Displaying 20 results from an estimated 1000 matches similar to: "'sip show users' shows NAT RFC3581"
2006 Jun 22
3
Showing Current Calls
Can someone recommend the best way to view current calls in progress on the Asterisk console?
Neither the 'show channels' or 'sip show channels' commands are easy to read.
hestia*CLI> show channels
Channel Location State Application(Data)
SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2)
SIP/2944079-e7f2
2006 Apr 20
1
Background() and Read()
I'm having some issues with Background() and Read() commands.
See the example below. This is when I wait for Background to finish playing the sound file, before entering '12345#'.
All works fine.
hestia*CLI>
-- Executing Answer("SIP/2944093-3366", "") in new stack
-- Executing Wait("SIP/2944093-3366", "1") in new stack
--
2006 May 11
4
'extensions reload' clears Regextens
I hope I have this wrong, but when I have a bunch of priority 1 NoOp's created from regexten in sip.conf, and I do an 'extensions reload', I lose all the priority 1 NoOps! This can't be right... this means that in a production environment, if you make a change to your dialplan and do an 'extensions reload', you lose your ability to terminate calls to phones on this system.
2011 Apr 24
1
problem with qemu
Hi All,
I use Ubuntu server 10.04 LTS as virtualization platform.
Actually running kernel 2.6.32-31-server #61-Ubuntu S
root at jupiter:~# uname -a
Linux jupiter 2.6.32-31-server #61-Ubuntu SMP Fri Apr 8 19:44:42 UTC
2011 x86_64 GNU/Linux
We have on running virtual
root at jupiter:~# virsh list --all
Id Name State
----------------------------------
1 kvmtik.4safety.cz
2014 Feb 19
1
Asterisk as a client: can I get the remote SIP server to ignore rport?
Hi list,
I have a fresh install of Asterisk 12.0.0 and I'm going to use it only
as a client. I'm trying to SIP REGISTER with a remote SIP provider.
The situation is that Asterisk is running in a VMware VM with a RFC IP
address (192.168.1.2). The provider of the VM performs static NAT from
the RFC IP address to a dedicated public IP address, however, they are
rewriting ports at will.
2007 Apr 25
2
No Audio with SIP to only one provider when switching servers
I have been running Asterisk for years on a machine with a public
IP. Most recently, I have been running 1.2.17, from the day it
came out, with no (noticeable) problems.
Yesterday, I switched over to a new server that is on the same
public subnet, one higher than the original server.
I built 1.2.17 from source on that machine (as I did on the old
server). My firewall on the new machine is
2006 Dec 08
1
Douglas Garstang <dgarstang@oneeighty.com>
On Fri, 2006-12-08 at 04:26 -0700, Douglas Garstang wrote:
> Hi Steve.
>
> Thanks, but unfortunately, I can't be involved in that. We are
> running Asterisk in a production environment and we're using
> 1.2, not 1.4. I don't have the resources to work with 1.4.
> Last time I filed a bug against 1.2 I got told off.
>
2006 Jun 26
7
'500 Internal Server' Error on SIP NOTIFY
Is anyone getting '500 Internal Server' errors back from their Polycom phones when Asterisk sends a SIP NOTIFY message to them?
I called Polycom tech support, who where utterly useless.
Of course Polycom won't officially support it anyway, as they only support Asterisk Business Edition. We're using 1.2.9, but it's been ocurring for quite some time. We have about 35 phones and
2006 Apr 19
1
Callerid matching in extensions.conf
I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in extensions.conf changed recently?
exten => 5555,1,NoOp(${CALLERID})
hestia*CLI>
-- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel" <2944093>") in new stack
== Auto fallthrough, channel 'SIP/2944093-d24d' status is 'UNKNOWN'
This
2016 May 26
1
Failed to join domain: failed to lookup DC info for domain '<EXAMPLE.COM>' over rpc: The object name is not found.
Try to ping from client to server with its hostname. Sounds like dns
problem.
ping server
Then try to ping its ip address.
Then try to add server address to host file.
Ex
192.168.8.30 server.example.com server
Best
M
On May 26, 2016 12:02, "Nico Speelman" <nico at speelmanrobben.nl> wrote:
> Hello,
>
> I've been trying to add a new server to my Samba 4 Active
2007 Apr 16
2
sip tcp support
Hi all,
i have asterisk 1.2.17 with sip tcp support and i am
trying to connect asterisk with HiPath 4000 V.3.0
using SIP. I can see the registration from the HG3540.
