Re: [asterisk-users] SPA941 WMI not lighting up when natted

[Mike A. Leonetti]

Michael Leonetti wrote:
> I have an a bunch of SPA941 Linksys phones for users in and out of the  
> office. When the phones are in the office (and on the same network as  
> the asterisk server) the WMI goes on when it should and off when it  
> should. But when the phone is on another network and natted it fails  
> to do so. The light always stays off. Has anybody had a similar  
> problem (and hopefully a resolve)?
>
>   
Never mind.  It was just me being stupid.  I did not define "mailbox="
in sip.conf.  And it's MWI for "Message Waiting Indicator" not WMI.  I
was way off on this one.

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Re: [asterisk-users] SPA941

[Dimitris Counalakis]

Thnx Mark ...
I think you are right, 941 doesn't support TLS at all.

Dimitris ...

M Hulber wrote:
> Unfortunately, I don't have this phone and I can't find any 
> documentation for the 941 that refers to TLS setting.  Here's what it 
> looks like when I set extension 4 to TLS on the 942:
>
> TLS
>
> Dimitris Counalakis wrote:
>   
>> Thnx Mark,
>> but there is no such option for 941 in 5.1.8. As far as I know,
>> 5.1.8 is the lastest I can get for this phone.
>> I also tried to enable TLS with > ua="na">TLS
>> and xxx.xxx.xxx.xxx;transport=tls
>> in the config file(s), but it didn't work.
>>
>> thnx anyway :)
>>
>> Dimitris ...
>>
>> M Hulber wrote:
>>   
>> 
>>> I'm looking at the 942 and if you look under the EXT 'n' settings in the 
>>> 'SIP Settings' section you can select SIP Transport.  There are 3 
>>> options:  UDP / TCP / TLS.  I'm using the 6.1.5 software.
>>>
>>> Dimitris Counalakis wrote:
>>>   
>>> 
>>>   
 Hi all,
 I'm new to this list, so forgive me if I'm not supposed to ask this:
 I currently own a Linksys SPA941 SIP phone with 5.1.8 firmware. Is there
 any way to use TLS with this phone<--->asterisk (v 1.6.0.9)?

 It is said that is supports TLS/SRTP but I don't see any of these
 options in the
 configuration file or the admin (advanced) SIP conf  panel.

 Am I missing something?
 Thnx in advance ...

 Dimitris ...



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>>> 
>>>   
>>
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Re: [asterisk-users] SPA941

[M Hulber]

Unfortunately, I don't have this phone and I can't find any 
documentation for the 941 that refers to TLS setting.  Here's what it 
looks like when I set extension 4 to TLS on the 942:

TLS

Dimitris Counalakis wrote:
> Thnx Mark,
> but there is no such option for 941 in 5.1.8. As far as I know,
> 5.1.8 is the lastest I can get for this phone.
> I also tried to enable TLS with  ua="na">TLS
> and xxx.xxx.xxx.xxx;transport=tls
> in the config file(s), but it didn't work.
>
> thnx anyway :)
>
> Dimitris ...
>
> M Hulber wrote:
>   
>> I'm looking at the 942 and if you look under the EXT 'n' settings in the 
>> 'SIP Settings' section you can select SIP Transport.  There are 3 
>> options:  UDP / TCP / TLS.  I'm using the 6.1.5 software.
>>
>> Dimitris Counalakis wrote:
>>   
>> 
>>> Hi all,
>>> I'm new to this list, so forgive me if I'm not supposed to ask this:
>>> I currently own a Linksys SPA941 SIP phone with 5.1.8 firmware. Is there
>>> any way to use TLS with this phone<--->asterisk (v 1.6.0.9)?
>>>
>>> It is said that is supports TLS/SRTP but I don't see any of these
>>> options in the
>>> configuration file or the admin (advanced) SIP conf  panel.
>>>
>>> Am I missing something?
>>> Thnx in advance ...
>>>
>>> Dimitris ...
>>>
>>>
>>>
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>>>   
>>   
>> 
>
>
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Re: [asterisk-users] SPA941

[Dimitris Counalakis]

Thnx Mark,
but there is no such option for 941 in 5.1.8. As far as I know,
5.1.8 is the lastest I can get for this phone.
I also tried to enable TLS with TLS
and xxx.xxx.xxx.xxx;transport=tls
in the config file(s), but it didn't work.

thnx anyway :)

Dimitris ...

