Now that I fully understand how Audiocodes devices work it only takes
about 5 minutes of config to get one going for me because all I have to
do is slightly modify the config file and set the IP settings,
username, and password and I'm done. True the factory doesn't offer
direct support but I've had good luck getting support from a company
called Scansource (bought the Mediant 1000 through Software House
International https://www.shi.com who outsources the support to these
guys) and they have direct contact with Audiocodes. The first year of
support should be included with all Audiocodes devices if bought from a
reputable vendor.
I have found that once an Audiocodes device is configured and set it up
in an environment with a good UPS that you usually never have to touch
them again.
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]
<[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]
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]
<[email protected]>
To: [email protected]<[email protected]>
Cc: [email protected]<[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]
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]
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
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]
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]
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]
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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