Displaying 20 results from an estimated 6000 matches similar to: "PAP2T-na Bricked?"
2011 Apr 25
3
PAP2T auto answer?
Hi all,
Is it possible to send a SIP header to a PAP2T or SPAxxxx and cause the device
to automatically answer? I can do this with my Polycom phones and would like
to do it with my ATA's.
Any ideas?
--
Take care and have fun,
Mike Diehl.
2010 Oct 02
2
Security - Using Linksys PAP2T from outside with a dynamic IP is there anyway to block all other traffic but those of the PAP2T?
Hi Everyone
I think PAP2T supports DynDNS and other Dynamic DNS providers. I have a box
that needs to be secured at all times. Currently it's not connected to the
internet. If it were connected, I would have iptables block any and all
traffic from outside but I want a single device - Linksys PAP2T - to be able
to connect back to the server. That is a stand alone device and doesn't
support
2006 Jun 08
2
Linksys PAP2T-NA - call goes through but phone doesn't ring
I'm trying out a Linksys PAP2T-NA. Calling out works great, no problems
there. Calling in, though, the phone doesn't ring. Caller ID shows up, I
can pick up the phone, and the call is connected, but no ring. I've tried
it on two analog phones, same behavior. Suggestions?
Asterisk SVN-branch-1.2-r31555.
- James Moore
2007 Apr 13
1
PAP2T-NA Jitter Buffer
Hi Folks,
I know the PAP2T-NA has a jitterbuffer.... however, it seems to be adaptive,
which is fine for most situations... however, is there some way I can
either:
A) Specify how long it waits before it starts to shrink?
B) Specifiy a fixed sized jitterbuffer?
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2010 Apr 29
4
ATA shootout: PAP2T versus Grandstream Handytone 286
I'm considering a situation where I buy about twenty ATA devices.
I've played with the Linksys / Cisco PAP2T, and got that working fine
with some inbound and outbound faxing. The web GUI was okay. I'm
seeing prices around $45 to $50 for this thing. It comes with two FXS
ports, but I only need one FXS.
I've seen the Grandstream Handytone 286 online. It looks promising as
an
2011 Jun 13
1
PAP2T provisioning via SRV record?
Hi all,
I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk
server. I'm using a provisioning file that contains an element like:
<Proxy_1_> _sip._udp.example.com </Proxy_1_>
However, the PAP doesn't seem to be able to find my server with this hostname.
The DNS records are in place because my Polycom and Grandstream servers work
just fine.
2009 Feb 13
1
linksys PAP2t and asterisk
Hi all:
when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one.
any suggestions please.
_________________________________________________________________
Windows Live?: E-mail. Chat. Share. Get more ways to connect.
2007 Aug 16
1
Asterisk, PAP2T and 2Wire DSL router
Here is Mexico the phone company uses a DSL router from 2Wire which in
my opinion is quite bad. I am having problems getting PAP2T adapters
connected to Asterisk using these routers. They connect fine but after
about 5 minutes I get a message on the Asterisk console that the ATA is
unreachable. So far the only way I have found for the ATA to stay
connected more than five minutes is to put it in
2007 Jun 07
3
Provisioning Linksys PAP2T ATA's
Does anyone know how the Linksys PAP2T ATA's can be mass provisioned?
Documentation seems to be sketchy, even on the Linksys web site.
Thanks,
Doug.
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2007 Aug 16
2
tone in linksys pap2t
i have the problem in the hardware linksys pap2t, I am install asterisk with
asterisk-gui and work fine but the hangup the phone (linksys pap2 t), no
tone and sound like tu,tu, tu , tu , tu , tu ,
tuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuuu
what is the problem with phone ???
add param special???
Note: i am mark number phone and wait ... sesonds and call.
thank you.
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2006 Apr 10
3
Vertical
Hi all.
I'm in the process of configuring a phone system for my family and friends.
I'm wondering if I should try to implement the "Vertical
Services" (http://www.nanpa.com/number_resource_info/vsc_assign) in the
Asterisk dialplan, or if I should delegate those functions to the various
ATA's.
For example, the Sipura SPA 2002 can handle*69 internally. On the other
2006 Dec 02
1
Linksys PAP2t-NA and Asterisk
I've got a PAP2 that I've got working with asterisk. At the moment, its
configured so that when a phone is picked up on it, it connects to Asterisk.
My hope is that I can let Asteirsk handle the entire dialplan, including
dial tone generation. What would my context in extenstions.conf look like
for this sort of dialing. More accurately, how can I get Asterisk to
generate the dial tone on
2012 Feb 09
0
Turning off splash ring on PAP2T
Hi all,
I'd like to know how I can turn off the "splash ring" voicemail waiting
indication on a PAP2T from the provisioning XML file. I can do it from the web
interface, but I need to do it on "a lot" of machines....
TIA,
--
Take care and have fun,
Mike Diehl.
2012 Jan 26
2
Too many open files
Hi all,
While trying to track down a T.38 issue, I came across a series of log
entries like this:
============================================================================
[Jan 26 10:23:31] WARNING[32508]: udptl.c:948 ast_udptl_new_with_bindaddr:
Unable to allocate socket: Too many open files
[Jan 26 10:23:31] ERROR[32508]: acl.c:488 ast_ouraddrfor: Cannot create
socket
2010 Oct 26
2
No media being sent in SIP call
Hi all,
I seem to be having a strange problem with a sip trunk.
On a fairly frequent basis, I'll make a call, ore receive a call, and there
will be NO sound. The strange part is that both endpoints are public IP
addresses so NAT isn't in play and a sniffer trace reveals that the packets
simply aren't being sent.
It only seems to happen on a particular trunk. The same phone
2010 Apr 13
2
All incoming calls landing in [customers] context
Hi all,
I'm trying to tighten things up a bit and I seem be be running into something
that doesn't make sense to me.
I've got 2 contexts, one for customers, and one for guests, that I include
into [customers] and [default], in extensions.conf, as below:
=============================================================
[default]
include = dial_GUEST
[customers]
include = parkedcalls
2012 Apr 27
1
No UDPTL ports remaining
Hi all,
Lately, I've been seeing more and more instances where I get a flood of warning
messages like this:
[Apr 26 14:09:50] WARNING[21054] udptl.c: No UDPTL ports remaining
The next thing I know, my server is dropping calls and starting to misbehave.
I use fax via T.38, so I can't just turn udptl off. I could expand the port
range, but I suspect that will just mask the situation.
2009 Jan 20
2
PAP2T provisioning
Anyone have an example XML file for the PAP2T?
Cheers,
j
2009 Mar 13
3
Initial silence during call
Hi all,
I've got a problem where many times, there is silence at the first 1-2
seconds of a call. Then it clears up and it's crystal clear. I've not
put a sniffer on it, yet, but I suspect that the media channel is still
being set up. The server shouldn't be too overloaded. Can anyone give
me some advise on how to solve/mitigate this problem?
Mike.
2004 Dec 15
7
VoIP Termination
Hi all.
I'm looking to change from a standard telephone line to a VoIP phone line at
home. I'm looking for recommendations for VoIP providers that I can use with
Asterisk.
One of the catches is that I often telecommute and sometimes I do some side
business; these practices violate many provider's acceptable use policies.
So, I need a provider who doesn't care how I use the