M Hulber wrote:
> I'm looking at the 942 and if you look under the EXT 'n' settings in the 
> 'SIP Settings' section you can select SIP Transport.  There are 3 
> options:  UDP / TCP / TLS.  I'm using the 6.1.5 software.
>
> Dimitris Counalakis wrote:
>   
>> Hi all,
>> I'm new to this list, so forgive me if I'm not supposed to ask this:
>> I currently own a Linksys SPA941 SIP phone with 5.1.8 firmware. Is there
>> any way to use TLS with this phone<--->asterisk (v 1.6.0.9)?
>>
>> It is said that is supports TLS/SRTP but I don't see any of these
>> options in the
>> configuration file or the admin (advanced) SIP conf  panel.
>>
>> Am I missing something?
>> Thnx in advance ...
>>
>> Dimitris ...
>>
>>
>>
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>
>   



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Re: [asterisk-users] SPA941

[M Hulber]

I'm looking at the 942 and if you look under the EXT 'n' settings in the 
'SIP Settings' section you can select SIP Transport.  There are 3 
options:  UDP / TCP / TLS.  I'm using the 6.1.5 software.

Dimitris Counalakis wrote:
> Hi all,
> I'm new to this list, so forgive me if I'm not supposed to ask this:
> I currently own a Linksys SPA941 SIP phone with 5.1.8 firmware. Is there
> any way to use TLS with this phone<--->asterisk (v 1.6.0.9)?
>
> It is said that is supports TLS/SRTP but I don't see any of these
> options in the
> configuration file or the admin (advanced) SIP conf  panel.
>
> Am I missing something?
> Thnx in advance ...
>
> Dimitris ...
>
>
>
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Hulber Technologies
asterisk-ad...@hulber.com

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Re: [asterisk-users] SPA941 -> Asterisk -> Voip provider -> PSTN -> ShoreTel garble

[Cliff Brake]

On 9/22/06, Rich Adamson <[EMAIL PROTECTED]> wrote:

> So, it seems there is some type of weird interaction between my system
> and the ShoreTel system if I use the SPA941 IP phone.
>
> Does anyone have suggestions as to how I can start debugging this?

Check the RTP Packet Size (under the Sip tab). Set it to .020 (20
milliseconds) and place another test call. For whatever reason, the
Linksys/Sipura products default to 30 milliseconds and has impacted the
quality of audio on some systems.


Setting the RTP packet size to 20ms seems to have fixed it.  Thanks
for the suggestion.

Cliff

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Re: [asterisk-users] SPA941 -> Asterisk -> Voip provider -> PSTN -> ShoreTel garble

[Rich Adamson]

Cliff Brake wrote:

I am using the following setup:

Linksys SPA941 -> Asterisk -> NuFone -> PSTN -> ShoreTel system

The system works great for the most part.  Most people I call say it
sounds good.  However, every time I call a certain company that uses a
ShoreTel system, they claim the sound is garbled (understandable, but
not pleasant to listen to).  Everything sounds fine at my end.  If I
make a call w/ the following setup, it sounds fine:

Analog phone -> Asterisk:TDM400 -> NuFone -> PSTN -> ShoreTel system

So, it seems there is some type of weird interaction between my system
and the ShoreTel system if I use the SPA941 IP phone.

Does anyone have suggestions as to how I can start debugging this?


Check the RTP Packet Size (under the Sip tab). Set it to .020 (20 
milliseconds) and place another test call. For whatever reason, the 
Linksys/Sipura products default to 30 milliseconds and has impacted the 
quality of audio on some systems.


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Re: [asterisk-users] spa941 call pickup?

[Rich Adamson]

Rich Adamson wrote:

I've been using *8# on my 7960's to pickup ringing phones in the office.

Anyone been able to do call pickup from a spa941?


Disregard; dumb mistake on my part. Forgot to add pickup to the sip.conf 
definitions for the extension.



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Re: [asterisk-users] spa941 and sip "bye"

[Rich Adamson]

Steve Davies wrote:

On 7/6/06, Rich Adamson <[EMAIL PROTECTED]> wrote:

Been testing a new spa941 with the latest firmware (sip to sip). I
noticed that unlike 7960's, if a user of a 7960 hangs up at the end of a
conversation, the 941 does not automatically hangup. Rather, the 941
sits there for about 5 seconds, then provides a fast busy.  The 941 user
"must" manually disconnect by either hanging up the handset, or,
pressing the speakerphone button (assuming the call was originally
answered via the speakerphone).

