[asterisk-users] SNOM 360

[--[ UxBoD ]--]

Hi,

I have got my new phone working with Asterisk, and must say it is very
very good combination. Now I have WMI working, but what I would like to
be able to do is press the DND button on the phone and for all calls to
my extension to be forwarded direct to my voicemail.

How can this be done please ?

TIA

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[asterisk-users] SNOM 360

[Erik Wartusch]

Hi,

I`ve a strange problem since an upgrade from Asterisk 1.2 to 1.4.1. 
If a SNOM 360 calls (internal) another phone type than SNOM (e.g.  Linksys, 
Thomson we have here), there is no audio transmission anymore.
I`ve upgraded the phone to the latest firmware with auto upgrade. No results.
With 1.2 it was working well.
Any suggestions?

Thanks

Erik
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[Asterisk-Users] Snom 360

[Colin E. McDonald]

I am having major problems with the first run of Snom 360s that rolled
out last month.
I am working with the US vendor and they in turn are working with Snom
but I wanted to see of anyone else was using these or having issues with
them.

Issues:

Speakerphone/Hands Free volume spikes up and down during a call. You
have to manually set the volume during every call. This makes it totally
unusable. The sound will cut out completely at the beginning of a call
sporadically. 

Call comes through speaker phone after you pick up handset and then cuts
to handset a couple of seconds later

There is a mnaufacturing defect where the display cable is disconnected
so you get what appears to be DOA desk sets. You can pop the case and
reconnect the cable or send them back.(This is known and workaround from
Snom)

Have to press the retrieve message button twice pretty regularly to get
it to dial vociemail (*97) in asterisk

Major problem with calls being dropped when you place callers on hold
after you set the busydetect=yes and busycount option in zapata.conf (If
I don't set these then my lines get used up and a soft hangup or reload
has to be executed).
I found a workaround (set the disconnect on hook for hold option to off
in Advanced Settings) but then you manually have to kill all of you
calls once they are completed.

Anyone else running into these types of issues?

I am so frustrated with these damn phones (should have paid closer
attention to when they were manufactured so I wouldn't be a beta tester)

Colin
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[asterisk-users] Snom 360

[Dovid Bender]

Hello List,
I am trying to configure QoS for the SNOM 360. I 
plugged the phone in to the internet and then had the customers computer plug in 
to the phone. Whith default settings when I talked on the phone it was great. As 
soon as I started a big download the phone call became unclear. I tried messing 
around with some settings but to no avail. Anyone have any advice ? 
Thanks.
 
Dovid
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[asterisk-users] SNOM 360

[Dovid Bender]

Hi List,Does anyone know how to set up QoS on 
the SNOM 360 ? Thanks.
 
Dovid
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Re: [asterisk-users] SNOM 360

[Giorgio Incantalupo]

Hi UxBoD,
just create a voicemail for your extension and Asterisk will do the rest!!!

Giorgio Incantalupo


--[ UxBoD ]-- wrote:

Hi,

I have got my new phone working with Asterisk, and must say it is very
very good combination. Now I have WMI working, but what I would like to
be able to do is press the DND button on the phone and for all calls to
my extension to be forwarded direct to my voicemail.

How can this be done please ?

TIA

  


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Re: [asterisk-users] SNOM 360

[Andrew Latham]

exten => 123,1,Dial(SIP/123|20)
exten => 123,n,Voicemail(u123)


would be a start, you can have all kinds of fun...


On 3/30/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote:

Hi,

I have got my new phone working with Asterisk, and must say it is very
very good combination. Now I have WMI working, but what I would like to
be able to do is press the DND button on the phone and for all calls to
my extension to be forwarded direct to my voicemail.

How can this be done please ?

TIA

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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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Re: [asterisk-users] SNOM 360

[--[ UxBoD ]--]

Hmmm, okay. But surely it will just try and ring the extension? Or do
you mean setup a seperate extension ie.

exten => 1001,1,Dial(sip/1001,20)
exten => 1001,2,VoiceMail([EMAIL PROTECTED],u)
exten => 1001,3,Hangup()
exten => 1001,101,VoiceMail([EMAIL PROTECTED],u)
exten => 1001,102,Hangup()
exten => 2000,1,VoiceMail([EMAIL PROTECTED],u)
exten => 2000,2,HangUp()

So on pressing the DND it will send all calls to extention 2000 ?

TIA

On Fri, 30 Mar 2007 12:57:16 -0400
"Andrew Latham" <[EMAIL PROTECTED]> wrote:

> exten => 123,1,Dial(SIP/123|20)
> exten => 123,n,Voicemail(u123)
> 
> 
> would be a start, you can have all kinds of fun...
> 
> 
> On 3/30/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > I have got my new phone working with Asterisk, and must say it is
> > very very good combination. Now I have WMI working, but what I
> > would like to be able to do is press the DND button on the phone
> > and for all calls to my extension to be forwarded direct to my
> > voicemail.
> >
> > How can this be done please ?
> >
> > TIA
> >
> > --
> > This message has been scanned for viruses and dangerous content by
> > MailScanner, and is believed to be clean.
> >
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> >
> 
> 

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Re: [asterisk-users] SNOM 360

[bails]

The snom360 DND button forces the phone to give a 480 do not disturb 
response.


Bails
--[ UxBoD ]-- wrote:

Hmmm, okay. But surely it will just try and ring the extension? Or do
you mean setup a seperate extension ie.

exten => 1001,1,Dial(sip/1001,20)
exten => 1001,2,VoiceMail([EMAIL PROTECTED],u)
exten => 1001,3,Hangup()
exten => 1001,101,VoiceMail([EMAIL PROTECTED],u)
exten => 1001,102,Hangup()
exten => 2000,1,VoiceMail([EMAIL PROTECTED],u)
exten => 2000,2,HangUp()

So on pressing the DND it will send all calls to extention 2000 ?

TIA

On Fri, 30 Mar 2007 12:57:16 -0400
"Andrew Latham" <[EMAIL PROTECTED]> wrote:


exten => 123,1,Dial(SIP/123|20)
exten => 123,n,Voicemail(u123)


would be a start, you can have all kinds of fun...


On 3/30/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote:

Hi,

I have got my new phone working with Asterisk, and must say it is
very very good combination. Now I have WMI working, but what I
would like to be able to do is press the DND button on the phone
and for all calls to my extension to be forwarded direct to my
voicemail.

How can this be done please ?

