.
So the ability to use DPMA with Asterisk RT is very important for our
large deployments.
Anyone willing to contribute towards a bounty for this feature?
--
Robert Broyles
On 5/7/15 7:14 AM, Matthew Jordan wrote:
On Fri, May 1, 2015 at 10:43 AM, Robert Broyles
rob...@webservicesaz.com
We love our Digium phones and DPMA - but we really need it to work on
our Realtime Platform. Otherwise we lose all the cool features and they
are just standard SIP phones.
Anyone working on a solution for this? Or anyone from Digium see this on
the roadmap?
--
I'm trying to solve a problem I have with agents hanging up on callers
before they even talk to them (caused by agents dropping their handset
or something.)
What I want is something like AgentLogin() where the agent has to press
'1' to accept the call. Does anyone know how to get this to work
Sebastian Milioto wrote:
Hi all,
I have to install 25 IP Phone in some building. I want just basic IP
Phones like:
Cisco-Linksys SPA922 u$s 146
Grandstream GXP-2000 u$s 105
Snom 300 u$s 119
The most valuables parameters for me are (in importance
Leif Neland wrote:
- Original Message -
*From:* Zhang Shukun mailto:bit...@gmail.com
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
mailto:asterisk-users@lists.digium.com
*Sent:* Friday, January 15, 2010 11:48 AM
*Subject:* [asterisk-users]
Zhang Shukun wrote:
2010/1/15 Leif Neland le...@neland.dk:
- Original Message -
From: Zhang Shukun
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, January 15, 2010 11:48 AM
Subject: [asterisk-users] Realtime queue not work
hi, all
i try to confiture
Hi,
So I'm using Asterisk Realtime Queues and Queue members on 1.4.28.
I've noticed if there are no people in the queue when a call enters,
even after a queue member enters, the call is never rang to him.
From the debug, it seems that Asterisk is only grabbing the queue
member list upon
Robert Broyles wrote:
Hi,
So I'm using Asterisk Realtime Queues and Queue members on 1.4.28.
I've noticed if there are no people in the queue when a call enters,
even after a queue member enters, the call is never rang to him.
From the debug, it seems that Asterisk is only grabbing
Or you could setup a VPS environment (perhaps openvz) and run Asterisk
in a virtual environment.
I've done this in the past and it works well.
ram wrote:
On Wed, Dec 30, 2009 at 5:29 PM, Saeed Akhtar
saeedakhtar@gmail.com mailto:saeedakhtar@gmail.com wrote:
hi all,
I
So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the
database structure for cdr_mysql is:
CREATE TABLE cdr (
calldate datetime NOT NULL default '-00-00 00:00:00',
clid varchar(80) NOT NULL default '',
src varchar(80) NOT NULL default '',
dst varchar(80) NOT NULL default
Tilghman Lesher wrote:
On Wednesday 30 December 2009 10:52:48 Robert Broyles wrote:
So I'm noticing from the docs/ on Asterisk Addons 1.4.10 that the
database structure for cdr_mysql is:
CREATE TABLE cdr (
calldate datetime NOT NULL default '-00-00 00:00:00',
clid varchar(80
Can anyone point me to a working how to on state_interface?
I found this little example:
http://www.freepbx.org/v2/ticket/3496
But it didn't work with the backport on v1.4.x, so I have v1.6.2 installed
on my test box, and it's still not working.
Are there any special settings that need to be
Any takers?
Still trying to get this resolved...
Thanks!
Robert Broyles wrote:
It's my understanding that the backport is available now in 1.4.
However, seem to be having some issues with it. Just wondering if I
have everything setup right.
I'm running 1.4.26.2 realtime.
queue_members
It's my understanding that the backport is available now in 1.4.
However, seem to be having some issues with it. Just wondering if I have
everything setup right.
I'm running 1.4.26.2 realtime.
queue_members:
`uniqueid` int(10) unsigned NOT NULL auto_increment,
`membername` varchar(40)
to themselves.
The logs also show asterisk bridging the incoming SIP connection with
the Phone, but when the agent hangs up, it shows the agent as hanging
up, but it's not closing the bridge.
Any ideas? Where should I begin in troubleshooting this? Thanks!
--
Regards,
Robert Broyles
Team Lead
So I'm guessing, I would disable any wrapup on the queue, and then in my
'h' extension pause the agent for a set period of time, with another
extension to unpause the agent if entered?
Or is there a better way to set the pause after the call is over?
Thanks!
--
Regards,
Robert Broyles
that?
--
Regards,
Robert Broyles
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Great backports! :-)
This should really be merged into 1.4.
--
Regards,
Robert Broyles
Atis Lezdins wrote:
Well, i can share mine backports of queue_log into mysql for 1.4.
Basically you need two backports (that's why there are numerous
files). Realtime store/destroy allows Asterisk
Problem is, without going to 1.6, I can't get the queue log or events
posted to MySQL in realtime.
There used to be a patch out there for queue_log, but it doesn't work
with versions 1.4.21 or higher.
