running different versions
of Asterisk, can you provide which server in your scenario is running
which version?
Thank you,
Noah Engelberth
MetaLINK Technologies
--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan
Doug:
You may want to apply the patch on ASTERISK-19923 - it fixes a critical
problem in app_voicemail in the latest version.
We are planning on releasing a new version of 1.8.13/10.5, which
will include this patch.
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW
on
asterisk 10.x.x - We need to be able to do FoIP (Fax over IP) as we
have no pstn lines available.
Do you know how to setup a reliable fax system, then we will pay you
to help us do this.
If you're looking for consultants, you may want to try the asterisk-biz
mailing list.
--
Matthew Jordan
Digium
be used with the
SRTP transport specified, e.g., RTP/SAVP or RTP/SAVPF.
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
offer, it will attempt to process it.
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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that
communicate with that Asterisk instance, the media_address setting could
potentially be used to specify an IPv6 address to send media to, while
keeping the signalling on IPv4.
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http
is greatly appreciated.
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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no longer
passes through Asterisk.
Kind regards,
Jonas.
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Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
will contain a Channel:
header with that value.
--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
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, Matthew Jordan mjor...@digium.com
wrote:
In your particular case, if I were writing a system that wanted to
associate
a created channel with an Originate Action, after I issue the
Originate,
I'd listen for a NewChannel event. If that NewChannel event
specified
a
channel
attempt to do several days a week, but it is not always possible to
do it every day.
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
, that the INVITE request will
be rejected.
I imagine that this is the case, as ASTERISK-19601 noted that
when this situation occurs, the NOTICE message indicates that
there is a failure to match the extension, as opposed to a failure
to match an allowed domain.
Matthew Jordan
Digium, Inc
- Original Message -
From: Yaroslav Panych panyc...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 17, 2012 6:56:17 PM
Subject: Re: [asterisk-users] Incoming SIP call is rejected always.
2012/4/18 Matthew
Niccolo:
I've reopened the issue and placed some comments on the issue requesting
more information. In the future, if you need an issue reopened, you can
contact a bug marshal in #asterisk-bugs.
Thanks,
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL
field value as it would normally
when sending an updated request.
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
- Original Message -
From: Benoit Panizzon benoit.paniz
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