TIA

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Re: [asterisk-users] SNOM 360

[--[ UxBoD ]--]

Ahhh, awesome.  Thank you :)

On Fri, 30 Mar 2007 18:47:30 +0100
bails <[EMAIL PROTECTED]> wrote:

> The snom360 DND button forces the phone to give a 480 do not disturb 
> response.
> 
> Bails
> --[ UxBoD ]-- wrote:
> > Hmmm, okay. But surely it will just try and ring the extension? Or
> > do you mean setup a seperate extension ie.
> > 
> > exten => 1001,1,Dial(sip/1001,20)
> > exten => 1001,2,VoiceMail([EMAIL PROTECTED],u)
> > exten => 1001,3,Hangup()
> > exten => 1001,101,VoiceMail([EMAIL PROTECTED],u)
> > exten => 1001,102,Hangup()
> > exten => 2000,1,VoiceMail([EMAIL PROTECTED],u)
> > exten => 2000,2,HangUp()
> > 
> > So on pressing the DND it will send all calls to extention 2000 ?
> > 
> > TIA
> > 
> > On Fri, 30 Mar 2007 12:57:16 -0400
> > "Andrew Latham" <[EMAIL PROTECTED]> wrote:
> > 
> >> exten => 123,1,Dial(SIP/123|20)
> >> exten => 123,n,Voicemail(u123)
> >>
> >>
> >> would be a start, you can have all kinds of fun...
> >>
> >>
> >> On 3/30/07, --[ UxBoD ]-- <[EMAIL PROTECTED]> wrote:
> >>> Hi,
> >>>
> >>> I have got my new phone working with Asterisk, and must say it is
> >>> very very good combination. Now I have WMI working, but what I
> >>> would like to be able to do is press the DND button on the phone
> >>> and for all calls to my extension to be forwarded direct to my
> >>> voicemail.
> >>>
> >>> How can this be done please ?
> >>>
> >>> TIA
> >>>
> >>> --
> >>> This message has been scanned for viruses and dangerous content by
> >>> MailScanner, and is believed to be clean.
> >>>
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> >>> To UNSUBSCRIBE or update options visit:
> >>>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>>
> >>
> > 
> 
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Re: [asterisk-users] SNOM 360

[J. Oquendo]

bails wrote:
The snom360 DND button forces the phone to give a 480 do not disturb 
response.


Bails
--[ UxBoD ]-- wrote:

Hmmm, okay. But surely it will just try and ring the extension? Or do
you mean setup a seperate extension ie.

exten => 1001,1,Dial(sip/1001,20)
exten => 1001,2,VoiceMail([EMAIL PROTECTED],u)
exten => 1001,3,Hangup()
exten => 1001,101,VoiceMail([EMAIL PROTECTED],u)
exten => 1001,102,Hangup()
exten => 2000,1,VoiceMail([EMAIL PROTECTED],u)
exten => 2000,2,HangUp()


Here is a better fix... If extension 1000 is unavailable whether in DND
or just not there... Call rolls over to extension 2000 with the caller
ID "1000 Unavailable" so the person at 2000 will know so and so didn't
answer their phone because 1000 was wasting their life away on youtube.

exten => 1000,1,Dial(SIP/1000|30|tr)
exten => 1000,2,Set(CALLERID(name)="1000 Unavailable")
exten => 1000,3,SayDigits(1000,f)
exten => 1000,4,Playback(vm-isunavail)
exten => 1000,5,Goto(SIP/2000,20|tr)

So say user @ 1000 is named John, you could change the caller ID to
"John UA" (UA short for the obvious (unavailable) as well as the fact
there isn't enough space for the entire string).

exten => 1000,1,Dial(SIP/1000|30|tr)
exten => 1000,2,Set(CALLERID(name)="Transferred Call")
exten => 1000,3,Wait,4
exten => 1000,4,SayDigits(1000,f)
exten => 1000,5,Playback(vm-isunavail)
exten => 1000,6,SIPAddHeader("Alert-Info: ")
exten => 1000,7,Set(CALLERID(name)="John UA")
exten => 1000,8,Dial(SIP/2000|30|tr)

... Works like this... If user John transfers the call... Whoever he
transfers it to will see its a transferred call. If John (extension 1000)
doesn't answer, the obvious occurs. (unavailable)

I currently use this scheme for one client using Snom 320's and 360's.
The caller ID works for most phones I've tested. Polycoms, Aastra's
however, don't expect Aastra's to play the wav file.

--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
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Re: [asterisk-users] SNOM 360

[Tim Koehler]

Hi Erik,


have tried to switch of RTP Encryption on the snom (in the account/identity
connected to Asterisk).

Cheers

Tim

On 4/26/07, Erik Wartusch <[EMAIL PROTECTED]> wrote:


Hi,

I`ve a strange problem since an upgrade from Asterisk 1.2 to 1.4.1.
If a SNOM 360 calls (internal) another phone type than SNOM (e.g
.  Linksys,
Thomson we have here), there is no audio transmission anymore.
I`ve upgraded the phone to the latest firmware with auto upgrade. No
results.
With 1.2 it was working well.
Any suggestions?

Thanks

Erik
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--
---
snom technology AG

Tim Koehler
Partner Manager
[EMAIL PROTECTED]
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Re: [asterisk-users] SNOM 360

[Erik Wartusch]

Is there a reboot for the phone neccessary? If no then it didn`t work. I 
tested to call a Linksys phone with deactivated RTP encryption, no audio 
transmission.

Erik

Am Donnerstag, 26. April 2007 10:28 schrieb Tim Koehler:
> Hi Erik,
>
>
> have tried to switch of RTP Encryption on the snom (in the account/identity
> connected to Asterisk).
>
> Cheers
>
> Tim
>
> On 4/26/07, Erik Wartusch <[EMAIL PROTECTED]> wrote:
> > Hi,
> >
> > I`ve a strange problem since an upgrade from Asterisk 1.2 to 1.4.1.
> > If a SNOM 360 calls (internal) another phone type than SNOM (e.g
> > .  Linksys,
> > Thomson we have here), there is no audio transmission anymore.
> > I`ve upgraded the phone to the latest firmware with auto upgrade. No
> > results.
> > With 1.2 it was working well.
> > Any suggestions?
> >
> > Thanks
> >
> > Erik
> > ___
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-- 
===

Erik Wartusch
Deuromedia Technologies GmbH
Barichgasse 40-42
1030 Wien
Austria

Phone: +43 16986442 1205
Fax: +43 1 6981274
email: [EMAIL PROTECTED]

www.deuromedia.com
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Re: [asterisk-users] SNOM 360

[Philipp Kempgen]

Him Tim,

Tim Koehler wrote:

> have tried to switch of RTP Encryption on the snom (in the account/identity
> connected to Asterisk).

