Hello All,
I've been doing some looking into VMX Locator(part of FreePBX from what I
see). One of my sales guys came from a company that was running FreePBX and
we are running straight asterisk installed using custom built RPM's.
Currently in the voicemail app the only key press that does
magic
myself. Thanks Ryan.
On Thu, Jun 23, 2011 at 10:06 AM, Ryan Wagoner rswago...@gmail.com wrote:
On Thu, Jun 23, 2011 at 7:45 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Hello All,
I've been doing some looking into VMX Locator(part of FreePBX from what I
see). One of my sales
be
done, but the vendor is telling them it can't. Thought I'd ask around here
and see if anyone has done it? Thanks.
Cheers,
Darrin Henshaw
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New
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something similar.*
*
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*Cheers,*
*
*
*Darrin Henshaw*
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Hello All,
I posted a bug on the 14th of this month, and haven't heard anything
back. However, I've since discovered that the problem is not in
chan_iax.c as I originally thought, it's actually app_mixmonitor.c.
Basically when I use 1.4.26.2 with an ilbc codec between two asterisk
servers trunked
My first though is using the isnull function.
http://www.voip-info.org/wiki/view/Asterisk+func+isnull
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
I'm basically trying to make an argument optional in a macro, I'm
starting to think it's not possible
If I do this in my macro
exten =
Mind posting the macro itself? I think we might need to store the
return value of isnull then test with execif.
On 16/10/2009, Ishfaq Malik i...@pack-net.co.uk wrote:
That fails to execute in both conditions
ABBAS SHAKEEL wrote:
Please try this
xten = s,2,ExecIf( 0EXISTS(${ARG3})=1
Something like:
exten = s,1,ExecIf(${${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Should work from what I read on voip-info.org.
On Fri, Oct 16, 2009 at 10:19 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Mind posting the macro itself? I think we might need to store the
return value of isnull
Actually just noticed a typo try:
exten = s,1,ExecIf($[${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Had { instead of [ in the ExecIf.
On Fri, Oct 16, 2009 at 10:26 AM, Darrin Henshaw
darrin.aster...@gmail.com wrote:
Something like:
exten = s,1,ExecIf(${${ISNULL(${ARG3})} = 1]|Set,ARG3=1)
Should
First suggestion is if this Asterisk server is accessible from the
internet put a secret in the peer definition. What you have now is
wide open. Second thing is if I understand it you are going:
PC(Soft Phone) ADSL Router Internet Asterisk box. Is that
correct? If not, can you descibe it
You could validate whether it has a physical connection I believe. Add
qualify=yes in the sip definition and use something like:
/usr/sbin/asterisk -rx sip show peer | grep UNREACHABLE | wc -l
Where is the name of the sip definition on your system. If the
return is 0 then all is well,
similar to audiohook_inherit, which we use to
allow mixmonitor to continue recording the call after it's been
transferred. I've looked around, but haven't found anything. Thanks.
Cheers,
Darrin Henshaw
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Hello,
The call center I manage previously had almost all calls entering a
single queue. In order to differentiate the calls to the techs we set
the callerid name based on the caller id number offered to us.
Basically, it was a gosubif the callerid number matches this, and in
the sub we set the
Bah, my mistake, as Steve said the entry goes in zapata.conf.
On 09/07/2009, Steve Totaro stot...@asteriskhelpdesk.com wrote:
resetinterval=never in zapata.conf.
you may want to reset them though, just not as frequently. The
resetinterval can take an integer as well.
Thanks,
Steve Totaro
I am under the impression that MixMonitor records both streams and
mixes them at the same time, meaning I'm not recording on the caller
or callee but both. However, I could be mistaken. Thanks.
On Tue, Jul 7, 2009 at 7:08 PM, Miguel Molina mmol...@millenium.com.co wrote:
Darrin Henshaw escribió
add resetinterval=never in your zaptel.conf, or chan_dahdi.conf
depending on what you are running. zaptel or dahdi.
On Wed, Jul 8, 2009 at 10:35 AM, Aman Dhallyaman.dha...@live.com wrote:
Hi All,
Hope you all are fine and good, Today i have found that Mine all PRI
Channels are restating
fine in 1.2, however, switching to 1.4 seems to have
introduced this into our environment. Thank you for any assistance you can
provide.
