,b8zs
span=11,3,0,esf,b8zs
span=12,4,0,esf,b8zs
Hope that helps.
Matthew Fredrickson
Digium, Inc.
On 12/20/12 10:42 PM, Dave George wrote:
I have a box with 12 T1s (4 Te410P cards). The PSTN provider is
reporting slips and ask me to update the clock source. I have my
system.conf
totchans=24
irq=50
type=digital-T1
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=B8ZS,AMI
framing_opts=ESF,D4
coding=B8ZS
framing=ESF
[root@aislecom28502 dahdi]#
Thanks,
Dave George
AIsleCom Inc.
1 473 520 1000
1 561 674 3838
I am using asterisk for voice mail. During DTMF collection Asterisk
stop sending any RTP Packets. The gap between two consecutive packets
are 4 seconds, which is huge enough to screw up the jitter buffer. When
ever asterisk stops to receive DTMF, the RTP stream is cut and we loose
audio.
I
, May 25, 2012 5:38 pm
To: asterisk-users@lists.digium.com
On 05/25/2012 04:30 PM, Dave George wrote:
I am using asterisk for voice mail. During DTMF collection Asterisk
stop sending any RTP Packets. The gap between two consecutive packets
are 4 seconds, which is huge enough to screw up
My users dial *120 get to an IVR menu that plays their balance and then
ask them for a voucher. Ater the balance is played and the request for
the voucher is played the user don't hear any other audio from the
asterisk box. I can see the asterisk server playing the files to ask
for the voucher
, then
contact Digium support.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave George
Sent: Tuesday, August 02, 2011 10:52 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re
TE410P card down.
I have three (3) TE410P in one machine running asterisk with SS7.
My problems started last week when one of my cards started switching to E1
every time after reboot. I set the following in dahdi.conf and that solve
the problem.
/etc/modprobe.d/
options wct4xxp
...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, August 02, 2011 10:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TE410P hardware problems
On Tue, Aug 02, 2011 at 09:22:46PM -0400, Dave George wrote:
TE410P card down.
I have three (3
] On Behalf Of Shaun Ruffell
Sent: Tuesday, August 02, 2011 10:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] TE410P hardware problems
On Tue, Aug 02, 2011 at 09:22:46PM -0400, Dave George wrote:
TE410P card down.
I have three (3) TE410P in one machine
My asterisk server is getting bogged down every 5 minutes. My ping time is
going from 60ms to 800 ms and the call quality is bad.
I have fail2ban running and I am using iptables. I have two ip connections
to the box.
How can I tell if the poor performance is due to sip attacks? I don't see
We have another gateway in the USA that will send traffic to both IPs. The
US gateway will load balance the traffic to both IPs.
This is not used for phones. It is used mainly for wholesale traffic.
Asterisk is being used as an SS7 gateway.
Each DSL limits us to about 16 calls. We are
Subject: Re: [asterisk-users] sip attack.. fail2ban not stopping attack
On Sat, Dec 25, 2010 at 04:04:59PM -0700, Dave George wrote:
My server is being attached all day and fail2ban is not stopping the
attack. I updated stamstamp to match fail2ban requirements.
How about posting your fail2ban config
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 --
-users@lists.digium.com
On Sat, Dec 25, 2010 at 1:18 PM, dave george dgeo...@teletoneinc.com 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
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 sip:7...@x.x.x.x'
failed for '38.108.40.94' - No matching peer
: Re: [asterisk-users] sip attack.. fail2ban not stopping attack
Make sure you have
dateformat=%F %T
in logger.conf
On Sun, Dec 26, 2010 at 1:04 AM, Dave George dgeo...@teletoneinc.com
wrote:
My server is being attached all day and fail2ban is not stopping the
attack. I updated stamstamp
I have my asterisk Server A registered as a client with another asterisk
Server B.