I can't seem to find a config option on the 941 to automatically drop
the call when a sip "bye" message is sent to it. Anyone know if that can
be changed?


I have encountered this and as yet found no solution. The phone
acknowledges that the call is over, but stays "off-hook". Weird - I
don't like the SPA phones much, but customers seem to like the
look/feel.


Personally, I actually like the spa941 better then the 7960 (at this 
time, anyway). Pleasantly surprised with it, but probably to early to 
put much into that statement. ;)


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Re: [asterisk-users] spa941 and sip "bye"

[Steve Davies]

On 7/6/06, Rich Adamson <[EMAIL PROTECTED]> wrote:

Been testing a new spa941 with the latest firmware (sip to sip). I
noticed that unlike 7960's, if a user of a 7960 hangs up at the end of a
conversation, the 941 does not automatically hangup. Rather, the 941
sits there for about 5 seconds, then provides a fast busy.  The 941 user
"must" manually disconnect by either hanging up the handset, or,
pressing the speakerphone button (assuming the call was originally
answered via the speakerphone).

I can't seem to find a config option on the 941 to automatically drop
the call when a sip "bye" message is sent to it. Anyone know if that can
be changed?


I have encountered this and as yet found no solution. The phone
acknowledges that the call is over, but stays "off-hook". Weird - I
don't like the SPA phones much, but customers seem to like the
look/feel.

Ho hum.
Steve
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Re: [Asterisk-Users] SPA941 and Echo

[Andres]

Mike Fedyk wrote:


Andres wrote:


Mike Fedyk wrote:

Try reducing the gain on the microphone.  These phones pick up room 
sounds *very* well.



WellI'm not using the speakerphone.  Plus there is no gain 
setting at all that I am aware off.  Just Handset Volume or Speaker 
Volume.


I'm not talking about the speakerphone.

Check the XML config file.


Got it.  Its called Handset Input Gain.   I will play with that to see 
if it helps.  Thanks!


--
Andres


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Re: [Asterisk-Users] SPA941 and Echo

[Mike Fedyk]

Andres wrote:

Mike Fedyk wrote:

Try reducing the gain on the microphone.  These phones pick up room 
sounds *very* well.



WellI'm not using the speakerphone.  Plus there is no gain setting 
at all that I am aware off.  Just Handset Volume or Speaker Volume.

I'm not talking about the speakerphone.

Check the XML config file.
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Re: [Asterisk-Users] SPA941 and Echo

[Andres]

Mike Fedyk wrote:

Try reducing the gain on the microphone.  These phones pick up room 
sounds *very* well.



WellI'm not using the speakerphone.  Plus there is no gain setting 
at all that I am aware off.  Just Handset Volume or Speaker Volume.


Thanks.

--
Andres


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Re: [Asterisk-Users] SPA941 and Echo

[Mike Fedyk]

Try reducing the gain on the microphone.  These phones pick up room 
sounds *very* well.


Andres wrote:
Has anybody else experienced bad echo issues with this SPA941 phone 
when calling SIP-SIP to another SPA ATA?  When I call remote office 
phones that are attached to SPA ATAs, I get very annoying echo.  One 
can sure blame it on the reflected signal from the phone on the remote 
end, but how can one deal with this echo?  (it happens with some 
phones really bad and with others its perfect)


The SPA941 does not handle echo itself so I am at a loss on how to 
approach this issue.  If I change the SPA941 for a regular SPA 1000 
with Echo Cancel enabled, I can hear perfectly.  Any ideas?


Thanks,


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Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.

[Asterisk]

FIXED.
Found and
fixed the problem.
Teach me to
cut code from AAH 2.8
Issue was..
exten =>
s,1,Set(__SIPADDHEADER=Call-Info: answer-after=0)   THIS DOES NOW WORK.
exten =>
s,1,Set(__SIPADDHEADER=Call-Info:\;answer-after=0)  THIS DOES.
 
Strange what
a type can do.  Also strange why the other worked on some other
handsets, but not yours..
 
YAY!
 
