onduct a
post-call survey (Jason N)
3. Re: Second Asterisk server SIP JOIN a call to conduct a
post-call survey (Joshua C. Colp)
--
Message: 1
Date: Mon, 01 Jul 2019 11:15:01 -0300
From: "Jos
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Second Asterisk server SIP JOIN a call to conduct
a post-call survey
On Sun, Jun 30, 2019, at 11:09 AM, Jason N wrote:
> I am designing a solution for a hotel booking call center with the
> following (mandatory) design:
And, how would [S] know that [H] has disconnected? (Is there an Asterisk
event that indicates one party has disconnected from a multi-party call)
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On
Behalf Of Jason N
Sent: Sunday, June 30, 2019 10:08 AM
To: 'Asterisk Users
I am designing a solution for a hotel booking call center with the following
(mandatory) design: After the call from the customer with the booking agent
is complete (and the Hotel PBX disconnects from the call), a second PBX
takes over to conduct a survey of how the call went. Both PBX's are
ask the customer to try that command
on the CLI. Your output is exactly what I expect (and what I see on other
systems)
The real mystery here is why is the AMI on this system responding strangely?!
Permissions? Corruption? Some asterisk config file setting I should look at?
Jason
on this system?I
confirmed the AMI connection has full read/write permissions. Why is the
call data missing from the response?
Jason
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e router(s)
between your source and destination it requires different handling.
Good luck!
Jason
On Mon, Mar 18, 2019 at 9:20 AM Jerry Geis wrote:
> I was trying to find if Asterisk supports Dante ?
>
> Dante -- https://www.audinate.com/
> AES67 -- http://www.aes.org/publications/standar
Objet : Re: [asterisk-users] Asterisk 13 attended transfer alcatel
On Fri, Jun 9, 2017, at 04:59 AM, Jason TOMLINSON wrote:
> Hello,
>
> Since upgrading from asterisk 11 to asterisk 13 (I have tested up to
> the latest 13.16.0 release), we have a problem with attended transfers
> to
flags that need setting, etc?
Thanks
Jason
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That's not something that is likely to be supported. Any configure
script in the tree will be run via the top-level build process, as
needed. Is there some reason you think you need to run the other
configure scripts yourself?
On 03/05/2014 08:54 AM, Gianluca Merlo wrote:
Hello everyone,
I
? Is there an
application that needs to pass the DTMF from the SIP user in sip.conf to the
conference application?
Thanks in advance,
Jason
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New
The packages currently do not support SRTP.
On 06/03/2013 10:56 AM, Daniel Pocock wrote:
I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.org
I followed this guide:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages
The SRTP support appears to be missing
On 06/03/2013 12:03 PM, Daniel Pocock wrote:
I tried building manually from the source RPM
Before running rpmbuild, I installed libsrtp-devel and I notice that
res_srtp.so is generated during the build
However, the rpmbuild fails for other reasons (see the other email I
sent to the list
On 05/21/2013 10:19 AM, Ahmed Munir wrote:
Hi,
Last year, I installed Asterisk 10.4.2 and enabled logrotate on daily
basis which was working perfect. Now in couple of months back, the
logrotate feature is not working at all but simply appending the logs
in 'messages' file. Listing down down
On 05/07/2013 05:13 AM, Olivier wrote:
2013/5/7 Matthew Jordan mjor...@digium.com mailto:mjor...@digium.com
2. 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
On 01/03/2013 02:23 PM, Markus Weiler wrote:
Am 03.01.2013 21:21, schrieb Nick Khamis:
Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008
do you mean 1_000_8 ?
Markus
I think he means 10007.
--
On 10/03/2012 10:46 AM, Eric Wieling wrote:
A port is not a door if there is nothing listening on the port.
Open ports are not a security issue. Stuff running on open ports are.
Do you have some external software listening on those ports when there isn't an
active call? Asterisk isn't
On 08/28/2012 10:04 AM, Andrew Latham wrote:
On Tue, Aug 28, 2012 at 11:00 AM, Johan Wilfer li...@jttech.se wrote:
2012-08-28 16:44, Andrew Latham skrev:
Try this to test with
http://www.digium.com/en/products/ivr/audio-converter.php and compare
your output first...
