Re: [asterisk-users] switching from simple_bridge technology to native_rtp issue

stream to inactive (when placed on ; hold) or to T.38, it will still be done, regardless of this ; setting. Note that direct T.38 is not supported. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW

Re: [asterisk-users] Asterisk crashes when reloading configs...

to start looking for the problem? Please get a backtrace illustrating the problem: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace Once you have a properly generated backtrace, open an issue on issues.asterisk.org. Thanks - Matt -- Matthew Jordan Digium, Inc. | Engineering Manager

Re: [asterisk-users] PJSIP question

://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_CHANNEL -- 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

Re: [asterisk-users] Hold

lifetime. -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] CDR custom variable on second call leg - via originate or .call file

are you talking about? What value do you want where? Keep in mind that unless all channels are answered, they won't show up in your CDRs (unless you have unanswered=yes set in cdr.conf). -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check

Re: [asterisk-users] WSS over Asterisk

ASTERISK-21930) that was fixed in Asterisk 11.9.0: http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-11.9.0-summary.html Which version of Asterisk are you using? Is it 11.9.0 or later? -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] WSS over Asterisk

-22961 -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] SSL/TLS weakness impact on Asterisk authentication

as a transport. If your OpenSSL version is one of those affected by the various vulnerabilities, then yes, you are at risk. This also applies to all other modules in Asterisk that use TLS, including AMI, the HTTP server, and others. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445

Re: [asterisk-users] 'restart when convenient'

stop now' command. You can, however, stop the 'core stop gracefully' by issuing 'core abort shutdown', which will cause Asterisk to stop the existing shutdown attempt and return to normal processing. You can then issue another shutdown command at your leisure. -- Matthew Jordan Digium, Inc

Re: [asterisk-users] 'restart when convenient'

On Wed, May 28, 2014 at 1:08 PM, Matthew Jordan mjor...@digium.com wrote: On Wed, May 28, 2014 at 12:47 PM, Doug Lytle supp...@drdos.info wrote: Perhaps i should join the -dev list to find out what 'convenient' actually means for the process... The dev list is for discussions of coding

Re: [asterisk-users] transmit_silence not properly recognized on 1.8 ?

that is preventing silence from kicking off. -- 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 -- _ -- Bandwidth

Re: [asterisk-users] BLF and notifyringing in Asterisk 11

if it was set on a peer by peer basis - that would be a useful improvement. -- 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

Re: [asterisk-users] Putting a notice in the logs from the dialplan

', 'WARNING', 'NOTICE', 'DEBUG', 'VERBOSE' or 'DTMF'. message Output text message. [See Also] Not available Ha! Just when you think you've found every corner of Asterisk, you turn around and there's something else. Just goes to show, you learn something new every day. -- Matthew

Re: [asterisk-users] SQLite3 astdb back-end

+Function_DB_KEYS * https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_DBdeltree Whether or not you store 'contact information' (and that could have a variety of meanings, so I won't interpret it specifically) is up to you. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan

Re: [asterisk-users] CBAnn channel not going away in Asterisk 12

occurring here. Thanks - 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 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Channel names

of it. Is it unwise to use channel names to extract the peers involved in a call? How a channel is named is a function of the channel technology. Which channel technology(ies) are you curious about? Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] Putting a notice in the logs from the dialplan

not. It would be a relatively trivial addition to add a dialplan application that could emit an Asterisk logging message at any one of the various levels, if someone were interested. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?

On Wed, Apr 30, 2014 at 8:37 AM, Administrator TOOTAI ad...@tootai.netwrote: Le 30/04/2014 15:19, Matthew Jordan a écrit : On Wed, Apr 30, 2014 at 8:13 AM, Administrator TOOTAI ad...@tootai.netmailto: ad...@tootai.net wrote: Please, people from Digium, Matt again closed the new bug

Re: [asterisk-users] CBAnn channel not going away in Asterisk 12

, make/make install, and re-run the scenario that reproduces the result. A refs file will be created in your Asterisk log directory - attach that to the issue along with DEBUG log. Thanks! -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us

Re: [asterisk-users] Reload problems with 1.8.27 and 11.9.0 - Someone else ?