When I place a call from Server A to B I get the following:
WARNING[834]: chan_sip.c:12673 check_auth: username mismatch, have SS72,
digest has openbts1
NOTICE[834]: chan_sip.c:19961 handle_request_invite: Failed
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Axelle
Sent: Friday, December 17, 2010 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] setting up callerid
Hi Dave,
On Thu, Dec 16, 2010 at 1:52 PM, dave george dgeo
-Commercial Discussion
Subject: Re: [asterisk-users] setting up callerid
Am 12.12.2010 20:49, schrieb dave george:
I am using Asterisk 1.6.2.5-0 running on ubuntu and I have a problem
passing called ID on calls to the PSTN
When I make a call to the PSTN the caller-Id is showing up
-Commercial Discussion
Subject: Re: [asterisk-users] setting up callerid
Am 12.12.2010 20:49, schrieb dave george:
I am using Asterisk 1.6.2.5-0 running on ubuntu and I have a problem
passing called ID on calls to the PSTN
When I make a call to the PSTN the caller-Id is showing up
I am using Asterisk 1.6.2.5-0 running on ubuntu and I have a problem
passing called ID on calls to the PSTN
When I make a call to the PSTN the caller-Id is showing up as
IMSI310410381554227
I want the number set in the callerid field to show up.
My peer is setup as follows:
/hack attempts have gone
up significantly, thanks to cloud computing I guess.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-09-17 5:28 PM, dave george dgeo...@teletoneinc.com wrote:
I am getting several hundred registration attempts on my aserterisk per
minute. I have fail2ban installed
I am getting several hundred registration attempts on my aserterisk per
minute. I have fail2ban installed but it's not stopping the attempts. Any
suggestions. Whatever they are using is changing the userid on each
attempt.
Latest IP: 209.172.57.219
Thanks,
Dave
--
We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI)
cards.
I am having trouble completing faxes. Carrier send calls to me using SIP.
Any recommendation to have some success with Fax.
We trying using T.38 pass through and using G711U codec.
Asterisk Version 1.6.1.1
- Non-Commercial Discussion
Subject: Re: [asterisk-users] Faxes
On Fri, Sep 3, 2010 at 10:49 AM, dave george dgeo...@teletoneinc.com
wrote:
We have asterisk connected to the PSTN VIA TE410P (ANSI SS7 to T1 PRI)
cards.
I am having trouble completing faxes. Carrier send calls to me using SIP
Subject: Re: [asterisk-users] Faxes
On Fri, Sep 3, 2010 at 11:50 AM, dave george dgeo...@teletoneinc.com
wrote:
The asterisk box is connected to the PSTN using TE410 cards. Asterisk
talk
SS7 to the PSTN. On the IP side I use SIP. I terminate calls onto the
PSTN.
You don't say the percentage
] On Behalf Of Kevin P.
Fleming
Sent: Friday, September 03, 2010 2:17 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Faxes
On 09/03/2010 10:50 AM, dave george wrote:
The asterisk box is connected to the PSTN using TE410 cards. Asterisk
talk
SS7 to the PSTN. On the IP side I use SIP
I have a box (Genband) expecting the following:
100 trying
180 ringing with SDP
Or
100 trying
183 with SDP
And asterisk is sending:
100 trying
180 ringing
183 with SDP
Any way to modify asterisk to send what he is expecting?
Thanks,
Dave
--
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] no ring back 180 with SDP
On Friday 11 June 2010 09:31:43 dave george wrote:
Any way to modify asterisk to send what he is expecting?
Probably, but what you really should be asking is why the endpoint is not
RFC-compliant
I tried no, yes and never in the sip profile for that carrier and it did not
make a difference.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von
Klitzing
Sent: Friday, June 11, 2010 1:59 PM
To:
I set it under the sip profile for the box sending calls to asterisk.
[BREKEKE]
type=peer
context=wholesale
host=x.x.x.x
nat=no
canreinvite=no
progressinband=yes
dtmfmode=rfc2833
insecure=port
disallow=all
allow=g729
Thanks,
Dave George
-Original Message-
From: asterisk-users-boun
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