Interesting. Didn't
On 08/28/2012 10:32 AM, Danny Nicholas wrote:
Does the .c program compile stand-alone or as an add-on?
g++ check_sounds.c
check_sounds.c: In function âint main(int, char**)â:
check_sounds.c:152: error: invalid conversion from âvoid*â to âdirent**â
check_sounds.c:154: error: invalid conversion
On 06/12/2012 02:56 PM, Danny Dias wrote:
Hi,
I'm just trying to install the DPMA on my Asterisk. I already made the upgrade
from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did:
/mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185
/
*compiling Asterisk-Cert2
On 06/05/2012 10:23 AM, Chet W. Stevens wrote:
During testing with the Digium phones I have run into a problem where I try to
make a change to the sip device name. I make the device name change in
sip.conf
then make the matching change to the lines in res_digium_phone.conf. I then do
'sip
On 05/22/2012 04:54 PM, Danny Dias wrote:
There are 4 files for each voicemail:
msg.gsm
msg.txt
msg.wav
msg.WAV
That is perfectly normal. The .txt file is metadata that contains things like
caller ID and duration. Asterisk will also save voicemails into every format
you
On 04/27/2012 01:39 PM, Don Kelly wrote:
What flavor are flashphoner minties?
--Don
Dailing flavored. What else?
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On 03/06/2012 12:31 PM, Ron Bergin wrote:
Mathew,
Each of those odbc modules are unavailable i.e., marked with XXX
I even deleted the asterisk build directory and started over, but had the
same results.
What prereqs do I need besides these:
mysql.i386
On 03/06/2012 03:44 PM, Karl Fife wrote:
It's not a question of whether the default directory permissions are
appropriate. I agree with those.
What we're talking about here is what happens during updates to an existing
directory. I can't see any rationale for changing the group permissions.
On 03/06/2012 04:24 PM, Patrick Lists wrote:
On 06-03-12 23:07, Karl Fife wrote:
Yep. That's what's happening. I'll file a bug.
AFAICT it's not a bug but the way RPM works.
Regards,
Patrick
He didn't suggest that he was talking about RPMs. If that's the case, then I
take back
On 03/05/2012 06:34 AM, Eric Germann wrote:
Does anyone have an idea on when 1.8.9.3 might show up in the RPM
repositories?
Thanks!
EKG
They should be available now.
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On 03/05/2012 01:49 PM, Eric Germann wrote:
Will a 1.8.10.0 build be imminent or should we go ahead and push this in to
production with testing?
Thanks!
EKG
~20 minutes
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On 03/05/2012 06:00 PM, Lefteris Zafiris wrote:
Some packages seem to lag behind, eg asterisk18-addons-mysql is compiled
against 1.8.7:
asterisk18-addons-core-1.8.7.0-2_centos5
asterisk18-addons-mysql-1.8.7.0-2_centos5
Is this a problem with the repo? Are these packages
obsolete/unmaintained
On 03/05/2012 06:22 PM, Karl Fife wrote:
I notice that the installation of Asterisk 1.8.8 thru 1.8.10 (probably
earlier versions too) remove the group write permissions from
/etc/asterisk/. which is different than 1.4. And 1.6.
Is this expected behavior?
If so, what's the rationale?
If not,
On 02/26/2012 06:22 PM, Patrick Lists wrote:
On 25-02-12 19:47, Jason Parker wrote:
yum and rpm do not support downgrades.
Incorrect. There is yum downgrade. See man yum.
yum downgrade is extremely broken. It fails, often, potentially leaving a
system in an unrecoverable state
yum and rpm do not support downgrades. You can try using `yum shell` to
uninstall one version and install another version in one transaction, but you'll
have to go it alone.
On 02/25/2012 11:49 AM, Ast Coder wrote:
Thanks Jason.
One more question: Is there anyway to go back on an Asterisk
On 02/23/2012 10:09 AM, Ast Coder wrote:
Hi,
I have followed instruction
on
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisites
to
add Digium Asterisk repositories but doing a, yum search asterisk only shows
me Asterisk 1.4, 1.6, and 1.8. There is
On 02/21/2012 02:55 PM, Bryant Zimmerman wrote:
Ok I now have the basics for dynamic parking working but for some reason when
a
caller calls in and is parked with a transfer the return call dials the sip
peer
of the caller and not hte peer of the last party that parked the call. Anyone
On 02/21/2012 05:34 PM, Stephen Brown wrote:
application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000
/var/lib/asterisk/sounds/music/Rolling In The Deep.mp3
Probably unrelated to your issue, but you're going to want to quote that
filename.