--
Regards,
Robert Broyles
Anthony Francis wrote:
Robert Broyles wrote:
I saw some
The patch I was referring to is:
http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queue_logging
It doesn't work for the current SVN 1.4
--
Regards,
Robert Broyles
Anthony Francis wrote:
Yeah, I need to make a new patch for 1.6 to go to it myself. I wrote a
patch way back
as I can - hoping the bug would 'resolve itself' -
but now I'm putting a bounty out on it.
http://bugs.digium.com/view.php?id=13691
--
Regards,
Robert Broyles
DISCLAIMER : This email and any files transmitted with it are property of
Poornam Info Vision Pvt. Ltd. This email contains
Yes, I've already posted notes on the bug.
I applied the patch, and when attempting to recompile, it fails.
--
Regards,
Robert Broyles
Team Lead - Customer Support Rep
Poornam Inc aka Bobcares
Phoenix, Arizona, USA 602.288.9145
Jason Parker wrote:
Robert Broyles wrote:
I saw some
By the way, I'm more than happy to send murf a case of rootbeer (or real
beer assuming he's legal :-P ) if this bug and/or related bugs can be
resolved soon. :-)
--
Regards,
Robert Broyles
Team Lead - Customer Support Rep
Poornam Inc aka Bobcares
Phoenix, Arizona, USA 602.288.9145
Jason
Actually, that's alcohol abuse. :-)
Regards,
Robert Broyles
Christian Victor wrote:
2009/3/4 Atis Lezdins a...@iq-labs.net mailto:a...@iq-labs.net
Bottle of Riga Black Balsam (45%), just have to figure out a way
to send it :)
Balsam??? By mail? Doesn't that count as liquid
Yea, that patch was tried, and doesn't resolve the issue either.
I will hold out on the bounty a little longer... maybe it will be
resolved soon. It's pretty important for us.
--
Regards,
Robert Broyles
Jason Parker wrote:
Tilghman Lesher wrote:
On Wednesday 04 March 2009 10:24:16
. In the meantime, I need to be able to
capture unanswered calls in the previously semi working method.
--
Regards,
Robert Broyles
Steve Murphy wrote:
On Mon, 2009-01-05 at 12:27 -0700, Robert Broyles wrote:
On 12/17/08 I updated to 1.4.22 from 1.4.21...
Now the CDR data isn't recording calls where
FYI to everyone...
It was an issue on Vitelity's end on the gateway I was assigned to. They
switched me, and it's working fine now.
--
Regards,
Robert Broyles
Brent Davidson wrote:
Robert Broyles wrote:
I turned on DTMF debugging. It looks like the extra digits coming in
are less than
in once.
When testing the dialplan internally, it accepts only the digits that I
key in.
Anyone else experienced this?
--
Regards,
Robert Broyles
DISCLAIMER : This email and any files transmitted with it are property of
Poornam Info Vision Pvt. Ltd. This email contains confidential
Btw, I'm using Asterisk SVN-branch-1.4-r178640
Robert Broyles wrote:
So I'm using the READ() application within an IVR, and having a strange
issue, and wondering if anyone else has had this problem.
When calling from an outside line, and entering the digits during the
read() part of my
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert Broyles wrote:
So I'm using the READ() application within an IVR, and having a strange
issue, and wondering if anyone else has
Not at all.
In fact, I found that relaxdtmf=yes is now available for sip.conf as of
1.4 as well.
However, that didn't resolve the problem.
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking
Yea, I tried that too. I have it: dtmfmode=rfc2833
--
Regards,
Robert Broyles
Brent Davidson wrote:
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric Wieling, Asteria Solutions Group wrote:
Robert
.
Can someone else check this on their system, and see if this is a problem?
--
Regards,
Robert Broyles
Brent Davidson wrote:
Robert Broyles wrote:
Okay. I'm using this all over SIP Trunking with Vitelity.
Any other suggestions?
--
Regards,
Robert Broyles
Eric Wieling, Asteria
Glad I could help!! :-D
Leonja Cerebro wrote:
To Robert Broyles,
Thank you very much, it is very helpful information.
Regards,
Leonid
2009/2/18 Robert Broyles rob...@poornam.com mailto:rob...@poornam.com
Hi,
You might want to check out this tutorial:
http://hostseries.com
Hi,
You might want to check out this tutorial:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/
It's a good place to start.
--
Regards,
Robert Broyles
Leonja Cerebro wrote:
Hi,
Sorry, I'm a newbee in Asterisk, and I want to call from one SIP trunk
of Asterisk B (registered
Check out this alternative:
http://hostseries.com/agentcallbacklogin-alternative/
Regards,
Robert Broyles
oumar ndiaye wrote:
Hi,
My queue used to work fine until I upgraded to 1.6. I am getting the
message:
No application 'AgentCallBackLogin' for extension (default, 31001, 1)
After
Why don't you use followme if you want to do that?
In fact, you can have followme, plus the local agents as mentioned in
the previous alternative that I mentioned.