, the original issue will get re-opened. -- 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 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] CBAnn channel not going away in Asterisk 12

log and the standard Asterisk DEBUG log showing the problem, that would help a lot in finding out what is going on. -- 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

Re: [asterisk-users] IAX2 trunk on IPV6

help with this issue. IAX2 does not support IPv6 in that version of Asterisk. IPv6 support was added to chan_iax2 in Asterisk 12 [1]. [1] https://wiki.asterisk.org/wiki/display/AST/New+in+12 -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

Re: [asterisk-users] Does CalDAV require neon-0.29 , not 0.30?

version is assumed to be incompatible with every other minor version. {quote} http://www.webdav.org/neon/doc/html/refvers.html You should either downgrade to 0.29, or else have a community developer determine if res_calendar_ews is compatible with later versions of neon. Matt -- Matthew Jordan

Re: [asterisk-users] Problem building Asterisk-12.2.0

-prereq script wasn't good enough. What distro are you building on? I'm running both Ubuntu 12.04 and CentOS 6.5 locally. Both have the libraries listed in install_prereq. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http

Re: [asterisk-users] Alembic - Asterisk 11

is far more useful. 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 -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] how to configure callcentric peer

square brackets [name] ; 2. Asterisk checks the From: address and matches the list of devices ; with a type=peer ; 3. Asterisk checks the IP address (and port number) that the INVITE ; was sent from and matches against any devices with type=peer -- Matthew Jordan

Re: [asterisk-users] AMI and pyst

/asterisk/starpy) extensively in the Asterisk Test Suite. It does lock you into using twisted (https://twistedmatrix.com/trac/) - which has both pros and cons - but it may be a viable alternative for you if pyst doesn't work out. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan

Re: [asterisk-users] Asterisk CLI Banner

the modified source in any fashion) and/or Section 2c. Unless you really know what you're doing with regards to software licensing, I would highly suggest not modifying the welcome message. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

Re: [asterisk-users] Asterisk 12 - CDR changes

plenty of ways to manipulate CDRs through the dialplan. A specification for CDR behaviour in Asterisk 12 is available on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW

Re: [asterisk-users] Wrong patch 1.8.26.1 at http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.26.1-patch.gz ?

older version, it would have been difficult to apply to 1.8.26.0. I've updated the patch on the downloads site such that it is now a patch against 1.8.26.0. Let me know if you have any other issues. Thanks - Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW

Re: [asterisk-users] Any help Address 0xfffffffe out of bounds in app_confbridge.casterisk-11.5.1 using confbridge.conf

the author of that code and ask them to fix the crash. 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

Re: [asterisk-users] PJSIP - Using multiple AOR contacts when dialing through an endpoint

of an AOR. -- 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 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

me DTMF!) (2) Accept the INVITE request but not have DTMF over RFC 4733. What you're seeing is option (2), which I think is better than rejecting the entire call simply because the thing you are talking to doesn't support the DTMF mode you configured it to have. -- Matthew Jordan Digium, Inc

Re: [asterisk-users] PJSIP - dtmf mode is not updated when the far end doesn't support rfc2833

that is analogous to the chan_sip 'auto' setting - what you configure for you endpoint today is what it will use. That's not a bug, just something not existing yet. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http

Re: [asterisk-users] Add SIPCALLID of egress leg to CDR

the MASTER_CHANNEL function to reach back to the parent channel and set the CDR variable there. 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

Re: [asterisk-users] h extension isn't processed after call file finishes.

want hangup logic and you're using Asterisk 11+, you could also use a hangup handler on the outbound channel. But otherwise, I would expect that the 'h' extension would always be fired for a channel executing dialplan, so long as it is in the same context. -- Matthew Jordan Digium, Inc

