2015-10-19 22:05 GMT+02:00 Telium Technical Support :
> If you’re still in the planning stage, there’s a lot more to think about.
> Your Asterisk failure detection will be very simplistic (is the process
> dead). Synchronization of data – without risking synchronization of
> corrupt data to a pee
If you’re still in the planning stage, there’s a lot more to think about. Your
Asterisk failure detection will be very simplistic (is the process dead).
Synchronization of data – without risking synchronization of corrupt data to a
peer. Prevent a deteriorating/failing peer corruption from co
Dear all,
I have drastically stripped down my Asterisk installation to about 70 modules
since I wanted it to be as slim as possible.
Nevertheless, Asterisk still opens an unwanted port:
root@spock:/install/asterisk-13.6.0# netstat -apnv|grep asterisk
...
udp0 0 192.168.20.48:5060
Hello,
I'm setting up an active-passive Asterisk solution on Debian Jessie
platforms.
I'm using heartbeat package.
As I'm not yet familiar with either systemd or heartbeat, I've got a couple
of questions:
1. At the moment, I'm using /usr/share/heartbeat/hb_standby or
/usr/share/heartbeat/hb_take
I wasn't able to make it back to the devcon after lunch or to as many of
the talks as I'd have liked (the excessive a/c exacerbated by symptoms
enough to be painful), so I probably missed something relevant to
this...
What is the syntax of an ALL subscription websocket url in ari?
I'd like to use
George and Mat
Here is the link to the Jar Issue.
https://issues.asterisk.org/jira/browse/ASTERISK-25477
Thanks
Bryant
From: "George Joseph"
Sent: Sunday, October 18, 2015 10:17 PM
To: "Asterisk Users Mailing List - Non-Commercial Discussi
Ok thanks Joshua
Do you know what this error means when I dial out in pjsip and the call
fails
Unable to create request with auth.No auth credent als for any realms in
challenge
Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECN
George, and Matthew
I can open an issue later today, but if you want to do it that would be
awesome as well. Please post the issue number back to this thread so I can
follow it.
Ideally the Like would work with all pjsip show commands so we can
reduce the list and drill down just like
On 15-10-19 09:12 AM, Andrew Colin wrote:
Do you know if this can be achieved with the standard sip stack in asterisk?
If you are referring to chan_sip I don't believe so but it is possible
there is some obscure option or method to do it that I am aware of.
--
Joshua Colp
Digium, Inc. | Seni
Do you know if this can be achieved with the standard sip stack in asterisk?
Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Switchboard: +27 (0)10 591 4600
Email: and...@convergedgroup.net
Web: http://www.convergedgro
On 15-10-19 08:17 AM, Andrew Colin wrote:
Hi Joshua
If i put the default_user option per endpoint would it work?
No, it's a global only option.
So what exactly does the contact_user option do?
It sets the Contact user in an outbound registration so that the URI
dialed by the remote SIP s
Hi Joshua
If i put the default_user option per endpoint would it work?
So what exactly does the contact_user option do?
I know that in freeswitch there is the option extension-in-contact.We
basically need to achieve the same functionality
Thanks
Original message
From: Joshua
On 15-10-19 07:41 AM, Andrew Colin wrote:
Hi Guys
We are using the wizard to configure our pjsip trunk(see below)
How do we get this setting to work
contact_user=username
We want to change the contact field in the sip invite to display the
username of the trunk
The Contact header can not c
Hi Guys
We are using the wizard to configure our pjsip trunk(see below)
How do we get this setting to work
contact_user=username
We want to change the contact field in the sip invite to display the
username of the trunk
[trunk_defaults](!)
type = wizard
transport = transport-udp
en
No, It directly goes the context astsms when we send the message. but it
still repeats the message sometimes.
On Mon, Oct 19, 2015 at 3:25 PM, jg wrote:
>
> I am using the asterisk 13 and I config my dialplan for the SIP messaging
> as the following :
>
> http://highsecurity.blogspot.com/2012/03
I am using the asterisk 13 and I config my dialplan for the SIP messaging as
the following :
http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html
[astsms]
exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(Fr
Dear All,
I have a query.
I want to know if there is any possiblity to modify SIP Messages Parameters
using the asterisk CLI mode.
I want to change the parameters for e.g in INVITE message. How it can be
done in asterisk.
Kindly assist me.
Regards,
*Waleed A. Khan*
--
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