--
Regards,
Robert Broyles
Anthony Francis wrote:
Robert Broyles wrote:
Check out this alternative:
http://hostseries.com
Hmm, this is all very interesting.
Looks like using a Macro and the 'M' Dial() option is the way to go for
now if you need the answer confirmation.
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Look at example #2, and adapt it for your needs.
--
Regards,
Robert Broyles
Philipp Kempgen
You guys... grr...
I'm still on 1.4 svn ... not ready to even think about 1.6 OR 1.8(when
it's released) for production right now. :-)
--
Regards,
Robert Broyles
Rob Hillis wrote:
...except that Macros are now deprecated and will most likely be removed
in 1.8.
Robert Broyles wrote:
Hmm
I think we'd be better off posting a regular FAQ, perhaps weekly, with some of
these suggestions, as well as providing a link to that FAQ from the mailing
list signup page, along with a STRONG suggestion to peruse the FAQ first.
I agree with this 100%
I'm still pretty new to the mailing
to read,
they ask stupid questions because they're too lazy do to the footwork.
Robert Broyles wrote:
I think we'd be better off posting a regular FAQ, perhaps weekly, with some of
these suggestions, as well as providing a link to that FAQ from the mailing
list signup page, along with a STRONG
Jared Smith wrote:
On Tue, 2009-01-27 at 10:13 -0700, Robert Broyles wrote:
I'm still pretty new to the mailing lists myself. I don't consider
myself a novice Asterisk user, but one of my biggest 'complaints' is
the lack of a well documented FAQ or Manual for Asterisk
Anyone know how soon this will be patched?
Or are we waiting on the new CDR structure/method?
Steve Murphy wrote:
On Mon, 2009-01-05 at 12:27 -0700, Robert Broyles wrote:
On 12/17/08 I updated to 1.4.22 from 1.4.21...
Now the CDR data isn't recording calls where the caller hung up while
. 942's do, for an extra $20.
Regards,
Robert Broyles
Julian Lyndon-Smith wrote:
Can anyone who has used both comment on the pros and cons ? Need to buy
about 30 of these, for a small company with limited IT support.
Julian
Are you planning on connecting your two Asterisk servers with SIP or IAX?
Check out this tutorial if using SIP:
http://hostseries.com/connecting-to-asterisk-servers-via-sip/
You should be able to adapt it to your needs. Good luck!
Paul wrote:
Can anyone tell me how I can completely move an
The Blackberry community has been begging for a SIP client for awhile.
Apparently there are some restrictions within the Blackberry OS. But
with the newer Blackberry models including wifi abilities, we should be
seeing something released soon... I hope! **Fingers Crossed**
Eric Moniz wrote:
On 12/17/08 I updated to 1.4.22 from 1.4.21...
Now the CDR data isn't recording calls where the caller hung up while
waiting on the Queue.
Sample CDR data BEFORE the upgrade:
2008-10-30 12:46:47;\John\
If you don't want to use the AEL, but want an easy way to have agents
login and out, check out this quick tutorial:
http://hostseries.com/agentcallbacklogin-alternative/
Ariel Dorfman wrote:
i have done some research, but there says that i can use a function called
AgentCallbackLogin, but
:27 -0700, Robert Broyles wrote:
On 12/17/08 I updated to 1.4.22 from 1.4.21...
Now the CDR data isn't recording calls where the caller hung up while
waiting on the Queue.
Sample CDR data BEFORE the upgrade:
2008-10-30 12:46:47;\John\
0006741103;0006741103;11621708182;incoming;SIP/carrier
With regards to storing queue_log data in mysql, it depends on the
Asterisk service your running.
1.6.x check out the following:
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
1.2.x OR 1.4.x check out the following patch/solution:
this.
Thanks for your time.
--
Regards,
Robert Broyles
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Would this set the periodic-announce filename just for this call?
Thanks!
--
Regards,
Robert Broyles
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Hmm,
exten = s,1,Playback(/home/Sounds/greeting)
exten = s,n,Set(PERIODIC_ANNOUNCE=/home/Sounds/queue2)
exten = s,n,Queue(CSR)
It's not working. It just plays the default announcement.
Same goes for:
exten = s,n,Set(GLOBAL(PERIODIC_ANNOUNCE)=/home/Sounds/queue2)
Btw, I'm using v1.4.22
That just plays back my announcement file before the caller enters the
queue.
It's still playing the default file once in the queue.
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Thanks. The thread points to an issue with the periodic-announce not
playing if the queue is set to ring, instead of musiconhold.
I have musiconhold with my queue.
My sample queue for testing purposes:
[CSR]
musiconhold = classic
retry = 1
strategy = ringall
joinempty = yes
Okay thank you.
This is something that I'm trying to avoid. I want to have one single
Queue, but based on the incoming DID, have a different periodic-announce
file played.
It would be awesome to be able to set all of the queue settings from the
dialplan, if so wished:
examples of what I mean:
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