Not it didn't,
Did you execute asterisk -r
or /usr/sbin/asterisk -r ?
If not working did you execute
asterisk -gc ?
Kind Regards
On Mon, Jul 22, 2013 at 10:41 AM, Meadows Hoa meadows_...@yahoo.com wrote:
We have Asterisk1.8.11 and can not move to a newer version right
Depends on used kernel and perhaps on other hardware you are using.
Am 23.07.2013 00:09, schrieb bilal ghayyad:
Hello
I need to deploy asterisk on production and same thing for DAHDI,
which version is recommended for this?
Regards
Bilal
--
The reason for this is we have one primary company office but there are two
entities and if someone call the Denver number it would be for 1 organization
and would ring differently helping our staff remember how to answer the phone
for the Denver organization rather than for the Colorado
Josh!
This is a typical application for using different SIP identities. Essentially, no programming is
involved---just configuration. Hopefully, your SIP phones allow for several identities.
In Asterisk you would dispatch the calls and connect to the proper SIP accounts. You would also
On Tuesday 23 July 2013, Josh Hopkins wrote:
The reason for this is we have one primary company office but there are two
entities and if someone call the Denver number it would be for 1
organization and would ring differently helping our staff remember how to
answer the phone for the Denver
You can use anything that is discriminable to connect a call to its proper endpoint. SIP headers
have the disadvantage that the format may depend on a the phone model.
jg
--
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-- Bandwidth and Colocation Provided by
A J Stiles wrote:
On Tuesday 23 July 2013, Josh Hopkins wrote:
The reason for this is we have one primary company office but there are two
entities and if someone call the Denver number it would be for 1
organization and would ring differently helping our staff remember how to
answer the phone
hello, you should provide more information, like:
what have you tried yet?
did you restart asterisk?
did you find something in the log?
greetings
On 07/23/2013 04:06 PM, Carlos Rojas wrote:
Not it didn't,
Did you execute asterisk -r
or /usr/sbin/asterisk -r ?
If not working did you execute
Hi all,
I have a Queue with 3 members:
SIP/100
SIP/200
SIP/300
When call arrives SIP/100 is ringing.. After given timeout ringing stops
but call is not routed to next member but SIP/100 starts ringing again.
I know that this is because SIP/100 is still available in the Queue but is
it any way
Read queue configuration esp. QEUUESTRATEGY and agent TIMEOUT.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jarek Jarzebowski
Sent: Tuesday, July 23, 2013 3:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
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