Nice to know you're on the list. :)

Grüße,
  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [Asterisk-Users] Snom 360

[Gary Stimson]

Hi Colin

I've been using a Snom 360 for 2 weeks and am generally pleased with it.

On Wednesday 11 May 2005 22:12, Colin E. McDonald wrote:
> I am having major problems with the first run of Snom 360s that rolled
> out last month.
>
> Issues:
>
> Speakerphone/Hands Free volume spikes up and down during a call.

Haven't seen that problem.

> You 
> have to manually set the volume during every call.

When you set the volume, press OK. Then it's stored for next time.

> This makes it totally 
> unusable. The sound will cut out completely at the beginning of a call
> sporadically.

Have you tried a different provider?

>
> Call comes through speaker phone after you pick up handset and then cuts
> to handset a couple of seconds later

I don't have that issue.

>
> There is a mnaufacturing defect where the display cable is disconnected
> so you get what appears to be DOA desk sets.

Nor that one. Maybe I was lucky!

>
> Have to press the retrieve message button twice pretty regularly to get
> it to dial vociemail (*97) in asterisk

Haven't got the VM button configured yet, or tried to.

>
> Major problem with calls being dropped when you place callers on hold

I haven't tried putting callers on hold yet.

Have you updated to the latest firmware? Copy the firmware URL from snom.com 
into the relevant box on the phone's web interface, save and reboot the 
phone.

Gary

-- 
Gary Stimson
Zedcore Systems

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Re: [Asterisk-Users] Snom 360

[David John Walsh]

Colin

Similar to Gary's response in that I haven't seem many of these issues.

One that is similar, is that of you saying you need to press voicemail
key twice to get *97 (or eqivilent code)

This as I understand it is not a fault of snom, but a "feature" of
asterisk and the whole MWI "protocol".  When asterisk signals the
phone to say it has voicemail (any phone) it sends in from an address
of "[EMAIL PROTECTED]".  the message text is basically that which pops up
on the bottom line of the display.

When you press the voicemail key, or even the soft voicemail key it
first tries to make contact with "unknown" as this helps ensure that
the right line acesseses its voicemail without the user having to be
aware of which line the voicemail is waiting for them on.

You have two choices, a change the address of the MWI indicator to
come from "[EMAIL PROTECTED]" on the asterisk box or add some lines in your
message-centre context that is similar to
exten => Unknown,1,Voicemail etc

Either of these will bring asterisk up to the level of the snoms features.

I have only one minor issue, and thats if I have several people
ringing into the phone, when I am not already on a call (all calls are
still in the setup phase) I can't choose by pressing the flashing
lights, I have to dump them using the soft "no thanks" or the hard x
key

You almost sound like you have a earlier firmware issue.  The latest
one is 3.60f

a direct link to the firmware is http://www.snom.com/download/share/

I tell a lie -the very latest firmware is 3.60h - as of the 4th May

David

On 5/12/05, Gary Stimson <[EMAIL PROTECTED]> wrote:
> Hi Colin
> 
> I've been using a Snom 360 for 2 weeks and am generally pleased with it.
> 
> On Wednesday 11 May 2005 22:12, Colin E. McDonald wrote:
> > I am having major problems with the first run of Snom 360s that rolled
> > out last month.
> >
> > Issues:
> >
> > Speakerphone/Hands Free volume spikes up and down during a call.
> 
> Haven't seen that problem.
> 
> > You
> > have to manually set the volume during every call.
> 
> When you set the volume, press OK. Then it's stored for next time.
> 
> > This makes it totally
> > unusable. The sound will cut out completely at the beginning of a call
> > sporadically.
> 
> Have you tried a different provider?
> 
> >
> > Call comes through speaker phone after you pick up handset and then cuts
> > to handset a couple of seconds later
> 
> I don't have that issue.
> 
> >
> > There is a mnaufacturing defect where the display cable is disconnected
> > so you get what appears to be DOA desk sets.
> 
> Nor that one. Maybe I was lucky!
> 
> >
> > Have to press the retrieve message button twice pretty regularly to get
> > it to dial vociemail (*97) in asterisk
> 
> Haven't got the VM button configured yet, or tried to.
> 
> >
> > Major problem with calls being dropped when you place callers on hold
> 
> I haven't tried putting callers on hold yet.
> 
> Have you updated to the latest firmware? Copy the firmware URL from snom.com
> into the relevant box on the phone's web interface, save and reboot the
> phone.
> 
> Gary
> 
> --
> Gary Stimson
> Zedcore Systems
> 
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[Asterisk-Users] Snom 360 problem

[altus]

Good day all
I have asterisk on a box with one network card
I have a 2 companies setup on the system.
To keep all apart I binded a different ip to the interface,i,o,w eth0
192.168.0.254 and eth0:1 192.168.1.254
And in sip.conf I took the bind setting out
So each company's phones are on a different ip range,and all worked well
So we decide to pull the snom190 out and exchange it with a snom360
This company is on the virtual interface(eth0:1)
The 360 register and can make outgoing calls fine
But when you try to make a call to it it does not work?
I gives this error in the cli
"Forbidden - wrong password on authentication for INVITE to '"301"
;tag=as3405ec0a"
But its 301 calling the snom360(user 310)???
BUT
If I change the phone's ip and tell it to connect to eth0,not eth0:1 it
works,same account settings same everything?
The snom190 worked this way
Any Ideas why?



-- 

Thanks
Altus Snyman
Stormcorp Network Solutions
+27 11 8071141 exten 301

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RE: [asterisk-users] Snom 360

[Christian Stredicke]

Welcome to VoIP... Your operator needs to take care about 
QoS when you are doing a download. Alternatively, there are some more-or-less 
tricky and buggy tricks to stop downloads when you are talking; this needs to be 
done on your IAD.
 
See for example http://www.voip-info.org/wiki-QoS.
 
CS

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Dovid 
  BenderSent: Wednesday, July 26, 2006 12:46 PMTo: 
  asterisk-users@lists.digium.comSubject: [asterisk-users] Snom 
  360
  
  Hello List,
  I am trying to configure QoS for the SNOM 360. I 
  plugged the phone in to the internet and then had the customers computer plug 
  in to the phone. Whith default settings when I talked on the phone it was 
  great. As soon as I started a big download the phone call became unclear. I 
  tried messing around with some settings but to no avail. Anyone have any 
  advice ? Thanks.
   
