Has there ever been an effort to configure sipx so it can configure the
Patton devices? I'm sure they are easy to configure once you have done
it once, but I was told on here the learning curve can be steep. That is
why I chose the audiocodes for my first gateway i had to do. I had it
done in 5 minutes without having to know everything about how it worked.
On 2/18/2010 6:38 PM, Tony Graziano wrote:
Nope. I like to have something that senses disconnects properly. Is
factory supported. That is highly configurable. Patton's that ticket
for me.
Imagine a text config file where you simple replace one or two pieces
of information everytime you add one, and paste the config in. It's a
wonderful life. I never spend more than 5 minutes when I get one in
from unboxing to deploying.
On Thu, Feb 18, 2010 at 6:39 PM, [email protected]
<mailto:[email protected]> <[email protected]
<mailto:[email protected]>> wrote:
It is possible. They have a million options, I know that much.
Pressing pound does force the call to dial immediately.
I know. It isn't a great device. For the price, and for what I
need it to do, it is worth a try. I have 2 audiocodes MP 114s on
the way too, but I know you aren't a big fan of audiocodes either.
On 2/18/2010 4:43 PM, Tony Graziano wrote:
Methinks these devices suck rottens eggs.
In any case, what happens if you end the dialed number with
its dial string
termination character (#?)? Does that speed it up?
When I need a cheap device I cringe, because the time ain't
worth the
troubles.
============================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: [email protected]
<mailto:[email protected]>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
----- Original Message -----
From: [email protected]
<mailto:[email protected]>
<[email protected]
<mailto:[email protected]>>
To: [email protected]
<mailto:[email protected]><[email protected]
<mailto:[email protected]>>
Cc: [email protected]
<mailto:[email protected]><[email protected]
<mailto:[email protected]>>
Sent: Thu Feb 18 17:38:33 2010
Subject: Re: [sipx-users] spa3102 for outbound calls
Maybe something related to the 2 stage dialing config? I
didn't notice any
delays like this using the config I sent in that PDF but I was
just thrilled
it could make calls at all and might just not have noticed the
delay. Maybe
plug in a butt-set or a parallel phone and listen for where
the delay is to
narrow it down (delay seizing line, delay before dialing,
delay or slow
dialing of digits, ...?).
-Eric
On Feb 18, 2010, at 4:33 PM, [email protected]
<mailto:[email protected]> wrote:
I have everything working except what I assume is a
dialing rule problem.
As soon as I hit send on the Ploycom, I do see the call
transferred to the
IP of the SPA.
If I have dialed a 11 digit number (1 + area code +
xxx-xxxx) the call
rings immediately.
If I dial a 10 digit number (area code + xxx-xxxx) it
starts ringing in
about 6 seconds.
If I dial a 7 digit number, the call doesn't start ringing
for 10 seconds.
Nothing I have done with the dialing rule seems to change
anything. I'm
assuming the PSTN Line is the place I need to change this.
Interdigit
Short Timer defaults to 5 and Interdigit Short Timer:
defaults to 10.
After reading what they do, I thought that had to be it
for sure.
http://www.bhatt.id.au/blog/fix-sipura-voip-ata-adapter-interdigit-dialing-delay/
I tried lowering those. It didn't seem to affect anything.
I'm assuming
that as soon as it shows the IP on the polycom, the call
has been
transferred to the SPA, so the change I need to make would
have to be in
the SPA. Any ideas?
On 2/18/2010 11:55 AM, Eric Varsanyi wrote:
I started with an Audiocodes gateway back in October,
it was the one
model (FXO+FXS) that sipxecs wouldn't configure and
the sipxecs
configuration stuff for FXO required it to be treated
as a homogenous
group of ports. Two things led me to return it:
1) The documentation and manual configuration of
the SPA3102 is pretty
good compared to Audiocodes (there were numerous
occasions when changing
what appeared to be a completely unrelated setting
resulted in no
dialtone on the FXS side, I think they just internally
bail if anything
is amiss and give you no diagnostics).
2) On a brand new unit they wanted me to buy a
service contract to get
the current firmware and download the manuals (such as
they are)
The SPA may be a buggy POS but Audiocodes was at least
as frustrating to
configure and, as a bonus, it was expensive too.
I expect someone using a model supported by sipXecs
for configuration
would have a better experience.
