.
-Gerard
On Fri, 2014-01-10 at 15:01 -0600, Matthew Jordan wrote:
On Fri, Jan 10, 2014 at 9:45 AM, gm1 g...@curtissystemssoftware.com wrote:
On connection to an incoming call via PSTN where
asterisk [11.7.0] is Dialing an internal extension
on answering the call there is about 6-7 seconds
On 03/21/13 14:14, Gerard wrote:
I think a simple tcpdump of the traffic will show the mystery. It can
be your provider doing something nasty. Have you tried using some
other cheap SIP termination? or arrange a fake termination yourself
on another server?
Leandro
I thought so too
() to the dialplan.
-Gerard
On 02/26/13 13:19, Gerard wrote:
Hi everyone,
I'm having a hard time figuring this issue out, we just switched from a
T1 PRI to a SIP trunk provider and that's when the issue started.
Now when someone forwards all calls on their phone to a cellphone, when
a customer
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gerard
Sent: Friday, March 01, 2013 9:33 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Delay before audio starts
I've found a workaround of sorts, If I change my below code
.
Asterisk 11.2.1 built by root @ phonesys2 on a i686 running Linux on
2013-02-23 01:40:02 UTC
Any help would be appreciated,
Thanks,
--
Gerard Saraber
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
.
A simple POTS line and you're out of luck.
If you have a T1 (i.e. ISDN-PRI) and a co-operative carrier, it may just
work or they may enable it if requested.
You could always use a co-operative SIP carrier (like Vitelity). A penny
or 2 per minute will keep your someone happy.
--
Gerard
On 10/13/10 14:52, Danny Nicholas wrote:
I think FOLLOWME is going to fix this for you
Can you elaborate please? is this a feature from our carrier? or
something that will be built into asterisk? sounds like a useful fix :)
--
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630
.
--
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)
--
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New to Asterisk? Join us for a live
the schematic:
customer - our office ---callforward-- cellphone
so should I call ATT and ask them to unlock our callerID so I can set
the outgoing callerID to the customer's number in my dialplan? or is
there some other way to handle this?
I appreciate any input,
Thanks!
--
Gerard Saraber
Network Admin
' its custom ringtone on
occasion.
but as you can see, no 8.12 is available for the 7962G..
-Gerard
On 10/05/10 16:55, James Miller wrote:
I know this doesn't answer your question directly, but Where are you
getting the Sip 9.0 software? It is not available on Cisco's website.
I have Sip 8.9
mind testing out a polycom phone though.
The chan-sccp guys are really awesome, it's just not quite ready for my
office at the moment, it's getting there though, maybe that's a better
bet. Especially if I can't get SIP sorted out.
-Gerard
On 10/06/10 07:56, JR Richardson wrote:
Hi list,
I
anyone know if I need to adjust my .cnf.xml file, or is it a bug of
some sort?
Thanks for any input,
--
Gerard Saraber
Network Admin.
Rarcoa, Inc
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
: v2 - 1792 (built by 'root' on 'Mon Aug 23 17:42:15
CDT 2010')
chan_sccp v3 crashed too much to be useful, so I went back to v2 for now.
Any input would be appreciated!
Thanks,
-Gerard Saraber
gsara...@rarcoa.com
/agi-bin/cid-to-acct.php': No such file or directory
AGI Tx 200 result=1
I have both #!/usr/binphp and #!/usr/bin/php5 tried out with the same
errortried searching online with no help.
Any ideas?
begin:vcard
fn:Gerard A. Matthew
n:Matthew;Gerard A.
email;internet:[EMAIL PROTECTED]
tel;home
platform...Before every hangup, the
account balance is sent to the user. Hope I'm clear on this.
Rgds,
Gerard.
begin:vcard
fn:Gerard A. Matthew
n:Matthew;Gerard A.
email;internet:[EMAIL PROTECTED]
tel;home:1 (206) 203-7608
tel;cell:1 (940) 337-3739
note;quoted-printable:DM Tel: +1 (767) 440-3940
Are your phones behind NAT?
This should be an issue with rtp port communication.
Gerard.
--Original Message--
From: John Koenig
Sender: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
ReplyTo: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Jul 15, 2008 6:47 PM
visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
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Asterisk-Users mailing list
7.4 to 7.5. Have you tried the 7.4 firmware
to see if that does you any good?
For what its worth, 7.4 seems to work great in my setup, I stayed away
from 7.5, luckily I read about the glitching before upgrading.
--
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL
heard that location does matter.
P.S.
My local Cisco reseller wants to sell me technical support agreement which
cost around 75$ for every phone!
--
Tomislav Parcina
tparcina#lama.hr
--
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED
:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
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an
RCA jack?
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Regards,
Gerard Saraber
Network Admin
,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
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:) and
it was surprisingly easy to implement.
--
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
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in repeated itself about 3 times. sorta like
this: ...your call will be answered as *digital sounding
beep*quickl*quickl*quickly as possible
Next up I'm going to try a different mainboard with only one TDM card in
it.
