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Scott Griepentrog
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the same go for issuing a
sip show peers through the AMI? And do you know where I could find
information of what asterisk versions may use cached information instead?
What would you suggest be better ways to monitor asterisk information?
On 24 April 2014 17:58, Scott Griepentrog sgriepent
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Scott Griepentrog
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:29 PM, Scott Griepentrog wrote:
Your bottleneck is most likely your drive bandwidth. Even with SAS
drives, you'll need to move to a raid 5+ solution with 6+ drives to
continue to increase the concurrent calls, or use a storage appliance.
To confirm this, install the tool nmon and use the v
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Scott Griepentrog
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For clarification: I was speaking of the directory.php which didn't
support realtime last I looked at the code.
The app_directory built in to Asterisk should support realtime.
Can you determine which one you're using?
On Wed, Jul 30, 2014 at 9:46 AM, Scott Griepentrog sgriepent...@digium.com
-boun...@lists.digium.com] *On Behalf Of *Scott Griepentrog
*Sent:* Wednesday, July 30, 2014 10:59 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Directory app not working with realtime
For clarification: I was speaking of the directory.php which
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via another app).
I would recommend filing an issue on the yaai project for 7 support. There
may also be some other resources I've missed.
On Thu, Aug 28, 2014 at 3:18 PM, Marek Cervenka cerv...@fpf.slu.cz wrote:
it's old. sugarcrm v7 is not supported
Dne 28.8.2014 v 14:54 Scott Griepentrog
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, Scott Griepentrog
sgriepent...@digium.com wrote:
It would appear that you have the Asterisk server on a public IP
address, your two endpoints are behind a NAT, and you have rewrite_contact
enabled in pjsip.conf.
In which case, what you are seeing is correct. In order to be able to
send a call
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Am 02.04.2015 um 22:00 schrieb Scott Griepentrog sgriepent...@digium.com
:
Actually, the IP address is still used to identify the incoming invite.
With the insecure=port option set, Asterisk will presume the invite to
still match the trunk account even if the NAT router has mangled (changed
not require authentication of incoming INVITEs
;insecure=port,invite ; (both)
Do I understand correctly that in this mode the IP address is not checked
and no authentication is required?
Am 02.04.2015 um 20:11 schrieb Scott Griepentrog sgriepent...@digium.com
:
I'd be curious if setting
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, and
then the usual methods of moving channels in to bridges with ARI could be
used.
On Sat, May 23, 2015 at 1:33 AM, Nick Awesome jl...@me.com wrote:
recreate Echo, if that is possible. trying to recode all dialplan to
stasis application
On 22 May 2015, at 19:29, Scott Griepentrog sgriepent
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test the most recent release of
Asterisk (version 13 at the moment).
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recent release of
Asterisk (version 13 at the moment).
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Digium, Inc · Software Developer
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with this. Even the original Queue code I
had includes the added SIP headers with it’s INVITE to the Agent.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Scott Griepentrog
*Sent:* Thursday, August 27, 2015 4:28 PM
:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Scott Griepentrog
*Sent:* Thursday, August 27, 2015 1:57 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Is it possible to perform PJSIP Add
Header prior to calling Queue and have it part
s:
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