  Dovid
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RE: [asterisk-users] SNOM 360

[Koopmann, Jan-Peter]

On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote:

> Does anyone know how to set up QoS on the SNOM 360 ? Thanks.

What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a 
Snom 360 that will manage things for you. AFAIK all you can do is tell the 
phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of 
the equipment managing the bottleneck (firewall, router whatever) to use this 
information and manage your traffic accordingly.

Regards,
  JP

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Re: [asterisk-users] SNOM 360

[Steve Davies]

On 7/28/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote:

On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote:

> Does anyone know how to set up QoS on the SNOM 360 ? Thanks.

What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch on a 
Snom 360 that will manage things for you. AFAIK all you can do is tell the 
phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the task of 
the equipment managing the bottleneck (firewall, router whatever) to use this 
information and manage your traffic accordingly.



As I understand it, you can set a QoS priority if the phone is in a
VLAN. When you configure the (Tagged) VLAN, you can specify the
priority of the packets in the VLAN.

Otherwise, newer firmware allows the setting of TOS values IIRC.

Regards,
Steve
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Re: [asterisk-users] SNOM 360

[Dovid Bender]

I am trying to have thier PC run thru the port on the phone and the phone 
give prioroty to itself and the rest to the PC. When my client does a big 
download the phone call gets real bad. The docs from SNOM on TOS (or 
DIFFSERV) is poor and I dont understand it well enough. Anyone have configs 
or docs on how they did this ?


Doid
- Original Message - 
From: "Koopmann, Jan-Peter" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Friday, July 28, 2006 3:17 AM
Subject: RE: [asterisk-users] SNOM 360


On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote:


Does anyone know how to set up QoS on the SNOM 360 ? Thanks.


What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch 
on a Snom 360 that will manage things for you. AFAIK all you can do is tell 
the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the 
task of the equipment managing the bottleneck (firewall, router whatever) to 
use this information and manage your traffic accordingly.


Regards,
 JP

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Re: [asterisk-users] SNOM 360

[Dovid Bender]

Also SNOM says by Vlan to set the vlan and then the value for the qos. When 
you set Vlan to 0 it is supposed to be no Vlan. However once I set it the 
vlan on the SNOM to 0 and I reboot the phone is no long accessable from the 
network and I have to reset it.


Dovid

- Original Message - 
From: "Koopmann, Jan-Peter" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Friday, July 28, 2006 3:17 AM
Subject: RE: [asterisk-users] SNOM 360


On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote:


Does anyone know how to set up QoS on the SNOM 360 ? Thanks.


What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch 
on a Snom 360 that will manage things for you. AFAIK all you can do is tell 
the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the 
task of the equipment managing the bottleneck (firewall, router whatever) to 
use this information and manage your traffic accordingly.


Regards,
 JP

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Re: [asterisk-users] SNOM 360

[Robbie Hughes]

I would be surprised if the problem is at the phone.
I have nearly a hundred 360s, 190s and not one of them suffers from that
problem in the default setting. The phone handles it automatically.
BUT..if I download from an external site and I pipe the call over the
internet without setting any traffic shaping on the router then it gets
jumpy. Also, you may experience the same problem if you're somehow
saturating the network interface on the switch or the asterisk server (both
which is highly unlikely).

Check you have some sort of traffic shaping on your router and ensure you
have a decent switch. I like m0n0wall for routers and cisco for switches.

> --
> 
> Message: 9
> Date: Fri, 28 Jul 2006 09:08:17 -0400
> From: "Dovid Bender" <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] SNOM 360
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Message-ID: <[EMAIL PROTECTED]>
> Content-Type: text/plain; format=flowed; charset="Windows-1252";
> reply-type=original
> 
> I am trying to have thier PC run thru the port on the phone and the phone
> give prioroty to itself and the rest to the PC. When my client does a big
> download the phone call gets real bad. The docs from SNOM on TOS (or
> DIFFSERV) is poor and I dont understand it well enough. Anyone have configs
> or docs on how they did this ?
> 
> Doid


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RE: [asterisk-users] SNOM 360

[Koopmann, Jan-Peter]

On Friday, July 28, 2006 3:08 PM Dovid Bender wrote:

> I am trying to have thier PC run thru the port on the phone and the
> phone give prioroty to itself and the rest to the PC. When my client
> does a big download the phone call gets real bad. The docs from SNOM
> on TOS (or DIFFSERV) is poor and I dont understand it well enough.
> Anyone have configs or docs on how they did this ?   

I would be surprised to learn that the Snom is actively doing traffic 
management itself. Traffic managment must be done at the bottleneck to be 
halfway successful. Let's assume you are doing a download and you snom would do 
traffic management giving itself priority. What if your co-worker is doing a 
huge download? How should your snom know and throttle his download? No way. 

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Re: [asterisk-users] SNOM 360

[Steve Davies]

On 7/28/06, Dovid Bender <[EMAIL PROTECTED]> wrote:

Also SNOM says by Vlan to set the vlan and then the value for the qos. When
you set Vlan to 0 it is supposed to be no Vlan. However once I set it the
vlan on the SNOM to 0 and I reboot the phone is no long accessable from the
network and I have to reset it.



The "Qos" field is part of the 802.11q header, so is only available if
a VLAN has been configured.

VLAN 0 is a perfectly valid VLAN, and will cause an 802.11q packet
header to be added to all the phone traffic. This will then only work
if the rest of the network understands "tagged" VLAN 0 packets.

Regards,
Steve
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Re: [asterisk-users] SNOM 360

[Steve Davies]

On 7/31/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote:

On Friday, July 28, 2006 3:08 PM Dovid Bender wrote:

> I am trying to have thier PC run thru the port on the phone and the
> phone give prioroty to itself and the rest to the PC. When my client
> does a big download the phone call gets real bad. The docs from SNOM
> on TOS (or DIFFSERV) is poor and I dont understand it well enough.
> Anyone have configs or docs on how they did this ?

I would be surprised to learn that the Snom is actively doing traffic 
management itself.
Traffic managment must be done at the bottleneck to be halfway successful. Let's
assume you are doing a download and you snom would do traffic management giving
itself priority. What if your co-worker is doing a huge download? How should 
your
snom know and throttle his download? No way.


That is a different problem entirely, and as you say, the snom cannot
do anything about a remote bottleneck (except perhaps theough QoS and
TOS flags in the data it sends).

The snom does seem to manage its two local ports properly though but
this cannot be hard. Worst case is that the snom needs about 128Kb/s -
Not hard on a 100Mb/s full duplex connection :)

Dovid - Have you identified where the bottleneck is in this case? You
do not specify as far as I can see. Is the VoIP call using the
internet, or is it local?

