rictrtp
setting.
-Gerard
On Fri, 2014-01-10 at 15:01 -0600, Matthew Jordan wrote:
> On Fri, Jan 10, 2014 at 9:45 AM, gm1 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 a
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?
>
w, as soon as I Answer() their ringback disappears
and the line goes dead while they wait for our guy to answer the phone.
I may start a separate post about getting ringback to work after Answer();
Thanks for the help by the way.
-Gerard
On 03/01/13 14:34, Leandro Dardini wrote:
>
> 2
From: asterisk-users-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 workarou
nks to
adding "Answer()" 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 fo
just upgraded to asterisk 11.2.1.
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
unce,s,1)
>
> Dependent on carrier and other considerations, you can also "spoof" the
> caller-id. That's a different google-search.
--
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)
--
_
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)
). A penny
> or 2 per minute will keep your someone happy.
>
--
Gerard Saraber
Network Admin.
Rarcoa, Inc
(630) 654-2580 x199
(630) 654-3556 (fax)
(630) 915-4122 (cell)
--
_
-- Bandwidth and Colocation Provided by ht
s the schematic:
customer -> our office ---callforward--> cellphone
so should I call AT&T 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 S
id,
I wouldn't 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 10/06/1
7;forgets' 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&
r with Asterisk anymore,
does 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:/
and
SCCP channel Release: 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
ute
'/var/lib/asterisk/agi-bin/cid-to-acct.php': No such file or directory" 2
== cid-to-acct.php: Failed to execute
'/var/lib/asterisk/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
as on a prepaid 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 Te
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
hes, even jumping from 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,
idth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
>
> BTW, I'm in Croatia (Hrvatska). I 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,
Ge
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> >
> > Asterisk-Users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
he new phonesystem for us :) and
it was surprisingly easy to implement.
--
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
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Asterisk-Users mailing list
To UNSU
isk in a reliable way via an
> > > RCA jack?
> > >
> > >
> > > ___
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > Asterisk-Users mailing list
> > > To UNSUB
such an external audio device into asterisk in a reliable way via an
> RCA jack?
>
>
> ___
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
&g
d:usb3
217:1170531 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
PIC-level wctdm
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
old music) coming 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 P
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
0 IO-APIC-level ohci_hcd:usb2
> > > 209: 0 IO-APIC-level ohci_hcd:usb3
> > > 217:5577811 IO-APIC-level wctdm, wctdm
> > > 225:2769262 IO-APIC-level wctdm
> > >
--
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-
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 comple
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 de
ed on major
196
Feb 9 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]
___
--Bandwidth and Colocation provided by Easynews.com --
went away after a second or two at the beginning 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, Rarc
have more 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,
G
e, I'll put the 104d 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 th
with
some fancy 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 e
pe TDM400P cards have decent resale
value ;)
--
Thanks,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
[EMAIL PROTECTED]
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update opt
e kernel with it all
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,
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
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 Pag
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...
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 F
other that gives me DIDs and 800s (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
How do you guys deal with voip problems? do you have multiple backups
such as land lines, and different voip providers?
Regards,
GM
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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 tel
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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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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e the same 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
= Contribut
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