Tim Johnson wrote:
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
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,
On Fri, 2010-02-12 at 17:23 -0400, Tim Johnson wrote:
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
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
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 and
I am having an odd problem with a linksys pap2 ata and asterisk...
Asterisk won't detect digits from it until I issue a 'sip debug'. As
soon as I turn on sip debugging, everything works perfectly (classic
heisenbug)!
Asterisk is latest Debian 'etch' packaged 1.2.13. sip.conf looks like:
I am having an odd problem with a linksys pap2 ata and asterisk...
Asterisk won't detect digits from it until I issue a 'sip debug'. As
soon as I turn on sip debugging, everything works perfectly (classic
heisenbug)!
Instead of SIP debug, try capturing the traffic with tcpdump etc. on
the
I am having an odd problem with a linksys pap2 ata and asterisk...
Asterisk won't detect digits from it until I issue a 'sip debug'. As
soon as I turn on sip debugging, everything works perfectly (classic
heisenbug)!
Instead of SIP debug, try capturing the traffic with tcpdump etc. on
I am having trouble setting this system up and wonder if some one help me.
Does anyone know what is missing if anything to get 2 phones on my
asterisk home server to be able to call each other.
I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2
extensions 5060/5061, this is
Linksys PAP2 is known to accept a single G729 call, at a time.
Maybe this explains your issue ...
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Perhaps it is defective? does the status show l2 registered? Can you access
the IVR? (#)
On 11/30/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I have a Linksys PAP2 connected to Asterisk. Have one of the FXS ports
working fine. I am unable to get the other to work. Does anybody have
I have a Linksys PAP2 connected to Asterisk. Have one of the FXS ports
working fine. I am unable to get the other to work. Does anybody have an
example configuration to make both work. Both are registering fine but
there's just no dialtone on the non working port.
TIA
Hello,
I nearly forgot about this mailing list! I accidentally bought a vonage
enabled PAP2 to use on my asterisk, however it's locked and I have no
access to the admin password. Anyone unlocked it before? Please send
procedure. Make a cc to my address please. Thanks a million.
Best regards,
It is not easy (by a mile), but can be done.
have a look at http://www.dslreports.com/forum/remark,14450684
M
wrote:
Hello,
I nearly forgot about this mailing list! I accidentally bought a vonage
enabled PAP2 to use on my asterisk, however it's locked and I have no
access to the admin
Hi List,
Can anyone confirm if the Linksys PAP2-UK works with Asterisk. I can get
the device to register with my Asterisk box ( v1.2.12.1 ) but I don't get a
dial tone. I have no firewall on my asterisk box and all my other IP
phones work ok.
Thanks in advance.
Phil.
Lists, Hi! good day, i have been task to install/replace our legacy pbx with asterisk, most of the people here in our office was amaze on how asterisk really works, except for i'm having a problem with dtmf detection on PAP2 ATA converterhere is the call flowATA - SIP - TDM400 - PSTN
Nabeel Jafferali wrote:
I don't belive there is a way to turn it off, but you can prevent the IVR
menu being used to factory reset the device using the provisioning tools.
Would you happen to have any more specifics on this? Perhaps an example
of the reset option being disabled? Can any of
Does anyone know of a way to disable access to the TUI interface
(accessed via ) on the PAP2 devices? I'm looking at using these
devices for lobby and door phones and would like to remove/disable the
TUI interface if at all possible.
--
Jamin W. Collins
AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] PAP2 TUI Configuration Menu
Does anyone know of a way to disable access to the TUI
interface (accessed via ) on the PAP2 devices? I'm
looking at using these devices for lobby and door phones
Hello Folks,
I'm an Asterisk newbie, that being said I have managed to get an
SPA941 working with 1.2.8. I've got some issues (like getting the
voicemail button to work as it should, and making the message
indicator light work) but overall I'm pretty happy.
I'm now trying to get a PAP2-NA to
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] pap2 bridging problems
Well NAT is set to yes, but DMZ etc isn't an option for this pap2 since
it needs to be able to roam around different networks. What other
options are there?
William Piper wrote:
Sounds like
I'm having a real problem with one of my linksys pap2. On outgoing
calls the callee will ring, but caller (pap2) will not here it ring
When the callee answers, no audio is transmitted either way. Asterisk
reports the call connected and bridged correctly.
