RE: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk

[Alex Ternero]

Title: Linksys SPA-841 Disconnects from Asterisk








I don t have problems, after upgrade the
firmware to the latest version.

 

Alex

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow
Sent: Thursday, November 24, 2005
3:49 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Linksys
SPA-841 Disconnects from Asterisk



 

Hi
all, I wonder if anyone out there has experienced an issue I am having with my
Sipura / Linksys SPA-841 phones. 

They
work fine generally, but occasionally, incoming calls are missed.  It's
like the SIP registration is expiring.  Does anyone know how to force the
phone to re-register automatically?  

 

David
A. Morrow 
Technical
Systems Lead 
Autodata
Solutions Company 
[EMAIL PROTECTED]

http://www.autodata.net


* PLEASE NOTE THAT EFFECTIVE DEC 1,2005
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NEW
!!! Tel: (519) 963-3020 
Fax:
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[Asterisk-Users] FAX pass through

[Alex Ternero]

I would like to receive/send faxes, from a SIP gateway. Asterisk can do that
???.

A phone can receive the fax call, then transfer to the SIP gateway, where a
FAX machine is connected.

We have problem, but I don t know is the problem is the gateway (Addpac)???


Thanks

Alex Ternero


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RE: [Asterisk-Users] sipuras 841 bad sound

[Alex Ternero]

Upgrade the Firmware.

Alex

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Juan Jose
Comellas
Enviado el: martes, 20 de septiembre de 2005 23:23
Para: asterisk-users@lists.digium.com
Asunto: Re: [Asterisk-Users] sipuras 841 bad sound


Have you tried upgrading the firmware? I had several problems with the
outbound volume of these phones until I upgraded them.


On Tuesday 20 September 2005 20:46, Anton Krall wrote:
> Hi Guys!
>
> I have a problems with some sipuras 841 and asterisk 1.0.9.
>
> Im using 841 with asterisk 1.0.9 with a digium card (single e1 span) with
> steve's unicall.
>
> Everything compiled fine and in fact I can make and receive calls but I
> have a problem with bad sound when the sipuras call the outside E1's
lines.
> I can listen to the caller without problems but they heard me with a
choppy
> sound as if you were losing frames or cutting off. Calls between internal
> sipuras sound good (eventhough the speaker and headset sound comes and
> goes, for example, when you start talking, seems as if the sipuras takes a
> few seconds to catch up with you on volume so the remote user listen to
you
> as if the first words and the last were at low volume and the conversation
> in the middle sound good, any had that problem?)
>
> So, internal calls sound good between 841's but sound volume is weird at
> the start and end of a sentence.
> Calling the outside lines via E1's, I can listen to people without
problems
> but they heard me as choppy or cut off.
>
> Anybody had issues like this? Is it asterisk or the phones or what?
>
> Hope you can help Guys, Im really banging my head against the wall here.
>
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--
Juan Jose Comellas
([EMAIL PROTECTED])

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[Asterisk-Users] HELP PANASONIC TDA200 AND ASTERISK

[Alex Ternero]

 
Someone  get a 
connection bewteen Panasonic E1 Pri Card and Asterisk Digium E1 Card TE 
110P
 
Please I need 
HELP..
 
 
Alex
 
 



  
  

  


  

  
  

  


  
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RE: [Asterisk-Users] How many TDM22P Card can be used on the same PC ?

[Alex Ternero]

I have a system with 12 FXs , 8 FXO, and E1 Card working very well.

Alex

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Jonathan k.
Creasy
Enviado el: jueves, 18 de agosto de 2005 11:59
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] How many TDM22P Card can be used on the
same PC ?


No

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Wednesday, August 17, 2005 7:38 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How many TDM22P Card can be used on the same
PC ?


 Is it possible to use 24 FXS/FXO lines(on 6 PCI slots) at the same time
on the
same PC? 

I wonder for sound quality and power issues. Can anyone convince me that
I
can(not) use 6 TDM22P cards?

Thanks in advance.
BDM.



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RE: [Asterisk-Users] Polycom IP301 and 501 with asterisk...

[Alex Ternero]

Hi

I work with Sipura 841, and works very well, after make a firmware update.


Alex

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Trevor
Peirce
Enviado el: jueves, 11 de agosto de 2005 18:56
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] Polycom IP301 and 501 with asterisk...


Ing. Marlo R. Beltran G wrote:

> Hi,
>
>  
>
> I am about to buy ip pbx asterisk system but what ip phones do you 
> recommend? Are polycom ip all functional with the ip pbx system???
>
>  
>
We just got a Polycom IP501 for testing and have thus far been 
unsuccessful at getting it to regiser with asterisk.  Outgoing cals work 
fine now (with authentication; verified with ethereal).


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RE: [Asterisk-Users] Re: Panasonic KX-TD500

[Alex Ternero]

Could ypu send me the zaptel.conf and zapata.conf files?


Alex

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Nenad
Radosavljevic
Enviado el: miércoles, 10 de agosto de 2005 15:26
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] Re: Panasonic KX-TD500


>
> Hi
>
> I have a connection with Panasonic TDA200.
>
> 1. For echo,,, echocancel=yes, must be before the channel definition of
> the
> PRI.

