May be as simple as this:
When you terminate a call you start the call before they even get it.
When they originate a call, they start the call before you get it.
Just a guess without really thinking about this too much.
On Thu, Aug 1, 2013 at 10:28 AM, Adam Moffett
on that.
Again, thanks for your detailed response.
On Tue, Apr 16, 2013 at 9:59 PM, Nathan Anderson nath...@fsr.com wrote:
On Tuesday, April 16, 2013 6:25 PM, Todd Routhier wrote:
New Problem, now operators can pick up the previous inbound only line and
dial out to anything that matches the patterns I
OK, it's been a while since I drank from the pool of wisdom hear on the
list.
After cracking my head against the wall for a few days trying to figure
this out, I have decided to swallow my pride and take the drink.
So, on to my question:
I have some agents/operators setup in sip.conf which
What I am trying to accomplish is to run an AGI script each time an agent's
line starts ringing. I currently have the AGI firing when the agent answers
the call using the Queue command, something like
queue(MyQueue,MyAgi.php). Works great but I need the AGI to run when
the agent's phone starts
the line
gets picked up or not.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
*Sent:* Tuesday, April 10, 2012 3:15 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk
queues involved.
--Todd
On Tue, Apr 10, 2012 at 3:34 PM, Carlos Chavez cur...@telecomabmex.comwrote:
On Tue, 2012-04-10 at 15:15 -0500, Todd Routhier wrote:
What I am trying to accomplish is to run an AGI script each time an
agent's line starts ringing. I currently have the AGI firing when
, Danny Nicholas da...@debsinc.com wrote:
You have read this thread?
http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
*Sent:* Tuesday
...@lists.digium.com] *On Behalf Of *Todd Routhier
*Sent:* Tuesday, April 10, 2012 3:55 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Run AGI while agent ringing instead of
only when connected
** **
Yes Sir.. Studied it pretty hard, did I miss a solution
:
On Thu, Mar 29, 2012 at 12:09 PM, Todd Routhier fonema...@gmail.comwrote:
I have been breaking my head on this, can't find a solution.
Anyone know a way to mute DTMF on SIP? I have already tried changing the
dtmfmode option and messing with different codec/dtmfmode settings but so
far
So, I have this customer with a completely bizarr issue.
She reports that on either of her shiny new Cisco 303g phone, calls are
disconnected at exactly 12 seconds after taking caller off hold.
To be clearer:
-Answers incoming call, can talk forever no problem.
-Places caller on hold and can
Is it possible to define a customize the which sound file is played when I
send a caller to VoiceMailMain()?
By default the sound file is vm-login.codec.
Is there a way to specify which sound file is played per context or some
other way to play a different sound file in place of vm-login?
I
instructions).**
**
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
*Sent:* Tuesday, February 21, 2012 10:53 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Define custom vm-login sound
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
*Sent:* Tuesday, February 21, 2012 11:31 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Define custom vm-login sound file per VM
context?
** **
Danny
if that will break anything else now or with future upgrades.
Thanks for all the help!
--Todd
On Tue, Feb 21, 2012 at 11:47 AM, Matthew Jordan mjor...@digium.com wrote:
From: Todd Routhier fonema...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users
Wow, that looks like good stuff.
On Tue, Feb 21, 2012 at 12:24 PM, Johan Wilfer li...@jttech.se wrote:
2012-02-21 19:20, Todd Routhier skrev:
OK, this will work and is probably a better solution than the language
idea. Although, the language idea just sounds easier and a little more
fun
Version: Asterisk 1.8.x
Question: Is it possible for an agent to answer a call from a queue, then
place the call back in the queue in the same position they were in?
Seems that the answer would be yes to the remove from queue, then place
back in by having the agent just transfer the call back
queue in the order the agents
answered the first queue.
--
Jim Dickenson
mailto:dicken...@cfmc.com dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Jan 8, 2012, at 10:26 AM, Todd Routhier wrote:
Version: Asterisk 1.8.x
Question: Is it possible for an agent to answer a call from a queue
Happy New Year to all!
Asterisk 1.8.x
I have a queue to which I add agent channels like SIP/300 dynamically using
the manager interface. Once logged in, there SIP/300 of course rings when a
call is distributed to them.
How can I also get the agents cell phone to ring without actually adding it
wrote:
On 01/03/2012 01:06 PM, Todd Routhier wrote:
Happy New Year to all!
Asterisk 1.8.x
I have a queue to which I add agent channels like SIP/300 dynamically
using the manager interface. Once logged in, there SIP/300 of course
rings when a call is distributed to them.