But when i try to place a call from Asterisk to
HiPath, the call fails with SIP/2.0 603 Declined.
The strange thing is that the first INVITE uses tcp
and the response is a 100 TRYING, the next 7 INVITE
are using udp and the respose
2006 Jun 26
1
Email notification
Is there a way to get asterisk to send you a email when it looses or an extension doesn?t re-register
Roger Workman
Business Development
Upperclassman/Universal Holdings LLC
Voice: 304.324.3800
Fax: 304.324.3801
ICQ: 4447584
Website: http://www.upperclassman.net
Billing Questions: billing at upperclassman.net
Rental Questions: rentals at upperclassman.net
Maintenance: help at
2006 Mar 29
1
Realtime Users/Peers/Friends - Ick
I've been going in circles for a few weeks now with Realtime SIP.
My extconfig.conf has:
sipusers => mysql,dbname,ast_sip_users
sippeers => mysql,dbname,ast_sip_users
When I do a 'sip show peers' I see all my phones. When I do a 'sip show users' I only see a few of them. I can't work out why this is the case. They are also coming up with NAT as
2006 Jun 15
1
Distributed ACD Queues
It seems that I am having a heck of a time explaining my attempts at distributing ACD Queues to the list. Here's one little problem, that's only a piece of the puzzle.
dundi.conf:
180q => global_dundi_q_pbx1,100,IAX,dundi1:${SECRET}@${IPADDR}/${NUMBER},nopartial
180q => global_dundi_q_pbx2,200,IAX,dundi2:${SECRET}@${IPADDR}/${NUMBER},nopartial
180q =>
2009 Jul 14
1
Polycom Spectralink 8002 WiFi Phones
Has anyone played with this phone? i cant seem to get it to work
properly, i manged to get it registered and can make calls from it, but
i havent been able to make it receive calls. Weird thing its that if you
make a call from it and while you are on that call you dial its number
does calls go thru in second line, but as soon as you terminate both
calls it wont recieve any calls again.
Heres
2016 May 26
3
Failed to join domain: failed to lookup DC info for domain '<EXAMPLE.COM>' over rpc: The object name is not found.
Hello,
I've been trying to add a new server to my Samba 4 Active directory, but I've been failing so far. I'm running the command "net ads join -k" and it fails with "Failed to join domain: failed to lookup DC info for domain '<EXAMPLE.COM>' over rpc: The object name is not found." The answers I found so far imply a problem with the RPC service, but
2016 May 26
0
Failed to join domain: failed to lookup DC info for domain '<EXAMPLE.COM>' over rpc: The object name is not found.
Try to ping from client to server with its hostname. Sounds like dns problem.
ping server
Then try to ping its ip address.
Then try to add server address to host file.
Ex
192.168.8.30 server.example.com[1] server
Best
M
On May 26, 2016 12:02, "Nico Speelman" <nico at speelmanrobben.nl[2]> wrote:
Hello,
I've been trying to add a new server to my Samba 4 Active directory, but
2010 Nov 06
2
One way voice with Asterisk
Let me explain:
When I dial into Asterisk ( I have a SIP trunk - which I need to make
sure is not faulty), I only get one-way voice communication.
The calling party, from the SIP trunk hears nothing - the extension
rings on the Asterisk server (you can see it in the CLI and hear it at
the computer), and the softphone rings
However, when you answer the SIP softphone , you can only hear the
2011 Jan 10
0
No subject
do not know why. Anybody has a clue what could be wrong ? Is this a
bug ?
[I rebooted asterisk, and now it works.]
Regards
Axelle.
Logs of failed registration:
> sip show users
Username Secret Accountcode
Def.Context ACL NAT
IMSI208011234567890 sip-local
No RFC3581
IMSI208302141472352
sip-external No
2006 May 30
5
Compiling Asterisk-addons
Did the following:
svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons
svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk
svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
svn checkout http://svn.digium.com/svn/libpri/trunk libpri
Compiled and installed zaptel, libpri, asterisk and finally asterisk-addons.
Following errors ocurrs when compiling