A similar issue happens to us.
Make sure that, for inbound AND outbound calls rtp packets are reaching the
other endpoint.
If a NAT device(s) is between the endpoints make sure that the device NATs the
traffic on BOTH ways (inbound AND outbound).
Regards
On Saturday 27 September 2008 23:54:37
On Sun, Sep 28, 2008 at 8:16 AM, Brad [EMAIL PROTECTED] wrote:
does anyone have a sample dialplan for vici dial that does not include any
pri stuff.
I am running exclusively SIP for everything and trying to edit the sample
dialplan and removing anything to do with a pri card is becoming a
On Sun, Sep 28, 2008 at 3:56 AM, Babcock, Michael Alex
[EMAIL PROTECTED]wrote:
can a2 billing work on the same system that directadmin is installed?
should not be a problem
ram
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
I'm using Sipp to load test, but it lost some SIP message when I
increment Call Per Second more than 9.
Regards
Grey Man escribió:
I've used both the Hammer Call Analyzer software and als to the Hammer
XMS system which is a server that they install in your rack to do the
packet captures and
thanks;
i'm messing with freepbx and what not for the time being. and am going
to give that a try. Now, also i'm going to use that asterisk system i
have installed on a dedicated box for roleback... smile
On Sep 27, 2008, at 10:55 PM, ram wrote:
On Sun, Sep 28, 2008 at 3:56 AM, Babcock,
i'm using lylix, does anyone know of a good freepbx mailing list? or
can i use this mailing list for freepbx questions?
mike
On Sep 28, 2008, at 12:00 AM, Babcock, Michael Alex wrote:
thanks;
i'm messing with freepbx and what not for the time being. and am
going to give that a try. Now,
We plan to use asterisk for conferencing. As I understand, it requires
either a separate hardware like x100p clone or ztdummy. What are the
pro cons of x100p vs ztdummy. Any other hardware suggestions for
conferencing? It should be able to handle few simultaneous
conferences.
Thanks
Jim
Steve Underwood wrote:
If I were building a terminal, I'd make mine announce 8000, but accept
8000 or 16000 to try to maximise compatibility. It seems people don't do
that.
Looking at debug output from 1.6 (using a grandstream), it looks like
that is what it does.
Hello and thank you for replyes.
Eric, I looked for it on the mailing list and google and did not find something
relevant to be 100% sure that this feature is not supported.
Some information clare I founded in
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups where
it
no.. it's directly connected to the internet.. it's not an issue of accepting
calls.. see.. the problem is the call gets to the server.. the server tries to
route it..
but as if the dial plan is not there.. it rejects the call because it doesn't
know what to do with it..
for example of my
On Sun, 28 Sep 2008 12:05:06 +0800, Steve Underwood wrote:
There is no error in RFC3551. There is a clear statement that an earlier
RFC did things wrong, due to a typo, and classifying G.722 as an 8000
sample/second codec is, for better or worse, the standard. Its messy and
inconsistent,
This is a better question asked on a Fonality list. Maybe they have a
manual.
Thanks,
Steve Totaro
On Thu, Sep 25, 2008 at 10:21 AM, Tariq .. [EMAIL PROTECTED] wrote:
Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and
trixbox..
i tried to creat an SIP link
FreePBX has an IRC channel I believe, as well as a forum.
I have used ASTCC and ASTPP. Pretty simple, I have never really played with
A2billing.
Thanks,
Steve Totaro
On Sun, Sep 28, 2008 at 4:07 AM, Babcock, Michael Alex
[EMAIL PROTECTED]wrote:
i'm using lylix, does anyone know of a good
On Sun, 28 Sep 2008, Jim Boykin wrote:
We plan to use asterisk for conferencing. As I understand, it requires
either a separate hardware like x100p clone or ztdummy. What are the
pro cons of x100p vs ztdummy. Any other hardware suggestions for
conferencing? It should be able to handle few
On Sat, Sep 27, 2008 at 5:54 PM, Philip Prindeville
[EMAIL PROTECTED] wrote:
I've got the following situation. I'm running Asterisk 1.4.18 on a
firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones
behind it.
I'm peering SIP with a Coppercom switch sitting behind an SBC.
On
Hi all,
This might be a little off topic, but I need some help with this phone and
hopefully someone on this list is able to assist me.
When establishing a conference call I am not able to hang up the call I
connected to my original call. I have tried pressin ghte conference button, but
nothing
Well, things just got a lot more interesting... Adding Monitor() to an
extension ends the one-way voice problem on inbound calls!
So an incoming call gets handled as:
[ctc-incoming]
exten = 208345,1,Noop()
exten = 208345,n,Log(NOTICE: RDNIS: ${CALLERID(rdnis)} ANI:
${CALLERID(ani)})
this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk
read the dial plan!!
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date: Sun, 28 Sep 2008 10:00:56 -0400From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]:
On Sun, Sep 28, 2008 at 08:13:08PM +, Tariq .. wrote:
this is not a TrixBOX .. i'm asking a simply question.. why doesn't asterisk
read the dial plan!!
What is the dialplan?
ls -ld /etc/asterisk /etc/asterisk/extensions.conf
And what is the contents of extensions.conf ?
What is the
I'll look into using Record() or Monitor() to capture the phone call,
but if there's any conversion being done by codecs then that won't
eliminate the possibility that the code itself is misconfigured or buggy
and generating a bad stream on one of the legs...
Anyone have an idea about how to
Hello, when with my client X-lite try to register in the server that say me,
Registration error:501 Not implemented.
What isn't implemented? the registration in the sip.conf or extensions.conf?
how can i implemented that?
thanks.
Abel
___
--
Go for it.
ztdummy is not an issue.
I have used ztdummy with 220 simultaneous participants in 18
different conference groups.
At one time, I had 60 machines running simultaneously in a FARM all
of which were carrying
the same 18 conference groups with over 200 participants active on
each
Thanks Gordon Mike for the response.
What accuracy are you getting from zaptest/dahdi_test (and system info).
Two more questions:
1) Does ztdummy requires change into kernel? I am running 2.6.9 kernel.
2) What about CPU load?
Thanks
Jim
On Mon, Sep 29, 2008 at 5:02 AM, Mike Trest [EMAIL
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