Displaying 20 results from an estimated 2000 matches similar to: "Asterisk NAT friendly settings"
2014 Jan 15
2
Asterisk ignoring nat settings
Hello,
I have an asterisk box with a peer configured with nat=force_rport,comedia,
but asterisk keeps sending the audio to the private IP address and ignoring
the client peer nat settings.
If I check the "sip show peer extension", I see both symmetric RTP and
Force Rport are set to yes, but asterisk seems ignoring them.
Force rport : Yes
Symmetric RTP: Yes
Asterisk is behind a
2014 Jun 13
2
pull a call from a queue
We have a queue monitoring application running so we can see the caller
ID of callers in a queue. If we see a VIP in the queue, is there any
method to force that call to be first in line? If there's a softphone,
or queue managing application already written that does this, I'd love
to know.
2013 Aug 01
2
Asterisk 1.4 CDR vs VoIP Innovations CDR
When I compare my total minutes on the bill from VoIP Innovations, to
the number from our CDRs, I'm finding a smalish (3-4%) discrepancy in
the count of minutes. I'm wondering why it's there.
Are there different methods of counting the billable start or end point
of a phone call?
If it matters, I'm counting more termination minutes than they are and
they're counting more
2012 Sep 28
1
'Training mode'
I was asked today if we could somehow have a trainee on the phone with a
supervisor conferenced in, but somehow have it so anything the
supervisor says is only heard by the trainee and not the customer.
Is there a feature like that?
2011 Oct 04
3
Asterisk (Trixbox) - VirtualBox - Linux Host
someone have been installed Asterisk (Trixbox) on VirtualBox which is
installed on a Linux host (Ubuntu server 10.04 specifically).
I want to know if it is convenient or not, and the reaseons if i should on
shouldn't do it.
Thanks in advance.!
--
Esteban L. Cacavelos de Amoriza
Cel: 0981 220 429
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
2013 Aug 18
1
Asterisk SIP Trunk between two Asterisk Servers
Hi,
Am making a simple SIP trunk between two Asterisk server,
Server 1
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.30.2.58
context=man02-trunk
port=5060
qualify=yes
disallow=all
;allow=g729
allow=g729
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
insecure=invite,port
extensions.conf
[man02-trunk]
exten => _1X.,1,Dial(SIP/usman02/${EXTEN})
exten
2015 Jan 08
2
queue reload command
Hi
I'm using asterisk 1.8
Does anyone know how to use the queue reload command. The built in help
doesn't really help.
queue reload {parameters|membe Reload queues, members, queue rules, or
parameters
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
2015 Jun 07
3
Curious problem with NAT
Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb:
> What settings have you got for directmedia?
>
> Could you try
>
> nat=force_rport,comedia
> directmedia=no
Tried. Peer always unreachable, call not possible... :(
Other idea?
Thanks
Luca Bertoncello
(lucabert at lucabert.de)
2015 Jun 09
2
Manipulate extension state in 1.8.x
Hi
Is there any way to set the presence state of a peer to in-use in asterisk
1.8?
The idea is to integrate DND buttons on phones to BLF.
Regards
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: ish at pack-net.co.uk
w: http://www.pack-net.co.uk
Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
2015 May 28
3
Peer is UNREACHABLE
Darryl Moore <darryl at moores.ca> schrieb:
> Ahh. Seen that before! That suggests to me that you don't have your
> sip.conf records setup right.
>
> What's your sip.conf look like?
Well, here what I wrote in my sip.conf:
register => 00493511111111:MYSECRET at pbxluca/00493511111111
register => 00493512222222:MYSECRET at pbxfax/00493512222222
register =>
2014 Jul 21
1
TLS, STRP and ARA
Hi
I'm just about to upgrade to version 1.8.29.0 and have compiled with SRTP.
However, we exclusively use the asterisk realtime architecture using the
mysql connector.
Looking at tutorials we have to set encryption=yes and transport=tls for
any peer we want encrypted traffic for.
Having a look at contrib/realtime/mysql/sippeers.sql from the source code
shows that the encryption column is
2014 May 02
1
CDR billsec issue with calls forwarded through the Local channel
Hi
I'm using asterisk 1.8.23.1 but I've seen this same issue in previous
versions of 1.8. I have created some work arounds but the behaviour is
incorrect.
This is the scenario:
Call comes in and goes to appropriate dialplan
In the dialplan the call is forwarded to another number using a Local
channel (and using /n ) e.g.
Dial(Local/<my-number>@outbound-context/n,60)
The number is
2015 Sep 21
2
Call waiting for Queue Agents.
Hi All,
I have a question about the Queues.
I'm using Asterisk 11.13.0 , and I want to configure the following setup :
When there is an incoming call to the queue all agents should ring even
those that are already in call, they should receive a second call.
Is this doable in any Asterisk version ?
Thanks in advance.
-------------- next part --------------
An HTML attachment was
2016 May 11
2
How is Queue avg holdtime and avg talktime calculated
2014 Apr 16
2
FW: clients unable to auth
Hi Guys,
Just new to Asterisk and am completely stumped. I have created two accounts
as instructed. Please see below for the config of the user accounts.
[Peter]
type=friend
host=IP address
disallow=all
allow=ulaw
allow=alaw
callerid=Peter <6004>
secret=XXXXXXX
context=default
port=9060
nat=force_rport,comedia
deny=0.0.0.0
2015 Mar 13
2
ringing in queues
We use the ringall strategy for a small queue with 4 members. When a call comes in, if one of the members is busy, all the phones except the busy phone rings (as intended). While the other phones are ringing, if this busy phone becomes available again, we would like to have it start ringing. Right now it just sits idle.
Is this possible? I played with ringinuse (queues.conf) and callcounter
2017 Jan 24
2
Asterisk 13.13.1
Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
starting to complaint about packets loss, conversations are choppy!
I don't even know where to start looking! Choppy conversations happened
within users. I am using sip.conf
[1091]
type=friend
context=sip-phone
call-limit=2
trustrpid=no
callerid="dev1" <1091>
disallow=all
allow=ulaw
2015 Jul 06
2
Asterisk how to setup alarm too many outgoing calls from same user
Hello,
I would like to setup a mechanism to trigger an alarm if user is deal
too many numbers within a very short period of time. Safeguard against
users hacked accounts.
can someone help?
Thanks,
2015 Sep 22
2
How to set the global setting for each pjsip endpoint
I have many endpoints and each endpoint has some parameter in common so i
wonder is there any way to config one for all endpoints? Like in my example
I have two endpoints and I repeat the same thing,
[100]
type=endpoint
aors=100
auth=100-auth
allow=ulaw,alaw,gsm,g726
context=from-internal
callerid=device <100>
dtmf_mode=rfc4733
use_avpf=no
ice_support=no
2013 Jul 02
1
Asterisk trunking between two location
Am using Asterisk 11.2 in one location and 11.1 in another location.
when I trunk between two servers, the status is unreachable.
But with different server with 11.2 and 11.2 it works fine.
I tried both IAX and SIP.
the trunk in sip.conf what i have is,
[serverb]
type=friend
username=serverb
secret=serverb
host=10.10.10.5
port=5060
context=default
insecure=port,invite
dtmfmode=rfc2833