On Wednesday 22 January 2014, José Pablo Méndez Soto wrote:
Hello,
Is there anyway to encrypt or scramble a bit the secret used to register
with a provider? Im talking about the
register = fromuser@fromdomain:secret@host
directive in
Hi,
I face a problem which look like the same as David with his thread
Asterisk not receiving call from VPN address.
I had an Asterisk (Elastix) working well in a VM (Debian Wheezy - KVM)
having IP 192.168.111.14, my phone network is in the range 192.168.10.x
I updated lately to 11.7.0
Thanks that's perfect!
I would use the AMI but I have to run against Asterisk 1.4 and the ami command
'coreshowchannels' didn't appear until Asterisk 1.6
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of
Le 22/01/2014 14:01, Administrator TOOTAI a écrit :
Hi,
I face a problem which look like the same as David with his thread
Asterisk not receiving call from VPN address.
I had an Asterisk (Elastix) working well in a VM (Debian Wheezy - KVM)
having IP 192.168.111.14, my phone network is in
Le 20/01/2014 03:51, David Cunningham a écrit :
Hi,
We have a Kamailio and Asterisk cluster, both machines being on a real
103.x IP address and also on a 172.x OpenVPN address.
The problem is that when Kamailo receives a call from the VPN and
forwards it to the Asterisk server on it's 103.x
We fairly recently switched service providers for our 4 PRI circuits. Since
that time, we started to notice some failed inbound calls. These calls
terminate with an ISDN cause code 47 'resource unavailble'. Most of the
time I see this error on the first or second channel on the second span in
a
Hello,
is there a mailinglist where I can post questions regarding Digium
IP-phones ?
I have the following question :
I'm trying to provision a Digium D70 IP-phone from a https provisioning
server.
The Digium D70 contacts the provisioning server correctly but seems to
log in with the
On Mon, Jan 20, 2014 at 5:16 PM, Dale Noll dn...@wi.rr.com wrote:
We fairly recently switched service providers for our 4 PRI circuits.
Since that time, we started to notice some failed inbound calls. These
calls terminate with an ISDN cause code 47 'resource unavailble'. Most of
the time I
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 TEI=0 Call Ref: len= 2 (reference 6918/0x1B06) (Sent
from originator)
[Jan 14 12:56:04] VERBOSE[13262] chan_dahdi.c: [Jan 14 12:56:04]
PRI Span: 4 Message Type: RELEASE COMPLETE (90)
[Jan 14
I am sure I need outboundproxy :)
We use Kamailio in front of our softswitch as a type of firewall and
because not all sip endpoints can set the headers such that they will
work with said softswitch. Calling messages (and calling!) work just
fine through the proxy - that setting just doesn't
We use opensips as a type of firewall as well and don't need to set
qualify=yes.
N
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Okay - maybe I'm just suffering from a moment of horrible ADD - but, I'm a
little lost.
I see that Asterisk 12 has a nice REST API - very nice - something I can
use. However, and this is gonna sound dumb - but all the CLI commands are
different now. What did I miss?
Can anyone, please, anyone
Maybe it's just me if I'm not mistaken the three things you listed are
usually configured using the config files not on CLI.
Jacob
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James
Wystead
Sent: Wednesday, January 22, 2014
I looked on http://www.voip-info.org - maybe I missed it?
The Digium/Asterisk site - I see all sorts of cool things about the REST
API, but CLI - maybe I missed it!!?? - again, I could be looking in the
wrong place?
https://wiki.asterisk.org/wiki/display/AST/Home
To my knowledge the
Hello,
We tried Asterisk 1.8 and 1.6, but not yet Asterisk 11. We'll keep it in
mind.
In the meantime we've decided to try a different network configuration
instead, so the VPN network is separated from what Asterisk sees.
Thanks for all the advice given.
On 23 January 2014 00:42,
I'm looking at setting type=peer vs type=user (in both IAX and SIP conf
entries), and I found a comment attributed to digium
(http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer) in 2005 that
type=user is depricated and that we should only use type=peer
Is that still correct? Will
I'm creating an AMI client and I only want to get newchannel events (as well as
responses to any actions I initiate). What would I set the eventmask to to
only get the newchannel events?
For anyone else looking...is there a table somewhere online that maps events to
their eventmask
We use opensips as a type of firewall as well and don't need to set
qualify=yes.
As I said, I don't *need* to set qualify=yes for things to work. It's
just that trunk failover takes ~30 seconds. If the first/main trunk
in an outbound route is down, outbound calls just sit with dead air
for a
We use opensips as a type of firewall as well and don't need to set
qualify=yes.
As I said, I don't *need* to set qualify=yes for things to work. It's
just that trunk failover takes ~30 seconds. If the first/main trunk
in an outbound route is down, outbound calls just sit with dead air
for a
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