Regards,
Steve
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Re: [asterisk-users] SNOM 360

[Dovid Bender]

- Original Message - 
From: "Steve Davies" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Monday, July 31, 2006 6:01 AM
Subject: Re: [asterisk-users] SNOM 360



On 7/31/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote:

On Friday, July 28, 2006 3:08 PM Dovid Bender wrote:

> I am trying to have thier PC run thru the port on the phone and the
> phone give prioroty to itself and the rest to the PC. When my client
> does a big download the phone call gets real bad. The docs from SNOM
> on TOS (or DIFFSERV) is poor and I dont understand it well enough.
> Anyone have configs or docs on how they did this ?

I would be surprised to learn that the Snom is actively doing traffic 
management itself.
Traffic managment must be done at the bottleneck to be halfway 
successful. Let's
assume you are doing a download and you snom would do traffic management 
giving
itself priority. What if your co-worker is doing a huge download? How 
should your

snom know and throttle his download? No way.


That is a different problem entirely, and as you say, the snom cannot
do anything about a remote bottleneck (except perhaps theough QoS and
TOS flags in the data it sends).

The snom does seem to manage its two local ports properly though but
this cannot be hard. Worst case is that the snom needs about 128Kb/s -
Not hard on a 100Mb/s full duplex connection :)

Dovid - Have you identified where the bottleneck is in this case? You
do not specify as far as I can see. Is the VoIP call using the
internet, or is it local?

Regards,
Steve
It is using the internet. The problem is when a user starts a big download. 
The phone call goes to s***. 


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Re: [asterisk-users] SNOM 360

[Dovid Bender]

- Original Message - 
From: "Steve Davies" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Monday, July 31, 2006 5:57 AM
Subject: Re: [asterisk-users] SNOM 360



On 7/28/06, Dovid Bender <[EMAIL PROTECTED]> wrote:
Also SNOM says by Vlan to set the vlan and then the value for the qos. 
When

you set Vlan to 0 it is supposed to be no Vlan. However once I set it the
vlan on the SNOM to 0 and I reboot the phone is no long accessable from 
the

network and I have to reset it.



The "Qos" field is part of the 802.11q header, so is only available if
a VLAN has been configured.

VLAN 0 is a perfectly valid VLAN, and will cause an 802.11q packet
header to be added to all the phone traffic. This will then only work
if the rest of the network understands "tagged" VLAN 0 packets.

Regards,
Steve
When I set it to 0 I loose all conectivity to the phone. Cant ping it or 
anything. I have to reset it from the phone to get access to it again. 


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Re: [asterisk-users] SNOM 360

[Steve Davies]

On 8/4/06, Dovid Bender <[EMAIL PROTECTED]> wrote:

>
> The snom does seem to manage its two local ports properly though but
> this cannot be hard. Worst case is that the snom needs about 128Kb/s -
> Not hard on a 100Mb/s full duplex connection :)
>
> Dovid - Have you identified where the bottleneck is in this case? You
> do not specify as far as I can see. Is the VoIP call using the
> internet, or is it local?
>
It is using the internet. The problem is when a user starts a big download.
The phone call goes to s***.


Dovid,

There are devices on the market that claim to prioritise certain
traffic in favour of downloads and browsing. This can obviously only
prioritise traffic outbound, but on an ADSL link, this is often what
is needed.

Try googling for such a device? We don't use them ourselves as we
operate on the principle that a "free" call is worth what you pay for
it :)

Steve
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Re: [asterisk-users] SNOM 360

[Steve Davies]

On 8/4/06, Dovid Bender <[EMAIL PROTECTED]> wrote:

>
> The "Qos" field is part of the 802.11q header, so is only available if
> a VLAN has been configured.
>
> VLAN 0 is a perfectly valid VLAN, and will cause an 802.11q packet
> header to be added to all the phone traffic. This will then only work
> if the rest of the network understands "tagged" VLAN 0 packets.
>



When I set it to 0 I loose all conectivity to the phone. Cant ping it or
anything. I have to reset it from the phone to get access to it again.


Of course you do - nothing else on your network is in VLAN 0, so you
lose connectivity.

Steve
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Re: [asterisk-users] SNOM 360

[Julio Arruda]

Dovid Bender wrote:


- Original Message - From: "Steve Davies" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Monday, July 31, 2006 6:01 AM
Subject: Re: [asterisk-users] SNOM 360



On 7/31/06, Koopmann, Jan-Peter <[EMAIL PROTECTED]> wrote:

On Friday, July 28, 2006 3:08 PM Dovid Bender wrote:

> I am trying to have thier PC run thru the port on the phone and the
> phone give prioroty to itself and the rest to the PC. When my client
> does a big download the phone call gets real bad. The docs from SNOM
> on TOS (or DIFFSERV) is poor and I dont understand it well enough.
> Anyone have configs or docs on how they did this ?

I would be surprised to learn that the Snom is actively doing traffic 
management itself.
Traffic managment must be done at the bottleneck to be halfway 
successful. Let's
assume you are doing a download and you snom would do traffic 
management giving
itself priority. What if your co-worker is doing a huge download? How 
should your

snom know and throttle his download? No way.


That is a different problem entirely, and as you say, the snom cannot
do anything about a remote bottleneck (except perhaps theough QoS and
TOS flags in the data it sends).

The snom does seem to manage its two local ports properly though but
this cannot be hard. Worst case is that the snom needs about 128Kb/s -
Not hard on a 100Mb/s full duplex connection :)

Dovid - Have you identified where the bottleneck is in this case? You
do not specify as far as I can see. Is the VoIP call using the
internet, or is it local?

Regards,
Steve
It is using the internet. The problem is when a user starts a big 
download. The phone call goes to s***.



Dovid,
I would guess that:
First thing would be quick&dirty ASCII drawing, showing where is the PC, 
the SNOM and the "sources/destinations" of the Internet and VOIP traffic.
You mentioned download, assuming this is a DSL connection, this would 
be, when it arrive at the IP phone, would be too late to do anything, IF 
you are bumping into a bottleneck in the DSL downstream.
What 'direction' of the voice path is suffering, did you capture the 
traffic (is it suffering because of jitter, packet loss, ...) ?
Like others mention, QoS (the buzzword :-), is a very wide and generic 
term, and you will need to 'isolate' the problem to see if a solution is 
feasible.