--
I'm reading some information that recommends using SER / OpenSER for
large installation to offload SIP traffic from the Asterisk server.
http://www.voip-info.org/wiki/view/Asterisk+at+large
However, it looks like the information might be dated.
I'm looking at a potential 750 SIP phone and 150
Thanks for the info. As we move forward, we'll be testing and making a
phone selections. No doubt we'll run into this. Are you saying if the
phone is stated to be a 10/100 phone, it still may not work at 10?
On 2/13/2012 1:32 AM, Benny Amorsen wrote:
Jason W. Parksjason.w.pa...@gmail.com
for the response. Jason
Cheer up, the worst is yet to come.
On 2/13/2012 2:48 AM, Hans Witvliet wrote:
On Mon, 2012-02-13 at 09:32 +0100, Benny Amorsen wrote:
Jason W. Parksjason.w.pa...@gmail.com writes:
I can move my voice infrastructure to an IP-based one running 10Mbps,
utilize
for the example. That's good information.
On 2/13/12, Bryant Zimmerman brya...@zktech.com wrote:
Jason
A standard SIP VOIP phone will use less than 100k per voice call. For
example I have several bussiness customers that have a dedicated DSL line
and they do up to 6 lines very well
, 2012 at 7:13 AM, Vieri rentor...@yahoo.com
/mc/compose?to=rentor...@yahoo.com wrote:
--- On Wed, 2/8/12, Jason W. Parks jason.w.pa...@gmail.com
/mc/compose?to=jason.w.pa...@gmail.com wrote:
From everything I've researched to
date, my understanding is most
On 01/30/2012 11:06 AM, Eric Germann wrote:
We mirror off http://packages.asterisk.org to a staging server, then update
from there.
When will this show up on packages.asterisk.org?
Thanks!
EKG
The RPMs should be there in a few minutes.
--
In my neck of the woods...
A Cordless Phone refers to a cordless handset with a wired base. The
phone communicates with the base and can't work without it. It's usually
proprietary in nature as well.
A Wireless Phone usually refers to any phone communicating via 802.11.
No base required. A
On 12/28/2011 03:10 PM, Danny Nicholas wrote:
Can somebody point me to an explanation from Kevin or Tzafir or someone else
up the food chain explaining the differences/benefits of 1.6/1.8 vs
1.4/10.0?
Every branch (1.0, 1.2, 1.4, 1.6.0, 1.6.1, 1.6.2, 1.8, 10) of Asterisk contains
new
On 12/12/2011 09:26 AM, Danny Nicholas wrote:
I'm wondering if the bind 161 as root statement is a mis-statement or
if not, maybe somebody like Tzafir can explain why since none of the
other Asterisk binds require root access (this message is still in
10.0-rc3).
This is accurate. Non-root
On 11/15/2011 09:58 AM, Tony Mountifield wrote:
I see on my CentOS systems that certain users for particular subsystems
have standardised UIDs and GIDs. For example mysql=27, ntp=38, sshd=74.
My two questions are:
1. Is there a list of these standard assignments somewhere? Googling did
On 11/15/2011 10:42 AM, Tony Mountifield wrote:
Yes, I was hoping to use such a system user and group for asterisk, which
would not conflict with any other system package I might install in the
future, by virtue of being reserved for asterisk.
There shouldn't be any conflict either way.
Does anyone have a patch for 1.4.42 to enable shared_lastcall?
I've seen patches for 1.4.19 and 1.4.24.1 (http://goo.gl/WL6Fx).
Thanks,
Jason
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On 10/20/2011 05:16 PM, Paul Belanger wrote:
Greetings,
If you are planning on attending Astricon, please take the time to
attend the GPG key signing event. More information can be found on
the wiki page[1].
[1]
On 10/18/2011 09:52 PM, Luke Hamburg wrote:
I think this might actually be a bug.
https://issues.asterisk.org/jira/browse/ASTERISK-18137
It is indeed a bug, but it's not the bug you referenced. This issue
only exists in 1.8.8.0-rc1. It has been fixed for 1.8.8.0-rc2 which
will be released
On 10/17/2011 02:22 PM, Ioan Indreias wrote:
Hello,
Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM
version from Asterisk repo I found that asterisknow-version is needed
by package asterisk18-core-1.8.7.0-2
How could this be explained?