Re: [asterisk-users] Retaining P-Asserted Info

On Mon, Feb 17, 2014 at 4:29 AM, Nick Cameo sym...@gmail.com wrote: Hello Ishfaq, I just tried it and it did create a P-Asserted header however it contains the extension of the asterisk peer not what was passed by our switch. From the previous example: P-Asserted-Identity: 222

Re: [asterisk-users] Parking in Asterisk 12.0.0

as it knows that you have not yet safely left the bridge you are in. We'll take a look and see if there's a way to allow this to happen again. For now, you should use the one touch parking feature. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] how to get full channel name - AMI cuts off [solved]

On Thu, Jan 30, 2014 at 5:48 PM, Justin Killen jkil...@allamericanasphalt.com wrote: After posting this, I ran across 'core channel show concise', which gives the data in a more machine friendly format. That may work over AMI, but in general, it isn't recommended. The command class

Re: [asterisk-users] grp_lock error when compiling against pjproject

providing them usually leads to more confusion, not less. [1] https://wiki.asterisk.org/wiki/display/AST/Installing+pjproject#Installingpjproject-IssuesandWorkarounds Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] [HELP]: Auto-answering calls placed from call files

as well [1]. [1] http://lists.digium.com/pipermail/asterisk-users/2008-January/204375.html 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

Re: [asterisk-users] grp_lock error when compiling against pjproject

? -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] IOPS required by Asterisk for Call Recording

attempts to record the audio on the thread servicing the channel(s). 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

Re: [asterisk-users] CDR and Transfer, an asterisk scaring bug lasting from 1.4 version...

they should work; the behaviour of CDRs in Asterisk 12 and in future versions should be substantially more predictable. Matt [1] https://wiki.asterisk.org/wiki/display/AST/New+in+12 [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification -- Matthew Jordan Digium, Inc

Re: [asterisk-users] AMI eventmask question

to either to have that attribute displayed. Matt [1] https://github.com/asterisk/publish-docs -- 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

[asterisk-users] Asterisk Community Code of Conduct

project - and the Asterisk community! -- 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 -- _ -- Bandwidth

Re: [asterisk-users] asterisk 11.7.0: Delayed audio

=yes. Consider changing this value ; if rtp packets are dropped from one or both ends after a call is ; connected. This option is set to 4 by default. ; probation=8 Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http

Re: [asterisk-users] Failed to get 160 samples from read factory , asterisk-11.5.1 app_confbridge.c

that you've written with no context and asking someone to debug the problems you're seeing is unlikely to generate the help you want. Thanks - Matt [1] http://lists.digium.com/mailman/listinfo/asterisk-dev [2] http://lists.digium.com/pipermail/asterisk-dev/2014-January/064504.html -- Matthew

[asterisk-users] Asterisk 10 EOL Notice

in Asterisk 10. So, to everyone who helped make Asterisk 10 successful, thank you! Matt [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http

Re: [asterisk-users] A Question about Management/Control Protocol Licensing

/pipermail/asterisk-app-dev/2013-October/000127.html 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 -- _ -- Bandwidth

Re: [asterisk-users] Not able to get remote channel variables containing RTCP values

-- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Voicemail greeting playback issues?

configure the probationary period as well as whether or not strict RTP checking is enabled in rtp.conf. 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

[asterisk-users] Asterisk 10 EOL Approaching

. For more information on Asterisk versions and their supported lifetimes, please see the following wiki page: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Thank you for your continued support of Asterisk! -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW

Re: [asterisk-users] Asterisk 1.8.20 crashing

be useful in analyzing how the system got into that state. -- 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

Re: [asterisk-users] [asterisk-dev] how determine mandatory modules to slimming asterisk

that, you should look at what applications and functions a module provides to determine if you need it. Asterisk: The Definitive Guide has some excellent information in Chapter 2. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] warnign

. Alexandr Anikin is the maintainer of chan_ooh323 [1]; he may be able to correct the issue for you. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Open+Source+Maintainers -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out

Re: [asterisk-users] warnign

specific hangup cause code is completely up to the channel driver. Not all channel drivers support it; if someone wanted to add that functionality to chan_ooh323 that'd be great; but it's completely different than the condition that the OP is seeing. Matt -- Matthew Jordan Digium, Inc. | Engineering

