I know this is slightly off topic, but I was wondering if anyone can help
with a problem getting my PAP2's to connect to Asterisk. I use a
provisioning file, and I recently re-wrote the files for each PAP2. I had
a small typo and the PAPs logged it as a corrupt file. I corrected the
file, however,
Possibly OT?
I've hooked up a Linksys PAP2 to my Asterisk 1.6 fairly painlessly. The
only issue I can't beat with it is the dial delay when calling internal
or external numbers.
No matter what it seems to take 10 -15 seconds to actually dial. I've
altered the device removing all *xx combos
I have asterisk setup in my house and I have a SPA-3102-NA and a PAP2T. This
is probably just me not understanding what is going on, but I was playing
around last night and I used the sip unregister extension command on the
CLI. I thought the boxes would re-register when their registration
I have asterisk setup in my house and I have a SPA-3102-NA and a PAP2T.
This
is probably just me not understanding what is going on, but I was playing
around last night and I used the sip unregister extension command on the
CLI. I thought the boxes would re-register when their registration
- Original Message -
From: Jeff LaCoursiere j...@jeff.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 21, 2009 9:55 AM
Subject: Re: [asterisk-users] PAP2T provisioning
On Wed, 21 Jan 2009, Stefan Schmidt
Is it possible to allow reinvites to/from specific devices?
For example;
exten 2001 and 2002 can reinvite to each other, but not 2003 and 2004
exten 2003 and 2004 can reinvite to each other, but not 2001 and 2002
Can that be done? Devices 2001 2002 are behind one firewall, and
2003 2004 are
, and the CID should be passed on from the POTS line.
Tim Johnson
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Quoting Jaap Winius [EMAIL PROTECTED]:
Quoting Tim Johnson [EMAIL PROTECTED]:
Your caller ID is probably being over-ridden by the settings in your
sip.conf file. Remove the caller ID from your PSTN section of the
sip.conf, and the CID should be passed on from the POTS line.
That sounds
Quoting Jaap Winius [EMAIL PROTECTED]:
Hi list,
After failing to get a Sipura/Linksys SPA3000, which I've configured
as a PSTN gateway, to pass on the Caller ID, I decided to try my luck
with a Linksys SPA3102 after hearing some promising stories.
Unfortunately, I've run into a completely
Quoting SIP [EMAIL PROTECTED]:
Adam Moffett wrote:
In all seriousness, my requirements were a little silly. A Cisco router
can fail just as a netgear router can. But I think we would find Cisco
failures to be statistically less likely.
I also think we can agree that not all devices of a
Quoting Jaap Winius [EMAIL PROTECTED]:
Hi list,
Hopefully, some of our Dutch members can help with this one. I'm also
based in the Netherlands and am using a Sipura (Linksys) SPA-3000
(firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test
system. It works fine, except that
allow=ulaw
allow=alaw
allow=gsm
Tim Johnson
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(such as 3102) and $SN is the
devices serial number.
If you figure out how to specify settings for the PSTN line, please
share it with the list.
Tim Johnson
Quoting Gopal krishnan [EMAIL PROTECTED]:
Hi,
Try this
http://www.kcip.com/support/pap2uk.html
On Jan 25, 2008 4:18 PM, Rizwan
= _91NXXNXX,1,Dial(trunk type/name/${EXTEN})
or if you're like me and you're used to a cell phone and don't like dialing
the 1:
exten = _NXXNXX,1,Dial(trunk type/name/1${EXTEN})
On 8/7/07, Tim Johnson [EMAIL PROTECTED] wrote:
Hello all. I am just getting back
Hello all. I am just getting back into Asterisk and I am setting up my
Linksys SPA3102. I have incoming calls working fine, as is the phone
plugged into the unit. My problem is I cannot get the SPA3102 to dial
a phone number automatically. I can call the extention of the PSTN and
I get a
, this is
my first attempt at using hardward (and I've only been playing with
Asterisk/softphones for 5-6 days).
Any input would be appreciated.
Tim Johnson
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