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[Asterisk-Users] Snom 360 Opinions

[Curren C. Calhoun]

Title: Snom 360 Opinions



I’m looking to add in some Snom 360 phones, could anyone give thoughts or opinions about the speakerphone, general quality... Also the phone would need to be powered over Ethernet...

I like some of the listed features and the expandability of the phone but am open to any other suggestions as well...

Thanks


Curren



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[Asterisk-Users] Snom 360 problems

[Brian Kennedy]

Anyone have a Snom they're happy with?   How did you manage that?  :)

I have a system of:

Asterisk 1.2.3
2 Wildcard TDM400P  Rev I and E/F
1 Snom 360 + sidecar 
~15 Sipura/Linsys SPA-841

~15 Grandstream 101

Everything (currently) is on the same network, not a router to be seen 
between any two.  Also everything, except the snom, is working sweetly.


The main problem is ECHO.. awful echo and only on the Snom.  When using 
a Zap line or to another sip phone.  I've tweaked the * for echo and 
managed to only create echo and piss everyone else off, pounded the 
settings in the Snom trying to find anything, and updated the firmware 
to Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2 
v3.36 after noticing a changelog that sounded like it may have related 
to echo.  Not even a slight reduction in echo so far.


A second serious problem is Call join.   Even with "Call join on Xfer (2 
calls)" OFF if the user is doing a transfer of one call when a second 
starts ringing the 2 callers get bridged, no transfer.  Really nice, now 
I have two customers talking to each other with no clue what's going on 
and neither gets who they were trying to reach.


Any ideas on what I can try next?

Thanks...
...Brian
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[Asterisk-Users] Snom 360 problems

[Brian Kennedy]

Anyone have a Snom they're happy with?   How did you manage that?   :)

I have a system of:

Asterisk 1.2.3
2 Wildcard TDM400P  Rev I and E/F
1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841
~15 Grandstream 101

Everything (currently) is on the same network, not a router to be seen 
between any two.  Also everything, except the snom, is working sweetly.


The main problem is ECHO.. awful echo and only on the Snom.  When using 
a Zap line or to another sip phone.  I've tweaked the * for echo and 
managed to only create echo for everyone else, pounded the settings in 
the Snom trying to find anything, and updated the firmware to 
Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2 v3.36 
after noticing a changelog that sounded like it may have related to 
echo.  Not even a slight reduction in echo so far.


A second serious problem is Call join.   Even with "Call join on Xfer (2 
calls)" OFF if the user is doing a transfer of one call when a second 
starts ringing the 2 callers get bridged, no transfer.  Really nice, now 
I have two customers talking to each other with no clue what's going on 
and neither gets who they were trying to reach.


Any ideas on what I can try next?

Thanks...
...Brian
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[asterisk-users] snom 360 auto answer

[Jason Kim]

Hi,

I'm testing paging using snom 360.
Can someone correct my dialplan?

Regards,
Jason.

==
;exten => _99,1,SIPAddHeader(Call-Info:
Answer-After=0)
;exten => _99,n,SIPAddHeader(Call-Info:
\;answer-after=0)
;exten => _99,n,Dial(SIP/${EXTEN:2})

exten => _99,1,Set(__SIPADDHEADER=Call-Info:
answer-after=0)
exten =>
_99,n,Set(__SIP_URI_OPTIONS=intercom=true)
exten => _99,n,Set(__ALERT_INFO=Ring Answer)
exten => _99,n,Dial(SIP/${EXTEN:2})


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[asterisk-users] SNOM 360 Rejecting Calls

[Dovid B]

 Does someone know coincidentally the cause for the error message specified in 
the Subject?

The following scenario: Snom 360 behind one rout (wiederrum on a DSL line with 
static IP address hangs). The Snom has a private IP, routs accomplishes NAT. 
STUN and ICE are activated, as SIP haven 5060/udp are firmly used. Detailed 
packages passed on on haven 5060/udp of rout to the Snom.

The telephone registers itself as expected, and outgoing telephone calls can be 
led problem-free. Detailed telephone calls however do not function (a call is 
signaled, which rings then however after 3 time on the mailbox is sent).

The SIP log shows that the telephone sees the INVITE of the Registrar, it 
however in principle with "486 Busy here" answered (that is then also the 
reason, why the detailed call is sent on the mailbox). The message mentioned 
"Denying call appears contemporaneous id=X reason=unconditional" in the log, 
whereby X was so far always a negative, one-digit number.

Does someone have an idea?
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[Asterisk-Users] SNOM 360 extension lights

[Ross Kevlin]

I recently got a SNOM 360 and have been trying 
to get the extension lights to work. I can see the subscriptions with sip show 
subscriptions but I don't see any notifies when a call is made. I must be 
missing something because I've tried looking to see if anyone else has had this 
problem but the only solutions I've seen have been to put hints in and I have 
those. Any suggestions?
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[Asterisk-Users] SNOM 360 extension lights

[Ross Kevlin]

I contacted SNOM and they told me to change a 
couple of options but still no lights, here is what they told me
 
Line page SIP tab:o Long SIP-Contact (RFC3840) to "off"o 
Support broken Registrar to "on"Advanced page:o Filter Packets 
from Registrar to "off"
 
And please ask the Asterisk community for help, I'm sure they solved 
thatissue 100%, and we are not knowing so much about Asterisk.Your 
snom support Teamhas anyone gotten a 360 to work 
with the lights? what options and modifications to .conf files did you have to 
make?
 
here are the subscribe and notifies.
it seems it terminates the subscription as soon as 
its created. I don't think its a proxy authentication problem
because it eventually sends the proxy 
authentication information
 
Using latest SUBSCRIBE request as basis 
requestSending to 192.168.2.230 : 2051 (non-NAT)Found peer 
'83'Transmitting (no NAT) to 192.168.2.230:2051:SIP/2.0 407 Proxy 
Authentication RequiredVia: SIP/2.0/UDP 
192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049nFrom: 
;tag=z6kvtd67buTo: 
;tag=as6c1cb2a5Call-ID: [EMAIL PROTECTED]CSeq: 
1 SUBSCRIBEUser-Agent: MVC 001Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, 
REFER, NOTIFYContact: Proxy-Authenticate: 
Digest realm="asterisk", nonce="16747f76"Content-Length: 0
 
---Scheduling destruction of call '[EMAIL PROTECTED]' 
in 15000 mssip1*CLI><-- SIP read from 
192.168.2.230:2051:SUBSCRIBE sip:[EMAIL PROTECTED];user=phone 
SIP/2.0Via: SIP/2.0/UDP 
192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n;rportFrom: 
;tag=z6kvtd67buTo: 
Call-ID: [EMAIL PROTECTED]CSeq: 
1 SUBSCRIBEMax-Forwards: 70Contact: 
Event: dialogAccept: 
application/dialog-info+xmlExpires: 3600Content-Length: 0
 
