On 02/24/2012 10:51 PM, Jared Geiger wrote:
On Thu, Feb 23, 2012 at 2:48 PM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
On Feb 29, 2012, at 7:25 AM, Jonas Kellens wrote:
The Asterisk server still stays in the SIP Signaling path of the call, just
media does not flow through the server. You can verify this by running a SIP
debug and looking at the media endpoints.
What is it that I should be looking for in
On Tue, Feb 28, 2012 at 8:28 PM, Alejandro Imass a...@p2ee.org wrote:
What you are saying seems impossible and makes no sense unless the
router is assigning a public IP or is SIP aware and knows how to
read the routing data contained inside the SIP packets, and none of
the consumer routers are
On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
[...]
Yes, I have had no problems with Grandstream first gen ATAs, configured with
server and credentials and shipped off, they just work.
We use the HT-286, the server is on a public IP the nat setting on
On 02/29/2012 08:22 AM, Alejandro Imass wrote:
We use the HT-286, the server is on a public IP the nat setting on
asterisk is set to yes and without port re-direction the ATAs have
never connected from a private network, so I honestly find this SIP
plug and play very hard to believe. But if it
Hi all,
Currently I'm getting this message after restarting asterisk service;
Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files:
cannot modify limit: Operation not permitted
Before when I had root access I was not facing this message after that
system administrator
Hi all,
Currently I'm getting this message after restarting asterisk service;
Starting asterisk: /usr/sbin/safe_asterisk: line 86: ulimit: open files:
cannot modify limit: Operation not permitted
Before when I had root access I was not facing this message after that
system administrator
This one is simple. Open /usr/sbin/safe_asterisk and put # in first
character of line 86 and 102. Or modfy /etc/sudoers to allow your sudo to
execute ulimit.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent:
On 02/29/2012 02:30 AM, Zohair Raza wrote:
You want to allow single IP or whole subnet ?
Regards,
Zohair Raza
On Wed, Feb 29, 2012 at 4:44 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
An outside device can't register:
WARNING: getnameinfo(): ai_family
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.com wrote:
On 02/29/2012 08:22 AM, Alejandro Imass wrote:
[...]
The number of 'plain' SIP endpoints deployed behind consumer-grade NAT
devices talking to Asterisk servers on public IP addresses is in the
millions, if not the
On 12-02-29 10:15 AM, sean darcy wrote:
On 02/29/2012 02:30 AM, Zohair Raza wrote:
You want to allow single IP or whole subnet ?
Regards,
Zohair Raza
On Wed, Feb 29, 2012 at 4:44 AM, sean darcy seandar...@gmail.com
mailto:seandar...@gmail.com wrote:
An outside device can't register:
On 02/29/2012 09:25 AM, Alejandro Imass wrote:
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Flemingkpflem...@digium.com wrote:
On 02/29/2012 08:22 AM, Alejandro Imass wrote:
[...]
The number of 'plain' SIP endpoints deployed behind consumer-grade NAT
devices talking to Asterisk servers on
On Wed, Feb 29, 2012 at 9:22 AM, Alejandro Imass a...@p2ee.org wrote:
On Tue, Feb 28, 2012 at 8:58 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
[...]
Yes, I have had no problems with Grandstream first gen ATAs, configured
with
server and credentials and shipped off, they just
On Wed, Feb 29, 2012 at 8:34 AM, Kevin P. Fleming kpflem...@digium.com wrote:
Certainly there are plenty of examples of SIP endpoints working poorly
behind NAT devices, and replacing that endpoint with an IAX2 endpoint curing
the symptoms. Invariably, this is caused by the fact that the NAT
On Wed, Feb 29, 2012 at 9:44 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 02/29/2012 08:22 AM, Alejandro Imass wrote:
We use the HT-286, the server is on a public IP the nat setting on
asterisk is set to yes and without port re-direction the ATAs have
never connected from a private
On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Agreed with one exception, the endpoint behind the NAT DOES need to be setup
correctly to keep the router from seeing inbound traffic to the device as
unsolicited and drop it. That is a function of the router but
On Wed, Feb 29, 2012 at 10:43 AM, Carlos Alvarez car...@televolve.comwrote:
On Wed, Feb 29, 2012 at 8:41 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
Agreed with one exception, the endpoint behind the NAT DOES need to be
setup
correctly to keep the router from seeing inbound
An outside device can't register:
WARNING: getnameinfo(): ai_family not supported
WARNING: chan_sip.c:14456 parse_register_contact: Domain
'69.xxx.yyy.zzz:5060' disallowed by contact ACL (violating IP )
sip.conf:
[general]
...
alwaysreject=yes
dynamic_exclude_static = yes
On Wed, Feb 29, 2012 at 8:58 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
So you turned it off on the phones but use it on the Asterisk side?
Do you set a value or just use qualify=yes?
Yes, just as I said, just qualify=yes.