Best regards,
Ioan
The
On 09/30/2011 09:53 AM, Tony Mountifield wrote:
In article 4e85d19f.4090...@digium.com,
Kevin P. Fleming kpflem...@digium.com wrote:
This is why the output was changed to microseconds from milliseconds; in
the older version, the lowest number that should be shown was 1
millisecond, even if
On 07/19/2011 01:02 PM, Michael wrote:
On Tue, Jul 19, 2011 at 3:34 PM, Kevin P. Fleming kpflem...@digium.com
mailto:kpflem...@digium.com wrote:
You don't need to install asterisk-addons to be able to store CDRs; you need
them to be able to store CDRs in MySQL specifically. If you
That's not the password.
I switched it to that in the config file for realism.
I always give some honey out to those who have a sugar tooth.
Any ideas on the fix?
On Mon, Jun 27, 2011 at 3:24 AM, Matt Darnell mattdarn...@gmail.com wrote:
When i reload asterisk, calendar show calendars
I am trying to integrate Asterisk 1.8.4.2 with Google iCal and I have been
unsuccessful.
libical-0.44.tar.gz - installed
neon-0.29.5.tar.gz - installed
i did a make clean, make make install in asterisk.
make menuselct
[*] res_calendar
[*] res_calendar_caldav
[*] res_calendar_ews
[*]
On 05/17/2011 07:18 AM, Stefan Gofferje wrote:
On 04/17/2011 02:13 AM, Stefan Gofferje wrote:
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
I finally figured it out.
For facebook chat to work you have to
On 05/16/2011 08:36 AM, Jerry Geis wrote:
I an running centos 5. I added this to the digium.repo file in /etc/yum.repos.d
directory.
[digium-current]
name=CentOS-$releasever - Digium - Current
baseurl=http://packages.digium.com/centos/$releasever/current/$basearch/
enabled=1
gpgcheck=0
On 05/12/2011 02:46 PM, Jason Parker wrote:
I'll make it a point to respond to this email when the new builds are available.
These builds are now available.
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On 05/12/2011 02:40 PM, Cassius Smith wrote:
Hi all
Usually I build asterisk from source, but recently have been doing a
couple of test installations with packages from the Digium repository.
About how long does it take to get from new release announcement into the
Digium RPM repository?
I know most billing software sell this as a monthly service. You get cd-rom
every month where they have collected the published tarrif tables filed with
the FCC. You load it on the software to analyze call costs. I'm guessing this
is a lot of labor hours/manual work thus they charge for
On 05/06/2011 01:30 PM, Bob Beers wrote:
Not sure if this will work, but I'd try adding, before line 86:
#Workaround for PAE
%if %{paevar} == PAE
Provides: kmod-dahdi-linux
%endif
Can't actually test it myself, sorry.
- Bob
You'd probably want to modify the kmodtool that comes with it, to
as needed, ect.
Thanks,
Jason
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On 03/15/2011 12:34 PM, Fellipe Paes wrote:
why I can't use _. in my dialplan?
Because it matches everything. In this case, it's matching the 'h' exten. So
when the call gets hung up, it goes to _. and does what you're seeing.
--
On 02/23/2011 12:43 PM, vip killa wrote:
I recognize all the options given yet as I explained before they are not viable.
I do not have the resources to pay someone, I do not have the expertise to fix
this issue because according to an asterisk developer any fix in that area
would be deeply
On 02/02/2011 02:14 PM, Frank Liu wrote:
Hi there,
Per the instruction from http://www.asterisk.org/downloads/yum , I
setup the yum repository on my Centos 5 x86_64 machine and did a
yum install asterisk18 asterisk18-configs
then I startup the asterisk (with no changes to config) just to see
On 01/19/2011 12:18 AM, randulo wrote:
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
That would fall under Quirk's Exception: Intentionally invoking Godwin's Law to
attempt to kill a thread is rarely successful.
On 01/19/2011 04:41 AM, Ishfaq Malik wrote:
Hi
Does anyone have any idea how long it will take for the new release of
asterisk 1.8 to make it to the digium yum repositories?
