On Sat, Dec 25, 2010 at 11:28 AM, Nick Ustinov wrote:
> Hello
>
> We have recently upgraded to Realtime engine (sip buddies and
> extensions) and now have problems with calling local SIP users.
> I have rtcachefriends=yes but tried with 'no' and it's even worse.
> (asterisk 1.8.1.1 + realtime mysq
On Sat, Dec 25, 2010 at 7:37 AM, Ryan Wagoner wrote:
> On Fri, Dec 24, 2010 at 7:40 AM, Jim Dickenson wrote:
>> If you set bindaddr in any conf file you will need to change the IP address
>> there.
>> --
>> Jim Dickenson
>> mailto:dicken...@cfmc.com
>> CfMC
>> http://www.cfmc.com/
>
> You will al
On Sat, Dec 25, 2010 at 7:41 PM, dave george wrote:
> Yes we have that set in logger.conf.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Ustinov
> Sent: Saturday, December 25, 2010 6:25 PM
> To: A
Yes we have that set in logger.conf.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Ustinov
Sent: Saturday, December 25, 2010 6:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Hello and Merry X-Mas,
can someone tell me, which USB GSM-Modem I should use, if I want to
setup a "GSM-To-SIP" gateway?
I need at least 4 GSM-Modem which must be Quad-Band to work in Europe,
North-Africa and Near-East.
Thanks, Greetings and nice Day/Evening
Michelle Konzack
--
On Sat, Dec 25, 2010 at 8:58 PM, Richard Kenner wrote:
> I'm confused exactly what's supported with LDAP and Asterisk. What I want
> to do is to have SIP peer information read directly (in realtime) from LDAP.
> Can this be done? If so, with what Asterisk versions?
>
> --
Here is the link
https
I'm confused exactly what's supported with LDAP and Asterisk. What I want
to do is to have SIP peer information read directly (in realtime) from LDAP.
Can this be done? If so, with what Asterisk versions?
--
_
-- Bandwidth and C
Make sure you have
dateformat=%F %T
in logger.conf
On Sun, Dec 26, 2010 at 1:04 AM, Dave George wrote:
> My server is being attached all day and fail2ban is not stopping the
> attack. I updated stamstamp to match fail2ban requirements.
>
> [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830
My server is being attached all day and fail2ban is not stopping the
attack. I updated stamstamp to match fail2ban requirements.
[2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830
handle_request_register: Registration from '"7002" '
failed for '38.108.40.94' - No matching peer found
[2010-12-2
Server will have two fix public ips.
Dave
> Original Message
> Subject: Re: [asterisk-users] load balance with 2 wan connections
> From: Alejandro Imass
> Date: Sat, December 25, 2010 1:58 pm
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
>
> On Sat, Dec
OK my sip.conf looked OK and to save ports, I changed the rtp.conf to
specify the following and altered the linksys to match.
rtpstart=1
rtpend=10500
I don't think I remember seeing Stun on the sonicwall, but it does seem
to have a lot of VOIP stuff. They do have a help for VOIP so I
On Sat, Dec 25, 2010 at 1:18 PM, dave george wrote:
> Need some advise or paid help on running asterisk on two WAN connection. I
> need load balancing and failover support.
>
> WAN: 1 DSL + 1 Cable ISP.
>
There are _many_ issues. First outgoing and incoming traffic is
completely different for wh
On Sat, 25 Dec 2010 09:49:29 -0500, John Ervin
wrote:
>So, assuming your Asterisk box is behind one firewall (Linksys/Tomato
>Software) and your Wireless SIP phone is behind another firewall
>(SonicWall 1260 Enhanced). Is there anything special that I have to do
>to the firewalls.
If the Soni
Need some advise or paid help on running asterisk on two WAN connection. I
need load balancing and failover support.
WAN: 1 DSL + 1 Cable ISP.
Dave
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Ne
Hello
We have recently upgraded to Realtime engine (sip buddies and
extensions) and now have problems with calling local SIP users.
I have rtcachefriends=yes but tried with 'no' and it's even worse.
(asterisk 1.8.1.1 + realtime mysql)
Here's an example:
User 1000 registers successfully and can t
Hello!
Ater several successful SRTP-enabled calls with SRTP set to Mandatory,
asterisk starts to give the following warnings in Log:
WARNING[13714] res_srtp.c: SRTP unprotect: authentication failure
(continiously)
and client hears no sound. After i restart the client program it works
fine again
Hello Michael,
You could try to achieve this functionality in dialplan by using the
applications AddQueueMember/RemoveQueueMember which are used to
dynamically add/remove queue members.
An example dialplan flow for agent login will be
1. get the SIP interface from which the agent is logging in
So, assuming your Asterisk box is behind one firewall (Linksys/Tomato
Software) and your Wireless SIP phone is behind another firewall
(SonicWall 1260 Enhanced). Is there anything special that I have to do
to the firewalls. I do have the Asterisk firewall configured to work
(ports 5060 & 1000
On Fri, Dec 24, 2010 at 7:40 AM, Jim Dickenson wrote:
> If you set bindaddr in any conf file you will need to change the IP address
> there.
> --
> Jim Dickenson
> mailto:dicken...@cfmc.com
> CfMC
> http://www.cfmc.com/
You will also need to change externip and localnet if those are set in sip.co
Greetings and Merry Christmas,
We're trying to implements a queue and agents login mechanism on our
Asterisk.
After going over the documentation, we're unsure if we got it right.
We wish to setup a "hotdesk" mechanism, where an agent comes to a station
with a PC & IP phone (that is defined as a
On Thu, Dec 23, 2010 at 1:39 PM, Tilghman Lesher wrote:
> On Thursday 23 December 2010 09:16:26 Bryant Zimmerman wrote:
>> In the voip-info posting
>
> Right here is why you fail. Voip-info is very often wrong. Refer to the
> documentation that comes with Asterisk for definitive information. In
On Fri, Dec 24, 2010 at 6:40 AM, Jim Dickenson wrote:
> If you set bindaddr in any conf file you will need to change the IP address
> there.
> --
> Jim Dickenson
> mailto:dicken...@cfmc.com
> CfMC
> http://www.cfmc.com/
>
>
> On Dec 24, 2010, at 4:30 AM, Asterisk Man wrote:
>
> Friends,
>
>
>
> Do
22 matches
Mail list logo