Did you submit a bug report? If it is easy to reproduce
Hi,
while testing asterisk 1.8.10-rc2 I stumbled across a weird behavior. I
want to notify a snom phone to reload its configuration. For this to
happen, I use the NOTIFY mechanism. I started the notify via AMI
command. Asterisk is bound to udp 25060, because all phones are
registered with a local
Hi,
a little extension to my previous post: The phone sends 200 OK for the
NOTIFY via proxy to asterisk, but asterisk seems to ignore this. About
500 ms later, the NOTIFY is repeated by asterisk. This continues up to
the final timeout (with the typical log message).
Karsten
--
On 2012-02-28 21:22:44 +, Kevin P. Fleming said:
On 02/28/2012 03:08 PM, Troy Telford wrote:
[myprovider]
type=friend
username=
secret=
context=somecontext
host=provider_server
qualify=1000
disallow=all
allow=g729
allow=ulaw
auth=md5,rsa
requirecalltoken=yes
trunk=yes
A serious bug with
I'm looking at replacing a PBX for a small business with an asterisk
box. I'm rather attracted to the idea of one of the iso distributions
where someone did most of the integration for us already ;)
Can anyone comment on the pros/cons of the various options? I'm seeing
several options out
Asterisk Now should serve your needs nicely.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett
Sent: Wednesday, February 29, 2012 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
FreePBX have also an ISO distribution - I would recommend to use that one.
HTH,
Ioan
On Wed, Feb 29, 2012 at 7:43 PM, Danny Nicholas da...@debsinc.com wrote:
Asterisk Now should serve your needs nicely.
--
_
-- Bandwidth and
On 02/29/2012 11:35 AM, Troy Telford wrote:
On 2012-02-28 21:22:44 +, Kevin P. Fleming said:
On 02/28/2012 03:08 PM, Troy Telford wrote:
[myprovider]
type=friend
username=
secret=
context=somecontext
host=provider_server
qualify=1000
disallow=all
allow=g729
allow=ulaw
auth=md5,rsa
On 2012-02-29 15:25:49 +, Alejandro Imass said:
We use SIP and IAX interchangeably, but had less hassle with IAX. The
topic of the discussion on this thread was that SIP is so awesome and
that IAX is a peice of crap.
The original question (mine) was that my sound quality when using IAX
On Wed, Feb 29, 2012 at 10:34 AM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
[...]
If you can post some SIP debug info from an ATA trying to register without
any redirection and also the relevant portions of your sip.conf, I am sure I
can help.
Do it from a new location with an el
On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com wrote:
On 2012-02-29 15:25:49 +, Alejandro Imass said:
We use SIP and IAX interchangeably, but had less hassle with IAX. The
topic of the discussion on this thread was that SIP is so awesome and
that IAX is a peice of
Hi,
I have a Portech mv-374 GSM IP gateway and I have to redirect all the
incoming calls to a certain phone number on every weeknight and all the
weekend. What would be the best solution? I have to do it by asterisk
because I have to record all the communication.
Thanks for your help in
On Wed, Feb 29, 2012 at 1:26 PM, Alejandro Imass a...@p2ee.org wrote:
On Wed, Feb 29, 2012 at 1:05 PM, Troy Telford ttelford.gro...@gmail.com
wrote:
On 2012-02-29 15:25:49 +, Alejandro Imass said:
We use SIP and IAX interchangeably, but had less hassle with IAX. The
topic of the
Thanks Danny,
I would like to know do I need to worry about this message? And why I'm
getting this ulimit message? Please provide reason briefly
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] Getting Ulimit Message after restart
asteriskservice
To:
The good folks at Asterisk wish to limit the number of open files used by
Asterisk to 32K (see line 32). If you aren't a super-user, chances are that
Linux will cut you off at a number much less than that anyway. The reason
you are getting the message; your user/sudo user can't execute ulimit.
Are there any particular reasons anybody would cite to choose one over
the other?
FreePBX have also an ISO distribution - I would recommend to use that one.
HTH,
Ioan
On Wed, Feb 29, 2012 at 7:43 PM, Danny Nicholasda...@debsinc.com wrote:
Asterisk Now should serve your needs nicely.
--
I would say that this is correct
http://support.freepbx.org/forum/freepbx/general-help/freepbx-vs-asterisknow
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett
Sent: Wednesday, February 29, 2012
I finally solve the problem,
in gentoo the permission of dir /var/run/postgresql/ is:
drwxrwx--- 2 postgres postgres 4096 Feb 29 18:09 postgresql
so if we want to connect asterisk to postgresql, we need to add the user
that runs asterisk to the group postgres
and with this finally I can
hello,everyone:
i'm a freshman on voip. there is a problem about asterisk .
there is a 4E1 with signalling(ss7) and three servers(a part has one
server and the other has two server). Two servers on the same part share
the same point code as a cluster to get load sharing Then the
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