Thanks
Ish
They've been there since yesterday afternoon. It's possible that you hit the
repository before the
On 12/20/2010 11:35 AM, Daniel Tryba wrote:
I was wondering why *...@default should match '_*[0-9a-zA-Z].*0.'
in 1.6.13. Who is making the parse error, * or me?
CLI dialplan show *...@default
'_*[0-9a-zA-Z].*0.' =
1. NoOp(${EXTEN}) [pbx_config]
2.
On 12/02/2010 02:03 PM, Danny Nicholas wrote:
Hi gang,
We are moving our computers from a cluster of physical machines to a VMWARE
server with virtual machines. We investigated and are looking to replace our
TDM400P/TDM410P with AEX410P cards. Can we run asterisk with the DAHDI drivers
from
Those boxes run around $50k USD, I've only seen them once back in the late
1990s.
At work for customer consulting we have very expensive site licenses for
Prognosis IPT Assessor which generate great looking pdf reports.
We also use Cisco IOS IP SLA however it doesn't have a reporting
Hi Guys,
Hope fully somebody out there will have experienced this and can shed some
light on how it was overcome.
Current setup includes asterisk 1.6.2.11, GNU GK and a Quintum Tenor CMS on
the same lan. Earlier I was unable to make a sip call from the CMS back to a
sip client registered on my
Hi,
I thought you'd cracked it, I simply turned off all sip by removing the
sip.conf
but after a few more days it did the same.
I've set logging permanently on again.
Any other suggestions?
Cheers,
Jason.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I'm not aware of an open source speech product.
Some great examples where speech recognition works well are 1-800-USA-RAIL,
Microsoft/Cisco corporate directory 425-882-8080 you can say the employees name
and be connected and those works great, 1-800-Goog-411 also works well.
Windows 7
what to do, except go back to
the old 1.4 server.
Cheers,
Jason.
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does.
Cheers,
Jason.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]on Behalf Of A J Stiles
Sent: 17 August 2010 10:36
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
WireShark does a good job showing the T38 communication. Most products you can
also set packet redundancy to send 2 packets.
Your setup was T.38 ATA to T.38 Gateway with PRI/ANALOG/PSTN/G.711. I've heard
various problems with SIP/PSTN and faxing, around jitter/packet loss and it's
not
On 07/28/2010 11:32 AM, Tilghman Lesher wrote:
They permit what packets will even reach user2
It should also be pointed out that the config option is permit, and not
allow.
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I normally work with other 3rd party IVRs, usually once the Agent is Reserved
we signal the phone system to play Music on Hold while it's ringing the Agent.
The trick here is to replace the Music on Hold with a fake ring file.
-Original Message-
From:
On 07/19/2010 01:23 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mattias
Sent: Monday, July 19, 2010 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voice
On 07/15/2010 08:16 AM, Vasiliy G Tolstov wrote:
Hello.
Who can add asterisk16-xmpp module to packages.asterisk.org or build
asterisk with support xmpp and update packages?
Thank You.
This is something we've been considering for a while. It should make its way
onto the list shortly.
--
I think you need to ask your SIP provider about Redirecting Header, ask what
they support and how-to.
I work more with Cisco CallManager and SIP Rediversion Header is new in
CallManager 8x. Not sure about Asterisk. We have this same problem with Cisco
Mobility/Single Number Reach, providers
On 05/26/2010 08:00 PM, cov...@ccs.covici.com wrote:
From another thread, I blacklisted netjet and now things are working.
But I wonder what is going on here and where did netjet come from -- it
doesn't look like an dahdi module to me.
It comes from mISDN. It is a very badly misbehaving
On 05/12/2010 01:03 PM, Robert Wagner wrote:
Hi,
when i include a sip configuration from another file in my sip.conf
using #include /etc/asterisk/sip-sipgate.conf everything seems to be
working.
The peer is listed when i execute sip show peers and Status is OK.
But the peer is not listed
not registered
I am not stuck with realtime, I just have a mysql database with info
that changes and needs to update the dialplan accordingly.
Jason
Confidentiality Statement Notice: This email is covered by the
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intended only
/queues_conf.html
The best part is no recording will be initiated while the people are
listening to music on hold
Jason
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Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and
intended only for the use of the individual
Michael Nausch wrote:
HI,
I tried to install asterisk and mISDN via
http://www.asterisk.org/downloads/yum
My machine is running with kernel-2.6.18-164.15.1.el5.i686
Packages for that kernel version were missing. That was an oversight and has
been corrected. A `yum update` should be
Olivier wrote:
Hi,
Between 1.6.1.9 and 1.6.1.18, handling of menuselect has changed in such
a way that I cannot script non-english sound files downloading anymore.