Now the kicker is that sometimes it
it to work.
bp
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Miles Scruggs
Sent: Thursday, May 25, 2006 11:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] pap2 bridging problems
I'm having a real problem with one
Sent: Thursday, May 25, 2006 11:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] pap2 bridging problems
I'm having a real problem with one of my linksys pap2. On outgoing
calls the callee will ring, but caller (pap2) will not here it ring
When
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Miles Scruggs
Sent: Thursday, May 25, 2006 11:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] pap2 bridging problems
Well NAT is set to yes, but DMZ etc isn't an option
Hi All,
I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd
PAP2-NA units all hooked up to Asterisk. As you can imagine, setting
them up took a while, and changing settings on them also takes a
while. In order to prepare for future deployments, I'd like to use XML
provisioning
Gonzalo Servat wrote:
I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd
PAP2-NA units all hooked up to Asterisk. As you can imagine, setting
them up took a while, and changing settings on them also takes a
while. In order to prepare for future deployments, I'd like to use
Ed wrote:
you can try spaconf utility. it can backup/restore config for sipura
devices, for linksys we need small patch.
oops... this patch is included already ;)
___
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when i use the pound key . there is no pause .so that means that the pap2 box is waiting for aditional key is it ? how do we fix this ?thanksGiridhar BandiOn 3/10/06,
indsat [EMAIL PROTECTED] wrote:
Here's a site that will help you with PAP2 Dial
2006/3/7, Giridhar Bandi [EMAIL PROTECTED]:
and when i place a call to local sip extension there is a long pause ( 15
sec )
before the call gets dialled
Use the # key as enter.
--
Alejandro Vargas
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7 de
março de 2006 14:47
Para:
asterisk-users@lists.digium.com
Assunto: [Asterisk-Users] pap2
Dial plan
Hi
i am using pap2 phone adaptors as clients to connect to asterisk server
i am able to make calls but i cannot access voice mail using phone
or start recording while call is in progress
Bandi
Enviada em: quinta-feira, 9 de
março de 2006 13:06
Para: Asterisk
Users Mailing List - Non-Commercial Discussion
Assunto: Re: [Asterisk-Users] pap2
Dial plan
Hi
thanks for the help .
vocie mail problem has been fixed
but the delay is still there i have changed Interdigit Long Timer =2
Here's a site that will help you with PAP2 Dial Plans
http://www.netphonedirectory.com/pap2_dialplan.htm
-- Original Message --
From: Filipe Mordhorst [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
]
[mailto:[EMAIL PROTECTED] On Behalf Of Giridhar
BandiSent: Tuesday, March 07, 2006 12:47 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] pap2 Dial
plan
Hi i am using pap2 phone adaptors as clients to connect to
asterisk server i am able to make calls but i cannot access
Giridhar Bandi
Enviada em: terça-feira, 7 de
março de 2006 14:47
Para:
asterisk-users@lists.digium.com
Assunto: [Asterisk-Users] pap2
Dial plan
Hi
i am using pap2 phone adaptors as clients to connect to asterisk server
i am able to make calls but i cannot access voice mail using phone
I have a Linksys PAP2. Identical setups for the two channels in both
the unit and in Asterisk. In particular, both channels enable g729 and
set it as the preferred codec, and have disallow=all and allow=g729 in
sip.conf.
If we make a call on one channel, it works (and uses g729), but if we
The PAP2 can only handle one g729 call at one time. Whether that's a
hardware limitation, or licensing, or both, I don't know.
Joseph Tanner
On 3/8/06, Warren Burstein [EMAIL PROTECTED] wrote:
I have a Linksys PAP2. Identical setups for the two channels in both
the unit and in Asterisk. In
This ATA can only do 1 g729 call at a time. The sipura 2002 is the
same way. It's outlined in the datasheet.
On 3/8/06, Warren Burstein [EMAIL PROTECTED] wrote:
I have a Linksys PAP2. Identical setups for the two channels in both
the unit and in Asterisk. In particular, both channels enable
On Thu March 9 2006 03:43, Warren Burstein [EMAIL PROTECTED] wrote:
I have a Linksys PAP2. Identical setups for the two channels in both
the unit and in Asterisk. In particular, both channels enable g729 and
set it as the preferred codec, and have disallow=all and allow=g729 in
sip.conf.