Re: [asterisk-users] Asterisk AMI 1.3 Specification

://wiki.asterisk.org/wiki/display/AST/New+in+10#Newin10-AsteriskManagerInterface * https://wiki.asterisk.org/wiki/display/AST/New+in+11 -- 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

Re: [asterisk-users] Asterisk-12 issue after successful installation

://svn.asterisk.org/svn/asterisk/branches/12/contrib/scripts/install_prereq 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

Re: [asterisk-users] Launched new Asterisk Cilck 2 Call for websites

. If verification is required, please request a hard-copy version. Please don't print this e-mail unless you really need to. FYI: This is a silly disclaimer when sent to a public mailing list. (And yes, I don't feel I should have received this.) Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445

Re: [asterisk-users] Capture Media IP in CDR (CDR)

]. [1] http://svn.asterisk.org/svn/asterisk/branches/11/COPYING [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties 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-users] New mailing list - asterisk-app-dev

audience may be available to assist you. Thanks - and we all look forward to lots of productive discussions on the new mailing list about building applications that use Asterisk as their communications engine! Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW

Re: [asterisk-users] Capture Media IP in CDR

to wonder, what is prompting this request? 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 -- _ -- Bandwidth

Re: [asterisk-users] PJSIP question

for review, as it will give you more bang for the buck. You can download the patch from here: https://reviewboard.asterisk.org/r/2846/diff/raw/ Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http

Re: [asterisk-users] How to customize CDR(src) value ?

- my-src or something like that - you can add a new value to your CDR record by using the CDR function, i.e., Set(CDR(my-src)=+34123456789). Certain CDR backends - such as cdr_custom or cdr_adpative_odbc - have the ability to store custom values. Matt -- Matthew Jordan Digium, Inc. | Engineering

Re: [asterisk-users] DPMA for Asterisk 12?

: undefined symbol: __ao2_container_alloc Quite a lot, actually. Beyond just linking issues, there's that whole new SIP stack thing we'd like to get it using. We're working on it - stay tuned... Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL

Re: [asterisk-users] Asterisk crash issue

crashes. If this happens again, please obtain a backtrace using the instructions and file an issue on the Asterisk issue tracker [2]. [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira Thanks Matt -- Matthew Jordan Digium, Inc. | Engineering

Re: [asterisk-users] Asterisk 12.0.0-alpha1 Now Available!

be somewhat klunky and difficult to manage - particularly for complex bridging scenarios. Hope that helps! 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

Re: [asterisk-users] How to reply with 480 Call-limit to incoming SIP call ?

+Cause+Mappings -- 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 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Am I being hacked?

in Asterisk 10. (Just in case someone is still running that version...) -- 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

Re: [asterisk-users] Dial application b subroutine arguments not passing?

: https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers And: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial -- 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

Re: [asterisk-users] AMI timeouts

Originate actions. 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 -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] CEL custom variable in outbound channel

SIP channel when the Local channel performs a Dial to the actual SIP device. [1] https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers 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-users] Recommended in Asterisk Wiki E-Mail

more e-mails from the Asterisk Wiki unless you explicitly choose to watch a page. Sorry for the spam! 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

Re: [asterisk-users] AMI timeouts

, a synchronous Originate can block that session from receiving events until it completes. 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

Re: [asterisk-users] Upgrade from 10.2.4 to 11.4.x, the Reader's Digest version?

in there. -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] problem with dtmf detection in asterisk 11

information to be able to tell what is going on. RTP debug for the endpoint in question would most likely help. 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

Re: [asterisk-users] FTP server down?

into it ASAP. 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 -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Calls drop after transfer

at a trace of the SIP messages during the transfer using 'sip set debug on peer' (set it for both the transferer as well as the transfer destination). That should show why the requests are rejected and/or why a call is hungup. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis

Re: [asterisk-users] Asterisk crash

are always bugs. Thanks! 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 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to new

say.conf config file, you don't have an extension that matches datetime. You have one that matches date and time, but not the combination of the two. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http