The following used to work (unattended) with 1.6.1.9 (for instance):
cd /usr/src/asterisk-${ASTERISK_VERSION}
./configure
I am getting a bunch of Primary D-Channel on span 1 up but there was not
a down message before that.
Is this normal?
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intended only for the use of the individual or
Pablo Ruiz wrote:
Hello,
Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary
packages at packages.asterisk.org http://packages.asterisk.org?
Greets.
Packages for 1.6.2 will be available Real Soon Now. It's near the top of my
short list.
They exist, and are sitting
bruce bruce wrote:
Thanks for the update Jason,
How do the upgrades work if v1.6.0 is already install and one wants to
upgrade to 1.6.2 (once it's available)?
yum upgrade asterisk*
???
Thanks
It should be as easy as a `yum update`. That's the goal, anyways
It seems that my realtime is not assigning channel variables correctly.
INFO
Asterisk 1.6.0.26
Exten.conf
exten = _X.,1,NoOp()
exten = _X.,2,Set(DEVICE=${CUT(CHANNEL,,1)})
exten = _X.,3,Set(NULL=${REALTIME(agents,device,${DEVICE})})
exten = _X.,4,NoOp(DEVICE is ${DEVICE})
exten =
I'm trying to test a Diaglogic BrookTrout SR140 card. It uses H.323.
Trying to find a way I could use my laptop to send a fax over H323 to the
BrookTrout card for testing. Any thoughts? Normally I'd setup a FXS interface
on a Cisco router and setup a h323 dial peer to the BrookTrout, but I
Brian J. Murrell wrote:
I wonder if Asterisk's skinny/sccp channel driver could be used as a
client to register with a Cisco PBX. That is, along with a SIP
client, say, have Asterisk and said SIP client stand in for a Cisco
phone, or an IP Communicator.
Anyone done this?
Cheers,
b.
I'm experiencing runaway ringing too, can we make this a class action
against someone?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
Brower
Sent: Wednesday, March 10, 2010 10:20 PM
To: Chris Owen
Cc:
could do a tcpdump on port 5038 and try to fish out
the bad username or password but I wasn't able to see any passwords or
usernames in plain text.
Any way I could maybe change the logging in Asterisk to show me the
username that is not able to authenticate?
- Jason
Jay Vocaire wrote:
Thanks for researching this for me. If you actually look at the link
you sent me, you will see that the latest is:
asterisk16-core-1.6.0.21-1_centos5.x86_64.rpm 20-Jan-2010 15:45 11M
So, we come back to my original question: is there a reason for the
delay on getting
stephen.hindma...@bt.com wrote:
rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec
snip
error: Failed build dependencies:
kernel-devel = 2.6.18-164.11.1.el5 is needed by
dahdi-linux-kmod-2.2.1-1_centos5.2.6.18_164.11.1.el5.i386
Add a --target=i686 to your rpmbuild
Brian wrote:
Each time the server is rebooted Asterisk duly
deletes the manually created /var/run/asterisk directory - quite why it
does this I just don't know - perhaps it is a bug?
Your assumption is incorrect. Some Linux distributions will empty /var/run/ on
boot, just as they do with
/DAHDI chunk size and that directly affects system load.
Second question - the document explains how to change the chunk size for
Sangoma hardware. Is there a general way to do that for DAHDI?
Thanks is advance!
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
Rochester, NY 14624
Office
Doug Lytle wrote:
Dave Fullerton wrote:
Note num and not number I don't know if that was a change from 1.4
to 1.6 or if Doug mistyped it.
Not a mistype. I've been using number all along, but looking at the
docs shows that I've been incorrect. It must concatenate the number
down to
We use 3Com managed gigabit switches that support QoS and priority for
VoIP.
3Com Unified Gigabit Wireless PoE Switch 24
and
3Com Baseline Switch 2924-PWR Plus
Jason Baker
IT
Coordinator
Glastender, Inc.
5400 North Michigan Road
Saginaw,
Michigan 48604 USA
Phone: 989.752.4275 ext.
228
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