How can I also get
On Tue, Dec 20, 2011 at 8:53 PM, Todd Routhier fonema...@gmail.com wrote:
On Tue, Dec 20, 2011 at 7:03 PM, M Maki mma...@verizon.net wrote:
I have a system working great with the exception of the sendvoicemail=yes
voicemail.conf option. I can not figure out what I am missing or have
On Tue, Dec 20, 2011 at 7:03 PM, M Maki mma...@verizon.net wrote:
I have a system working great with the exception of the sendvoicemail=yes
voicemail.conf option. I can not figure out what I am missing or have
configured wrong...
While in voicemail after selecting 3 for advanced options,
Richard,
Thanks, I'll give it a whirl.
I upgraded Asterisk to 1.8.8.0 last night and this fixed the original issue
I was having. Now calls to the other extensions continue as normal, even
when one of the SIP extensions is unreachable.
Caller-id is still lost on anything that hits follow me. I
I have attempted to set the fromstring option on a per context basis in
voicemail.conf but it doesn't seem to work. I would like to somehow either
set this based on context, number dialed into or some other way.
Would it be possible to set this option in the general section to a channel
variable,
Perfect, patched my install and it fixed the issue!
Thanks a ton Richard..
--Todd
On Fri, Dec 16, 2011 at 10:17 AM, Richard Mudgett rmudg...@digium.comwrote:
OK, read all about the patch, thanks for the fix Richard.
I would like to apply this patch to my current 1.8.7.1 but I am
] fromstring in voicemail.conf
- Original Message -
From: Todd Routhier fonema...@gmail.com
To: asterisk-users@lists.digium.com
Sent: Friday, December 16, 2011 11:32:31 AM
Subject: [asterisk-users] fromstring in voicemail.conf
I have attempted to set the fromstring option
*
Summary:
I need to be able to ring multiple numbers in followme.conf at the same
time, even if one of the SIP extensions is unreachable.
This works in 1.4.8 but not in 1.8, just barfs and sends to voice mail
instead of ringing the other 2 extensions on the same line in the
No, I get no error in the CLI at all, just shows that the followme is being
executed then dumps straight to Vmail which is defined in my dialplan on
the next line after calling the followme.
I checked out the link and it also shows problems with callerid not
passing, this is also a problem for me
Asterisk
version and recompile everything and start over? I know I can save my
configs but I was hoping for a simple fix without having to recompile
Asterisk from source etc.
Thanks for any help.
--Todd
On Thu, Dec 15, 2011 at 9:07 PM, Todd Routhier fonema...@gmail.com wrote:
No, I get no error
-users-boun...@lists.digium.com]
*On Behalf Of *Todd Routhier
*Sent:* Monday, November 28, 2011 2:31 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] Recommendations
I am currently running Asterisk 1.4.8 and have been for quite a while, it
has served me well
whacky,
trying to get use to it.
--Todd
On Mon, Nov 28, 2011 at 4:50 PM, Danny Nicholas da...@debsinc.com wrote:
1.4.42 is newer than 1.4.8.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Todd Routhier
*Sent
Network Consultant
Impala Networks
P: 505.327.7300
.
** **
*From:* Todd Routhier [mailto:fonema...@gmail.com]
*Sent:* Monday, November 28, 2011 3:55 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Recommendations
So, it's been a while and I am just lately getting back into Asterisk stuff.
I am trying to remember/understand how CDR works and after lots of trial and
error and searching the archives, google etc, I am stuck and have a few
questions.
I have setup custom CDR and am trying to figure out the
billsecs is a field in the CDR, it's already there.. Just don't bill based
on the duration field, bill based on the billsecs field and you should have
what you want.
On Wed, Sep 9, 2009 at 11:03 AM, B.Masoud @ SH i...@saudihome.com wrote:
Can you provide me some code for that?
I am NOOB
Trying to do something like this in the sip.conf under my incoming provider
profiles:
setvar=CDR(accountcode)=${EXTEN}
It seems to show up in the CDR but it's showing up exactly like this
${EXTEN}.
Is there a way to stuff the DNIS (number dialed) into the accountcode for
CDR?
I have already
Wow, been a long time since I have been on the list.. A few years to be
exact :)
Glad to be back in the land of Asterisk..
I have a box running Asterisk 1.4.8 that's been real solid and I have a
bunch of custom stuff running on it.
I am trying to move this to a new piece of hardware and
.
Two problems, I don't know how to check them by phone just yet and I will
likely never check them by phone.
Any thoughts?
Thanks,
Todd Routhier
Lightwave Technologies, LLC.
--
Start Your Dialup Internet Service!
http://www.YourOwnISP.com
Lightwave Technologies, LLC.
http
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