Cheers,
Darrin Henshaw
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asterisk-users
Check out http://www.voip-info.org/wiki/view/Asterisk+iax+qualify.
I've ran into problems with home routers not keeping the connection alive, udp
timeouts most likely. These options particularly, the qualifyfreqnotok will
have asterisk send out a poke to the soft phone if it reports the phone
. If it doesn't exist it
might be a decent feature. Thanks.
Running: 1.4.25, on CentOS 4.7
Cheers,
Darrin Henshaw
This email and its attachments may be confidential and are intended solely for
the use of the individual or parties' to whom it is addressed. All
.
Cheers,
Darrin Henshaw
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, June 18, 2009 15:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
Hello,
I'm supporting an Asterisk setup in the Cayman Islands(I'm in Canada), which is
running a PRI to a local telecom provider. We are looking at improving the
setup, setting up high availability etc. My manager is interested in putting a
TDMOE device in place, so we can easily switch the
have a feeling I'm still missing a lot of stuff. Anyone have any recent links
or information?
Also, anyone know of a decent way to generate the config files? I'd hate to
have to go through all of it manually? Thanks.
Cheers,
Darrin Henshaw
This email and its attachments may be confidential
.
Cheers,
Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC
Ignition Support Center | www.ignition.bm
Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288
Atlanta | Bermuda | Cayman | Halifax
-Original Message-
From: asterisk-users-boun
for it are causing me
headaches. Any suggestions would be helpful. Thanks.
Cheers,
Darrin Henshaw
This email and its attachments may be confidential and are intended solely for
the use of the individual or parties' to whom it is addressed. All comments
a call will wait in
the queue before being automatically disconnected? I tried looking through the
code directly, but I humbly admit my programming skills are lax.
I'm running Asterisk 1.2.31 on CentOS 4.7. Thanks.
Cheers,
[cid:image001.jpg@01C9A33E.72349BC0]
Darrin Henshaw | IT Administrator
} != ANSWER]|Dial|(SIP/out1/${EXTEN})
exten = s,n,ExecIf($[${DIALSTATUS} != ANSWER]|Dial|(SIP/out2/${EXTEN})
That's kind of rough but it should work.
Cheers,
Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC
Ignition Support Center | www.ignition.bm
Bermuda (441) 496-4319
have one context for outgoing calls, meaning
once I invoke MixMonitor, I need to strip the fist digit from the call and send
it back into that same extension.
Is this possible? I haven't tried it before, and am not sure if it will work.
Thanks.
Cheers,
[cid:image001.jpg@01C97187.D1841DF0]
Darrin
, January 08, 2009 12:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Goto Question
On Thursday 08 January 2009 09:56:04 Darrin Henshaw wrote:
My situation I have is based on the contexts already in place, particularly
for outbound calls, I need to do a Goto
Check out the r parameter,
http://www.voip-info.org/wiki-Asterisk+cmd+Queue
Cheers,
Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC
Ignition Support Center | www.ignition.bm
Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288
Atlanta | Bermuda
I believe you are correct Atis.
Philipp within your queue setup do you have any announcements? If so read the
posting on
queues.conf(http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf),
announcements will have an effect on the order that calls are picked up.
Cheers,
Darrin
Usually aren't those loaded using zaptel. On my machines you edit the
/etc/sysconfig/zaptel file, and comment out the unused modules leaving only the
ones you need.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David fire
Sent: Tuesday, December 09, 2008 10:31
To: Asterisk
]
Darrin Henshaw | IT Administrator | MCTS: Exchange 2007 | MCSE 2003 | LPIC
Ignition Support Center | www.ignition.bmhttp://www.ignition.bm/
Bermuda (441) 496-4319 | Cayman (345) 947-4357 | Halifax (902) 482-1288
Atlanta | Bermuda | Cayman | Halifax
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED
Yeah what Doug said ;), for more info check out:
http://www.voip-info.org/wiki-Asterisk+cmd+Read
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Thursday, November 20, 2008 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial
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