In sip.conf, make sure you set nat=yes and canreinvite=no . I had similar
problems, and those two settings made a big difference.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Tue, 8 Jun 2004, Echchelh Zouhair wrote:
Hi,
I have a problem to make call behin
if the problem goes away.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Fri, 4 Jun 2004, Gary Franczyk wrote:
I don't think the question you answered is the same as the one I asked. The
problem with mine is the dropping of all the lines/calls. It resets all
calls, but the dropped calls occur both on PRI and
non-PRI calls.
Has anyone observed these messages, and if so, do you know what they mean
and are they something to be concerned about?
Thanks in advance.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
I think this is worthwhile. I have a simplistic followme implemented in
extensions.conf, but it's not as flexible as your proposal.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Tue, 1 Jun 2004, Brian D'Arcy wrote:
Hello all,
I'm going to tackle
that will tell me who is causing
the call to be destroyed and why?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
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for
outside calling.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Fri, 28 May 2004, Michael George wrote:
I did take a quick look at it, but the header indicated that DISA
allows incoming calls to dial back out. I am just trying to emulate
the feel of our
If you have used * to support a pri as pri_net (as opposed to pri_cpe),
either to talk to another * system or a PBX of some sort, I would be very
interested in hearing about your experiences. Imparticular, I would like
to know that it works before I invest in the extra hardware.
TIA
Bruce
a way to use it to accomplish what I want. I've checked
the archives, etc., but I don't see that anyone has ever done this.
If you have, please respond.
Thanks
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
with the auto-attendant, because it answers right away, but it's kind
of a problem for the other DID numbers. I stuck Ringing statements
every I could think of, but to no avail.
Is there something specific I must do to cause the caller to hear ringing?
Bruce Komito
High Sierra Networks, Inc
I'm running CVS-HEAD-05/06/04-17:52:43 . Do you think the problem is
fixed in this version, or do I have a different problem?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Tue, 25 May 2004, Eric Wieling wrote:
Bruce Komito wrote:
I have a PRI
Tim, I would double check the timing. It seems odd that you would supply
clock rather than the switch, and if you get clock slips, that could
certainly account for what you are seeing. Feel free to contact me
off-list if you need more info or have any questions.
Bruce Komito
High Sierra
I am having exactly the same problem with two phnes connected to a Sipura
behind a Linksys. I'm sure this is NAT, because it works fine when I move
the Sipura out from behind the Linksys.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Mon, 24 May 2004
, as the clock slippage increased to the point of being
out of tolerance. Eventually, the entire PRI had to be restarted. When
we installed the clock, all of the problems disappeared.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Mon, 24 May 2004
I'm not the original poster, but I have the same problem with a Sipura.
In my configuration, I have line 1 set to port 5060 and line 2 set to port
5061. I assume that is what you are suggesting, right?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Mon
John, In my case, the two ports are not using the same IP port (one is on
5060, the other on 5061), but of course, they are on the same IP address.
I think that is what is confusing the NAT server, but I don't know what to
do about it.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
In sip.conf, try setting canreinvite=no for both lines.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Mon, 24 May 2004, Barry Fawthrop wrote:
The problem is probably that both of your SIP phones are using the same
port. I played with two 7960's
Not to beat a dead horse, but I had the problem even with the two lines on
different ports. The canreinvite=no thing solved the problem.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Mon, 24 May 2004, John Fraizer wrote:
Bruce Komito wrote
on, but I had to
turn stun off because it was causing a long, inexplicable delay after
dialing before the call would complete.
I'm realizing NAT with VoIP is a real problem. Anyone have a silver
bullet they wish to share?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800
the Sipura is behind a NAT server, but
the constant stream of warnings from * make me think I'm doing something
wrong. Anyone have any ideas?
Thanks in advance!
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Wed, 19 May 2004, David Creemer wrote:
Hi-
I'm totally stumped configuring my TDM400P with one FXS and one FXO
module. Before I got the FXO module, I used to have an X101P, and
everything was working
running CVS-HEAD-05/06/04-17:52:43.
TIA
sip.conf:
[7752365815]
type=friend
username=5815
secret=
host=dynamic
mailbox=7752365815
context=vpbx-wpti
qualify=1000
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
nat=yes
voicemail.conf:
[vpbx-wpti]
7752365815 = ,Bruce Komito,[EMAIL
to make this work properly, like may in the sip device itself?
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
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, however, work fine, and that would point to the TDM400P.
For what it's worth, I set busycount=12 (had been 4) and still have the
problem, callprogress=no .
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Tue, 18 May 2004, Barton Hodges wrote:
Hi, I had
as over a sip gateway. Calls between sip extensions do not have
this problem.
Has anyone ever experienced this?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
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We are interested, too, but from a practical standpoint. We have IMT
trunks from the LECs, and it would be great to be able to terminate them
with Asterisk. That can't happen without SS7 support, so instead we can
only terminate PRIs, which, of course, requires a switch that supports
SS7.
Bruce
, either at install or run-time.
TIA
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
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At first, I didn't, but I later installed mpg123 and then rebuilt *. That
didn't seem to make any difference, but since I'm not seeing any errors
logged, it's hard to say exactly what's going on.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Fri, 7 May
I don't think that's the case. Mpg321 isn't installed and the mpg123
object I built is called mpg123.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Fri, 7 May 2004, Carlton J. O'Riley wrote:
And be sure you don't have mpg321 linked as mpg123
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