--- (12 headers 0 lines)---Ignoring this SUBSCRIBE requestFound 
peer '83'Transmitting (no NAT) to 192.168.2.230:2051:SIP/2.0 407 Proxy 
Authentication RequiredVia: SIP/2.0/UDP 
192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049nFrom: 
;tag=z6kvtd67buTo: 
;tag=as6c1cb2a5Call-ID: [EMAIL PROTECTED]CSeq: 
1 SUBSCRIBEUser-Agent: MVC 001Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, 
REFER, NOTIFYContact: Proxy-Authenticate: 
Digest realm="asterisk", nonce="16747f76"Content-Length: 0
 
---Scheduling destruction of call '[EMAIL PROTECTED]' 
in 15000 mssip1*CLI><-- SIP read from 
192.168.2.230:2051:SUBSCRIBE sip:[EMAIL PROTECTED];user=phone 
SIP/2.0Via: SIP/2.0/UDP 
192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x;rportFrom: 
;tag=z6kvtd67buTo: 
Call-ID: [EMAIL PROTECTED]CSeq: 
2 SUBSCRIBEMax-Forwards: 70Contact: 
Event: dialogAccept: 
application/dialog-info+xmlProxy-Authorization: Digest 
username="83",realm="asterisk",nonce="16747f76",uri="sip:[EMAIL PROTECTED];user=phone",response="15d72104244317e2c0afa3499220e4ab",algorithm=md5Expires: 
3600Content-Length: 0
 
--- (13 headers 0 lines)---Using latest SUBSCRIBE request as basis 
requestSending to 192.168.2.230 : 2051 (non-NAT)Found peer 
'83'Looking for 117 in localusers-C2021-1Transmitting (no NAT) to 
192.168.2.230:2051:SIP/2.0 200 OKVia: SIP/2.0/UDP 
192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7xFrom: 
;tag=z6kvtd67buTo: 
;tag=as77c7b911Call-ID: [EMAIL PROTECTED]CSeq: 
2 SUBSCRIBEUser-Agent: MVC 001Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, 
REFER, NOTIFYExpires: 3600Contact: 
;expires=3600Content-Length: 0
 
---Scheduling destruction of call '[EMAIL PROTECTED]' 
in 361 msReliably Transmitting (no NAT) to 192.168.2.230:2051:NOTIFY 
sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 
192.168.2.252:5060;branch=z9hG4bK56396cd7;rportFrom: 
;tag=as77c7b911To: 
;tag=z6kvtd67buContact: 
Call-ID: [EMAIL PROTECTED]CSeq: 
102 NOTIFYUser-Agent: MVC 001Event: dialogContent-Type: 
application/dialog-info+xmlContent-Length: 203
 
 entity="sip:[EMAIL PROTECTED]">terminated
 
---sip1*CLI><-- SIP read from 192.168.2.230:2051:SIP/2.0 
200 OkVia: SIP/2.0/UDP 
192.168.2.252:5060;branch=z9hG4bK56396cd7;rport=5060From: 
;tag=as77c7b911To: 
;tag=z6kvtd67buCall-ID: [EMAIL PROTECTED]CSeq: 
102 NOTIFYContent-Length: 0
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[Asterisk-Users] SNOM 360 extension lights

[Ross Kevlin]

 I have a snom 360 that im trying to get the extension lights working i can
see the subscription being sent and a reply but the reply is a terminate.
Using sip show peer i can see the subscriptions but no uri. i have the hints
in place but i dont see any notifys when a line is in use.


>Sorry Ross I must have missed your first postings, but what are you
>trying to achive?

>David

>On 03/06/05, Ross Kevlin  wrote:
>
> I contacted SNOM and they told me to change a couple of options but still
no
> lights, here is what they told me
>
> Line page SIP tab:
>
> o Long SIP-Contact (RFC3840) to "off"
> o Support broken Registrar to "on"
>
> Advanced page:
>
> o Filter Packets from Registrar to "off"
>
> And please ask the Asterisk community for help, I'm sure they solved that
> issue 100%, and we are not knowing so much about Asterisk.
>
> Your snom support Team
>
> has anyone gotten a 360 to work with the lights? what options and
> modifications to .conf files did you have to make?
>
> here are the subscribe and notifies.
> it seems it terminates the subscription as soon as its created. I don't
> think its a proxy authentication problem
> because it eventually sends the proxy authentication information
>
> Using latest SUBSCRIBE request as basis request
> Sending to 192.168.2.230 : 2051 (non-NAT)
> Found peer '83'
> Transmitting (no NAT) to 192.168.2.230:2051:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n
> From: ;tag=z6kvtd67bu
> To: ;tag=as6c1cb2a5
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 1 SUBSCRIBE
> User-Agent: MVC 001
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: 
> Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76"
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call
> '3c2670ad35b6-68nuemr6pg58 at snom360' in 15000 ms
> sip1*CLI>
> <-- SIP read from 192.168.2.230:2051:
> SUBSCRIBE sip:117 at 192.168.2.252;user=phone SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n;rport
> From: ;tag=z6kvtd67bu
> To: 
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 1 SUBSCRIBE
> Max-Forwards: 70
> Contact: 
> Event: dialog
> Accept: application/dialog-info+xml
> Expires: 3600
> Content-Length: 0
>
>
> --- (12 headers 0 lines)---
> Ignoring this SUBSCRIBE request
> Found peer '83'
> Transmitting (no NAT) to 192.168.2.230:2051:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n
> From: ;tag=z6kvtd67bu
> To: ;tag=as6c1cb2a5
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 1 SUBSCRIBE
> User-Agent: MVC 001
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: 
> Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76"
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call
> '3c2670ad35b6-68nuemr6pg58 at snom360' in 15000 ms
> sip1*CLI>
> <-- SIP read from 192.168.2.230:2051:
> SUBSCRIBE sip:117 at 192.168.2.252;user=phone SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x;rport
> From: ;tag=z6kvtd67bu
> To: 
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 2 SUBSCRIBE
> Max-Forwards: 70
> Contact: 
> Event: dialog
> Accept: application/dialog-info+xml
> Proxy-Authorization: Digest
> username="83",realm="asterisk",nonce="16747f76",uri=
> "sip:117 at
192.168.2.252;user=phone",response="15d72104244317e2c0afa3499220e4ab",a
> lgorithm=md5
> Expires: 3600
> Content-Length: 0
>
>
> --- (13 headers 0 lines)---
> Using latest SUBSCRIBE request as basis request
> Sending to 192.168.2.230 : 2051 (non-NAT)
> Found peer '83'
> Looking for 117 in localusers-C2021-1
> Transmitting (no NAT) to 192.168.2.230:2051:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x
> From: ;tag=z6kvtd67bu
> To: ;tag=as77c7b911
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 2 SUBSCRIBE
> User-Agent: MVC 001
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Expires: 3600
> Contact: ;expires=3600
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call
> '3c2670ad35b6-68nuemr6pg58 at snom360' in 361 ms
> Reliably Transmitting (no NAT) to 192.168.2.230:2051:
> NOTIFY sip:83 at 192.168.2.252 SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport
> From: ;tag=as77c7b911
> To: ;tag=z6kvtd67bu
> Contact: 
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 102 NOTIFY
> User-Agent: MVC 001
> Event: dialog
> Content-Type: application/dialog-info+xml
> Content-Length: 203
>
> 
>  version="0" state="full"
>  entity="sip:83 at 192.168.2.252">
> 
> terminated
> 
> 
>
> ---
> sip1*CLI>
> <-- SIP read from 192.168.2.230:2051:
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP
> 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport=5060
> From: ;tag=as77c7b911
> To: ;tag=z6kvtd67bu
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 102 NOTIFY
> Content-Length: 0
> ___
> Asterisk-Users ma