I feel your pain, the SPA sure is a PITA to get going.
Happy to help if I
can, all those hours spent beating my head on the damn
thing might as
well go to some good :)
-Eric Varsanyi
On Feb 18, 2010, at 11:46 AM,
[email protected]
<mailto:[email protected]> wrote:
This ebay auction is starting to look tempting :)
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=350265531020&rvr_id=&crlp=1_263602_263622&UA=%3F*I7&GUID=d74220781250a0e203017dc1fbe784da&itemid=350265531020&ff4=263602_263622
<http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=350265531020&rvr_id=&crlp=1_263602_263622&UA=%3F*I7&GUID=d74220781250a0e203017dc1fbe784da&itemid=350265531020&ff4=263602_263622>
Audio Codes MP-114 FXO VOIP Gateway - NEW
US $249.99
On 2/18/2010 11:17 AM, Eric Varsanyi wrote:
For debugging if you set it up to send syslog
messages and turn the
level all the way up it sometimes produces
semi-useful output. You
don't have to have a syslog server set up to
catch it if you can run
tcpdump or socat.
If you can capture traffic to/from the device
with tcpdump that's
probably the next step if the syslog stuff
doesn't pay off (it kind of
sounds like either its ignoring you or
sipxproxy isn't really sending
the invite where you hope its going).
-Eric Varsanyi
On Feb 18, 2010, at 11:09 AM,
[email protected]
<mailto:[email protected]> wrote:
I have no doubt it is a PEBKAC, I just
don't where to look. I changed
it to 5061 (I now see that setting in the
PSTN Line tab on the
spa3102). The logs look about the same to
me. I don't see anything
that even tells me it is making it to the
spa3102.
On 2/18/2010 10:54 AM, Eric Varsanyi wrote:
When I set mine up late last year the
only issue I had making
outbound calls (that wasn't PEBKAC)
was the thing didn't think there
was a line attached and returned
something like 'resource not
avaiable' to the invite. I had to
change the line voltage threshold
down in the international settings box
to fix this.
Ah, in the log I see you're using
5060, the FXO side by default is on
5061 (the FXS is on 5060). LIkely
that's your issue.
-Eric
On Feb 18, 2010, at 10:42 AM,
[email protected]
<mailto:[email protected]> wrote:
I'm trying to configure a spa3102
for outbound calls only. I know
people pull their hair out over
these devices, but I wanted to give
it a shot. My only gateways I've
worked with so far are sipxbridge
and an audiocodes configred from
within sipx, so I haven't really
done too much manual FXO
configuration.
I think I may be missing something
on the sipx end, because I don't
think the call is ever making it
to the spa3102. This is a new setup
and has no other gateways. I added
the spa3102 as an unmanaged
gateway. I enabled all the dialing
plans and added the gateway. I'm
using a polycom 550, Sipx 4.0.4,
bootrom 4.2.1, firmware 3.1.3C
split. I would show a siptrace,
but the merged file doesn't really
have anything in it. The sipx
server is at 10.81.1.5. The spa3102 is
at 10.81.1.6. I tried setting the
gateway in sipx to UDP manually
(that is what the spa3102 defaults
to) and specifying port 5060, but
that didn't seem to change
anything. There are only 2 logs
created,
so I attached those. Is there
something simple I'm missing? I read
through this,
http://sipx-wiki.calivia.com/index.php/HowTo_configure_sipX_to_work_with_FXO/FXS_SIP_Gateways
but I don't see anything that
sticks out at me. The only thig I
thought I might need to do is
something in authrules.xml, but I'm
still
not sure since the text around it refers to FXS and this is
FXO. I sort of
guess there has to be some some sort of authorization for the
spa3102 to
know the sipx call can be sent outbound, but I don't know
where to do this.
Sorry if I'm missing something obvious here. I think the fact
that I got an
audiocodes 8 port working inbound and outbound with no
questions (and
clearly not much knowledge on the subject) is a testament to
how well sipx
is able to configure it!
Thanks,
Matthew
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--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
Fax: 434.984.8431
Email: [email protected] <mailto:[email protected]>
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
Fax: 434.984.8427
Helpdesk Contract Customers:
http://www.myitdepartment.net/gethelp/
Why do mathematicians always confuse Halloween and Christmas?
Because 31 Oct = 25 Dec.
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