James



Hadley Rich wrote:

  On Thursday 04 May 2006 20:53, Asterisk wrote:
  
  
The handsets do not work with the SIP flag to make them AUTO-ANSWER. (As
documented is should)
Ie, you cannot use them with intercom or Page features.

  
  
Works fine here;

SIPAddHeader(Call-Info:\;answer-after=0)

hads

  




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Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.

[Tom Vile]

works for me as well.

On 5/4/06, kevin ling <[EMAIL PROTECTED]> wrote:

Hi,

But it's seems the auto-answer function work on my spa-941. Have you upgrade
to the latest firmware version?

Regards,
kevin

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Friday, May 05, 2006 9:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.



Asterisk wrote:

> Hello all,
> I want to report a BUG with the Linksys SPA94X so it is general
> knowledge and that we can all make noise about it so it will get fixed
> sooner..
>
> The handsets do not work with the SIP flag to make them AUTO-ANSWER.
> (As documented is should)
> Ie, you cannot use them with intercom or Page features.
>
> This works with the Sipura841 fine.  So linksys broke it.  Um..
> interesting is it not, considering it works with there SPA9000 unit...
> sounds a bit fishy to me..
>
> So any Linksys owners using Asterisk, do pass on some discontentment,
> and Email linksys tech support at [EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>
> And tell them you have this issue..
>
>
> James
>
Curious, as I tried to get this to work with the 841, and though the phone
does auto answer, the called or paged party hears dial tone as well as the
page, just as if one went off hook by pressing the speaker button. The Pager
does NOT hear dial tone.
I sent support some information, but so far no help. They asked for more and
I have yet to get back to them

Curious, very curious.

>
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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RE: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.

[kevin ling]

Hi,

But it's seems the auto-answer function work on my spa-941. Have you upgrade
to the latest firmware version?

Regards,
kevin 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Friday, May 05, 2006 9:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.



Asterisk wrote:

> Hello all,
> I want to report a BUG with the Linksys SPA94X so it is general 
> knowledge and that we can all make noise about it so it will get fixed 
> sooner..
>
> The handsets do not work with the SIP flag to make them AUTO-ANSWER. 
> (As documented is should)
> Ie, you cannot use them with intercom or Page features.
>
> This works with the Sipura841 fine.  So linksys broke it.  Um.. 
> interesting is it not, considering it works with there SPA9000 unit...  
> sounds a bit fishy to me..
>
> So any Linksys owners using Asterisk, do pass on some discontentment, 
> and Email linksys tech support at [EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]>
> And tell them you have this issue..
>
>
> James
>
Curious, as I tried to get this to work with the 841, and though the phone
does auto answer, the called or paged party hears dial tone as well as the
page, just as if one went off hook by pressing the speaker button. The Pager
does NOT hear dial tone.
I sent support some information, but so far no help. They asked for more and
I have yet to get back to them

Curious, very curious.

>
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Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.

[John Novack]

Asterisk wrote:


Hello all,
I want to report a BUG with the Linksys SPA94X so it is general 
knowledge and that we can all make noise about it so it will get fixed 
sooner..


The handsets do not work with the SIP flag to make them AUTO-ANSWER. 
(As documented is should)

Ie, you cannot use them with intercom or Page features.

This works with the Sipura841 fine.  So linksys broke it.  Um.. 
interesting is it not, considering it works with there SPA9000 
unit...  sounds a bit fishy to me..


So any Linksys owners using Asterisk, do pass on some discontentment, 
and Email linksys tech support at
[EMAIL PROTECTED] 


And tell them you have this issue..


James

Curious, as I tried to get this to work with the 841, and though the 
phone does auto answer, the called or paged party hears dial tone as 
well as the page, just as if one went off hook by pressing the speaker 
button. The Pager does NOT hear dial tone.
I sent support some information, but so far no help. They asked for more 
and I have yet to get back to them


Curious, very curious.



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Re: [Asterisk-Users] SPA941 SPA942 BUG. auto answer does not work.

[Hadley Rich]

On Thursday 04 May 2006 20:53, Asterisk wrote:
> The handsets do not work with the SIP flag to make them AUTO-ANSWER. (As
> documented is should)
> Ie, you cannot use them with intercom or Page features.

Works fine here;

SIPAddHeader(Call-Info:\;answer-after=0)

hads

-- 
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