Warren Burstein wrote:
I have a Linksys PAP2. Identical setups for the two channels in both
the unit and in Asterisk. In particular, both channels enable g729
and set it as the preferred codec, and have disallow=all and
allow=g729 in sip.conf.
If we make a call on one channel, it works
Joseph Tanner wrote:
The PAP2 can only handle one g729 call at one time. Whether that's a
hardware limitation, or licensing, or both, I don't know.
Joseph Tanner
Hardware. The PAP2 (and SPA2000) can only do one g729 call at a time.
Any other call will have to use g711.
--
Kristian
Actually its hardware related.
On 3/8/06, Nick Hoffman [EMAIL PROTECTED] wrote:
On Thu March 9 2006 03:43, Warren Burstein [EMAIL PROTECTED] wrote:
I have a Linksys PAP2. Identical setups for the two channels in both
the unit and in Asterisk. In particular, both channels enable g729 and
Hi i am using pap2 phone adaptors as clients to connect to asterisk server i am able to make calls but i cannot access voice mail using phone or start recording while call is in progress and when i place a call to local sip extension there is a long pause ( 15 sec )
before the call gets dialled i
Hi,
I have a problem with double ringback tone - outgoing connections to
PSTN. I do not use 'r' option in Dial function so I expect to hear
'real' sounds from pstn provider. But PAP2 generates extra ringback tone
itself! How to get rid of that?
Regards,
L
Hi list,
I have one E1 linK, PSTN, working very well, no sound problems. I was
testing in laboratory some ATAs PAP2-NA from Linksys, at the same
Asterisk's network, but I'm having one problem.
1. When I call another SIP phone everything goes very well and the
quality of sound is perfect in both
Has anyone had success using the SRV functionality in Linksys PAP2-NA's?
Regards,
David
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David Thomas wrote:
Has anyone had success using the SRV functionality in Linksys PAP2-NA's?
I have with the SIPura SPA-3000. Cisco/Linksys licensed the PAP2
hardware and firmware from SIPura. Then they bought SIPura. 8-)
___
--Bandwidth and
-Users] PAP2 and ringing issues
Hi,
I currently have several PAP2-NA units configured to an Asterisk
box, everything works fine except from the fact that after dialing a
number I can hear ringing tones. When I connect to the same
Asterisk box
using XLite or EyeBeam I hear only one, any ideas
Hi,
I currently have several PAP2-NA units configured to an Asterisk
box, everything works fine except from the fact that after dialing a
number I can hear ringing tones. When I connect to the same Asterisk box
using XLite or EyeBeam I hear only one, any ideas on what may be wrong
on the
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Humberto Aicardi
Sent: 01 November 2005 17:17
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] PAP2 and ringing issues
Hi,
I currently have several PAP2-NA units configured
Hello,
We use Asterisk with PAP2 and today we connected the FXS ports of PAP2
to CO ports of our Panasonic KX-TD1232. Problem is that Panasonic
doesn't ring - that is doesn't ring every time the PAP2 is ringing.
When we reset either Asterisk or the PAP2 it usually rings, but after
couple of
Hello, I need to know if there is an option in the PAP2-NA Web Configurator
like Enable IP dialing: yes/no
I need to make point to point calls with two PAP2-NA by IP address (The
PAP2-NA are in the same LAN, no Internet access). Is it possible ?
Thank you !!
At 11:40 AM 3/11/2005, you wrote:
the only way I found to do this
was have them register with a * server and have * connect them
Hello, I need to know if there is an option in the PAP2-NA Web Configurator
like Enable IP dialing: yes/no
I need to make point to point calls with two PAP2-NA by IP
Didn't try it, but quick Google search for sipura IP dialing gives:
http://www.sipura.com/Documents/faq/Section_2.html
3: How do I call by IP address?
A: This example illustrate calling via IP address from Line1 to Line2,
but can be generalized from one SPA to another SPA
- Go to line 1,
Yes, I saw it in Sipura website...but, the question is...does it work or
apply with Linksys PAP2-NA ?
-- Mensaje original --
Didn't try it, but quick Google search for sipura IP dialing gives:
http://www.sipura.com/Documents/faq/Section_2.html
3: How do I call by IP address?
A: This
Has anyone tried loading the PAP2-NA firmware on the PAP2 Vonage model?
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