On Mon, 2006-02-13 at 10:40 +1100, Mike Pollitt wrote:
Hi Gerard --
I found
NMI:143
LOC:1026004
ERR: 0
MIS: 0
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gerard Saraber
Sent: Saturday, 11 February 2006 2:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
:1170531 IO-APIC-level wctdm
225:1169188 IO-APIC-level wctdm
233:1167705 IO-APIC-level wctdm
NMI:195
LOC:1183483
ERR: 0
MIS: 0
--
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED
On Fri, 2006-02-10 at 16:05 -0600, Matthew Fredrickson wrote:
On Feb 10, 2006, at 1:21 PM, Gerard Saraber wrote:
Found it, going to go test it right now :) thanks!
So far reports have been positive on the echo, but its a slow day ;)
We're using cisco 7960 phones, they're pricy
So nobody heard these before? or did I do something stupid that anyone
should know and nobody wanted to yell at me for it ;)
On Wed, 2006-02-08 at 12:54 -0600, Gerard Saraber wrote:
Hi,
I've got some weird sound artifacts happening during calls, they're very
hard to describe, so I have a 122kb
On Fri, 2006-02-10 at 11:01 -0600, Clint Sharp wrote:
Gerard Saraber wrote:
Thanks! testing it now, on my test calls it appears to start out with
less echo then the Mark3 canceler, but it trains slower, seems like it
took a long time for the echo to completely disappear, the real test
225:2769262 IO-APIC-level wctdm
--
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
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card on the list of possibilities,
Thanks,
Gerard Saraber
[EMAIL PROTECTED]
On Wed, 2006-02-08 at 17:26 -0800, Canuck15 wrote:
Gerard,
Just get yourself a Sangoma card with hardware echo can and be done with it.
It is worth every penny just for the headaches it will save you. It's
specific information on why the module format is 'wrong' .
I would suggest after checking the /usr/src/linux symlink, to recompile
the kernel, the ztdummy module and booting into the newly compiled
kernel. its possible that all it takes is to recompile the module
though.
Regards,
Gerard Saraber
[EMAIL
of the call. (which I
can live with, but some of the calls are completely unusable due to
continuous or returning echos)
I'll go play with the mg2 and kb1 again and see what happens
--
Thanks,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED
14:47:51 [kernel] Zaptel Version: SVN-trunk-r934M Echo Canceller:
MG2
--
Thanks again,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
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turned off (2.6.16-rc2) doesn't appear to make any difference either.
I'm not sure what else to try, any input would be appreciated.
Thanks,
Gerard Saraber
[EMAIL PROTECTED]
hardware:
AMD64 1.8Ghz 512M ram
MSI nforce3 socket 754 mainboard
3 Digium TDM400P cards, 10 FXO + 2 FXS modules
/proc
cards have decent resale
value ;)
--
Thanks,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
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software, but if not we'll be going the hardware canceller
route.
Thanks,
Gerard Saraber
[EMAIL PROTECTED]
On Wed, 2006-02-08 at 11:45 -0800, Michael Collins wrote:
Gerard,
I'll bet your side is working great for echo cancellation. It sounds
like the equipment at the other end of the call might
Are you using wav or wav49? You can check in
/etc/asterisk/voicemail.conf under the format option... wav49 creates
much smaller files than normal wav and doesn't need a special player
like gsm files would and as far as using mp3, I'm not sure how to go
about that.
-Gerard
Ryan Pagquil wrote
This has come up a bunch of times on this list..
Take a look at http://www.voip-info.org/wiki-Asterisk+sound+files
Hope that helps
-Gerard
Mark Quitoriano wrote:
how can you play .gsm files what program can you use both in windows and
linux system
I get the same thing too.. Happens quite often for me. Its just
something I have come to live with with voipjet..
-Gerard
Garth Summey wrote:
Don't think there is anything wrong with your setup. We get the same
thing... Maybe they're down, but I would like a third opinion...
G
Michaël
Neither did I.. So I called digium this afternoon and they said they
would have someone look at it..
-Gerard
Huddleston, Robert wrote:
Is it my imagination or did I just drop off the list for several days
somehow... I didn't get any posts since Friday
I am having the same issues.
Regards,
GM
On 4/25/05, Jerry Geis [EMAIL PROTECTED] wrote:
I am having the same broadvoice issue at the moment.
jerry
Is anyone else having difficulty with their Broadvoice service? When I
dial my number right now it rings either fast busy or tells me
(still working out
kinks)...
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gerard
Marcel
Sent: Monday, April 25, 2005 1:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] voip problems
How do you guys deal with voip
How many gateways does broadvoice have? Does anyone know? I know
about sip.broadvoice.com. Are there other ones?
TIA,
GM
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To
Does anyone here know of any general, good voip mailing list? I am
having a hard time with broadvoice and the company is not answering
its phone.
TIA,
GM
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What are the best companies to use with an asterisk PBX box? We are a
linux shop and we want to implement VOIP using asterisk. I would like
to hear pros and cons about each company.
Thanks,
GM
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Timestamp for separate DTMF events.
Rgs,
Gerard.
User Datagram Protocol, Src Port: 6934 (6934), Dst Port: 1686 (1686)
Real-Time Transport Protocol
10.. = Version: RFC 1889 Version (2)
..0. = Padding: False
...0 = Extension: False
= Contributing source
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