Re: [asterisk-users] Endpoint call forwarding

responding with? A pastebin of a log showing DEBUG and higher level messages when a call forward attempt occurs would help. Thanks 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

Re: [asterisk-users] Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to new

On Mon, Jul 1, 2013 at 6:24 AM, Amit Patkar | ATPL a...@avhan.com wrote: Hi I am using following say.conf file. Its a default file, which comes with Asterisk installation. When I call SAY DATETIME AGI function, it simply returns without playing date time. Where as if I use mode=old

Re: [asterisk-users] Send event/notification from one channel driver ot another

to notify a channel that something has occurred, it queues a control frame on that channel. Control frames include things like media source changes/media updates, indications that signalling actions should take place, etc. What specific use case are you looking at? Matt -- Matthew Jordan Digium, Inc

Re: [asterisk-users] CEL logging and queue APP_START/END, maybe an issue?

efforts that went on in 12. Masquerades are now an implementation detail, so in the future, you won't have to deal with BRIDGE_UPDATE. 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

Re: [asterisk-users] Asterisk-11 loop behaviour

on startup (or periodically). I did a quick Google search and found out that this particular context is used by FreePBX 2.9's Time Conditions feature - see http://www.freepbx.org/forum/freepbx-distro/distro-discussion-help/odd-failed-calls-in-logs for more information. -- Matthew Jordan Digium, Inc

Re: [asterisk-users] Asterisk-11 loop behaviour

On Tue, Jun 25, 2013 at 12:18 PM, James B. Byrne byrn...@harte-lyne.cawrote: On Tue, June 25, 2013 09:57, Matthew Jordan wrote: On Mon, Jun 24, 2013 at 2:37 PM, James B. Byrne byrn...@harte-lyne.cawrote: It is not an infinite loop but it does go on for an inordinately long time. Does

Re: [asterisk-users] Questions about sRTP

the same function. An example of doing this is on the wiki: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics -- 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

Re: [asterisk-users] Problem with CEL logging and channel bridging

). The system you build on top of CEL has to understand the semantics of Local channels and tie the two together. 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

Re: [asterisk-users] MOH don't work after update

? Add-on modules are in the addons subdirectory. Typically, these modules are not built and installed by default, and have to be enabled in menuselect. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com

Re: [asterisk-users] Codec Negotiation problem

it disables video for the call between the two endpoints. 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

Re: [asterisk-users] MeetMe exit status?

of granularity. The closest available is the MEETMESECS channel variable, which tells you how many seconds the participant was in the conference. You can find a full list on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/MeetMe+Channel+Variables Matt -- Matthew Jordan Digium, Inc

Re: [asterisk-users] Confbridge doesn't kick chan_local

channel that is messing up the reference counting inside the ConfBridge. Otherwise, it's an error in ConfBridge. 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

Re: [asterisk-users] Is uniqueid/sequence a safe CDR table primary key ?

) is linkedid (which should be enabled and added to your schema), uniqueid, and sequence number, with the asterisk system name specified. -- 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

Re: [asterisk-users] Confbridge doesn't kick chan_local

On 06/03/2013 01:03 PM, Chris Gentle wrote: On Mon, Jun 3, 2013 at 11:52 AM, Matthew Jordan mjor...@digium.com wrote: snip If both (1) and (2) are successful, than there's some impact that the Ices application is having on the Local channel that is messing up the reference counting inside

Re: [asterisk-users] Most suitable version for Production ENV

for a production environment ? The Asterisk wiki describes the various versions of Asterisk: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http

Re: [asterisk-users] Executing a dynamic sequence of applications

-- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] asterisk-gui-2.1.0-rc1

/AST/Asterisk+Versions 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 -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Asterisk 1.8-cert and AGC

libspeex and its dependencies installed func_speex won't be available. You can verify whether or not you have all of the dependencies for func_speex in menuselect. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http

Re: [asterisk-users] Asterisk 11.3 and Kamailio 4.0 Realtime Integration Tutorial

-- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

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