I'm very glad if that worked for you, but in case of D500 (connected as
described in previus messages in this thread), it seems that there is no
solution for echo problem.
I was even in contact with Digium support and after testing, we came to
conclusion that calls that are entering Panasonic over a PRI and got picked
up by any Panasonic Extension and then transfered to Asterisk connected over
second PRI, are seen by Asterisk as analogue calls (no matter they entered
Panasonic over PRI and left Panasonic over second PRI and got to Asterisk
over the PRI :) ). This type of calls, and by my now 6 months expirience
with asterisk-panasonic D500 hybrid, ONLY this type of calls, always have a
noticable echo (tested with all different TX and RX leves, all kind of
echo-cancellation compiled in zaptel drivers, all kind of echocancel,
echocancelwhenbridged and echotraining settings).

So far no solution for that (except I'm trying to get external hardware
1xE1/PRI  echo-canceller, to put it in between asterisk and D 500, to see
will it solve problem :( ).

Regards,
Nenad




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RE: [Asterisk-Users] Re: Panasonic KX-TD500

[Alex Ternero]

Hi

I have a connection with Panasonic TDA200.

1. For echo,,, echocancel=yes, must be before the channel definition of the
PRI.

One question

>From Panasonic extensions can call to SIP Phones


Alex

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Guy C.
Guckenberger
Enviado el: martes, 19 de julio de 2005 18:45
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] Re: Panasonic KX-TD500


This is good info.  Has anyone else had this success? Can anyone address
the ECHO issue?


Thanks


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nenad
Radosavljevic
Sent: Tuesday, July 19, 2005 1:58 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Panasonic KX-TD500


We have Panasonic D500 and Asterisk with TE110P hybrid setup
successfully 
and it is possible to route DID to a PRI card in D500 if it is set up as
EXT 
card, and extension "number" of PRI card is defined for example as 2XX.
This 
gives you opportunity to have extensions 200 - 299 routed to PRI
interface 
in Panasonic D500 and after that setting up DID routing is easy.

On the other hand, in that configuration we have an echo problem with
the 
calls that entered Panasonic through the first PRI card connected to
Telco, 
and that got picked up by any EXT of Panasonic, and then got transferred
to 
the Asterisk extensions which are SIP phones (over that second PRI
interface 
in D500 connected to a TE110P).

Also we have a problem with SpanDSP as a FAX in that configuration (call
to 
a FAX DID on Asterisk ends after a 3-4 seconds).

Both this issues don't happen when the Asterisk is connected directly to
a 
Telco.

Regards,
Nenad



> On Mon, 18 Jul 2005, Guy C. Guckenberger wrote:
>
>> Anyone have any luck with connecting Asterisk to the Panasonic
KX-TD500.
>> I have Asterisk connected via crossover to the TE110P. We are able to
>> make internal calls into the Asterisk Box but the PBX vendor (I know
>> nothing about the KX-TD500) tells us it is not possible route DID
over
>> the trunk. I find this hard to believe.  Anyone have any luck with
this?
>
> It depends on what T1/E1 card you have in the Panasonic, I think. It
> most certainly depends on what you are trying to accomplish.
>
> On the 500 it _is_ possible to set up a dialplan that will route some
> extensions out over a PRI link. That is one of the qualitative
differences
> between the KX-TD1232/KX-TD816 and the KX-TD500.
>
> So, what kind of trunk card do you have in the Panasonic and what do
you
> expect it to do?
>
> Peter



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[Asterisk-Users] Connection Asterisk- Panasonic TDA200

[Alex Ternero]

Someone can help me 
in the configuration between Panasonic TDA200 and Asterisk via TE110P 
and Panasonic E1-PRI Card???
 
I get the 
connection, but the problem is : From Panasonic extensions , cannot make calls 
to SIP Phones???.
 
From SIP I can 
make calls to Panasonic extensions ,and PSTN .
 
 
The connection 
is:
 
Asterisk  TE 
110P>>>Panasonic E1-Panasonic E1--->>>PSTN 
E1 
 
 
 
 
 
 
 
 
Thanks
 
 
Alex 
 
 



  
  

  


  

  
  

  


  
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[Asterisk-Users] Panasonic TDA200 E1 -> E100P negotiation issues

[Alex Ternero]

Hi
 
 
I get a Connection 
between Asterisk and E1 PRi Card on TDA200 Panasonic pbx.
 
The connection 
is:
 
Asterisk ---> 
Panasonic TDA200---> Telephony Provider E1 PRI
 
From Ip Phones 
connected :
1. We can make call 
to any extensions
2. We can make call 
to Public phone service, any number, Cellular, intenational 
calls,etc.
 
The problem is 
:
 
From Panasonic canot 
make calls to Ip PHoNE.
 
I execute Pri debug 
span 1.
 
And i dont see any 
connection , when , a call is make from Panasonic 
estensions.
 
The people of 
Panasonic said to mee , they don t have dialtone from asterisk, to make a 
call.
 
Some one get the 
connection works , ASTERISK--PANASONIC TDA200, culd you send me your zapata.conf 
, and zaptel.conf files.
 
 
 
Thanks
 
Alex 
Ternero
 
 
 



  
  

  


  

  
  

  


  
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  Alex Ternero H.Gerente 
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[Asterisk-Users] Channel restarted en E1 Card

[Alex Ternero]

Every hour the 
channel in the E1 is restarted, and all active calls drop.
 
Is there a 
configuration to disable the restart parameters, configured in zapata.conf 
file??
 
There is a parameter 
: resetinterval = 3600 in zapata.conf
 
 
How can i avoid the 
continuos restart of channels??
 
 
Thanks
 
Alex
 
 
 



  
  

  


  

  
  

  


  
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Tecnico 
  Winet PeruAntequera 777 Piso 10 
  

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tel: fax: 
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