[Asterisk-Users] SNOM 360 extension lights

[Ross Kevlin]

the subscription is sent a reply and the reply has content that indicates
its state is terminated

From: ;tag=as77c7b911
 To: ;tag=z6kvtd67bu
 Contact: 
 Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
 CSeq: 102 NOTIFY
 User-Agent: MVC 001
 Event: dialog
 Content-Type: application/dialog-info+xml
 Content-Length: 203

 
 
 
 terminated
 
 

from what I understand the terminated state is the end of dialog and the
subscription should end, but I still see the subscription in asterisk.

>Ross Kevlin wrote:
>  I have a snom 360 that i'm trying to get the extension lights working I
can
> see the subscription being sent and a reply but the reply is a terminate.
>Terminate being?

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[Asterisk-Users] snom 360 conference button

[Christian Hiller]

Hello,

we have couple of snom 360 and we have implemtented all features of the 
phone with the asterisk, but the conference button.
Its just not working. And I also dont find any informations how to 
configure it.


Any help apreciated
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[Asterisk-Users] SNOM 360 extension lights

[Ross Kevlin]

i think this is the full packet. if it isn't, how do see the full packet

 
> Using latest SUBSCRIBE request as basis request
> Sending to 192.168.2.230 : 2051 (non-NAT)
> Found peer '83'
> Transmitting (no NAT) to 192.168.2.230:2051:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n
> From: ;tag=z6kvtd67bu
> To: ;tag=as6c1cb2a5
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 1 SUBSCRIBE
> User-Agent: MVC 001
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: 
> Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76"
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call
> '3c2670ad35b6-68nuemr6pg58 at snom360' in 15000 ms
> sip1*CLI>
> <-- SIP read from 192.168.2.230:2051:
> SUBSCRIBE sip:117 at 192.168.2.252;user=phone SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n;rport
> From: ;tag=z6kvtd67bu
> To: 
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 1 SUBSCRIBE
> Max-Forwards: 70
> Contact: 
> Event: dialog
> Accept: application/dialog-info+xml
> Expires: 3600
> Content-Length: 0
>
>
> --- (12 headers 0 lines)---
> Ignoring this SUBSCRIBE request
> Found peer '83'
> Transmitting (no NAT) to 192.168.2.230:2051:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n
> From: ;tag=z6kvtd67bu
> To: ;tag=as6c1cb2a5
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 1 SUBSCRIBE
> User-Agent: MVC 001
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: 
> Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76"
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call
> '3c2670ad35b6-68nuemr6pg58 at snom360' in 15000 ms
> sip1*CLI>
> <-- SIP read from 192.168.2.230:2051:
> SUBSCRIBE sip:117 at 192.168.2.252;user=phone SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x;rport
> From: ;tag=z6kvtd67bu
> To: 
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 2 SUBSCRIBE
> Max-Forwards: 70
> Contact: 
> Event: dialog
> Accept: application/dialog-info+xml
> Proxy-Authorization: Digest
> username="83",realm="asterisk",nonce="16747f76",uri=
> "sip:117 at
192.168.2.252;user=phone",response="15d72104244317e2c0afa3499220e4ab",a
> lgorithm=md5
> Expires: 3600
> Content-Length: 0
>
>
> --- (13 headers 0 lines)---
> Using latest SUBSCRIBE request as basis request
> Sending to 192.168.2.230 : 2051 (non-NAT)
> Found peer '83'
> Looking for 117 in localusers-C2021-1
> Transmitting (no NAT) to 192.168.2.230:2051:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x
> From: ;tag=z6kvtd67bu
> To: ;tag=as77c7b911
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 2 SUBSCRIBE
> User-Agent: MVC 001
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Expires: 3600
> Contact: ;expires=3600
> Content-Length: 0
>
>
> ---
> Scheduling destruction of call
> '3c2670ad35b6-68nuemr6pg58 at snom360' in 361 ms
> Reliably Transmitting (no NAT) to 192.168.2.230:2051:
> NOTIFY sip:83 at 192.168.2.252 SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport
> From: ;tag=as77c7b911
> To: ;tag=z6kvtd67bu
> Contact: 
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 102 NOTIFY
> User-Agent: MVC 001
> Event: dialog
> Content-Type: application/dialog-info+xml
> Content-Length: 203
>
> 
>  version="0" state="full"
>  entity="sip:83 at 192.168.2.252">
> 
> terminated
> 
> 
>
> ---
> sip1*CLI>
> <-- SIP read from 192.168.2.230:2051:
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP
> 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport=5060
> From: ;tag=as77c7b911
> To: ;tag=z6kvtd67bu
> Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
> CSeq: 102 NOTIFY
> Content-Length: 0
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[Asterisk-Users] Snom 360 Hinting tricks

[Colin Anderson]

I was always puzzled by posts to the list about people having problems
getting hints to work on a Snom, since I always seem to have no problem
making it work. That is, until today when I tried to get a sidecar to work.
All I could do was get a monitored extension light to light up continuously,
regardless of state. Frustrating! Going back to my working dialplans where I
got 1 or 2 lights working fine, I saw the pattern and the difference between
working and non-working, and I realized that other people were experiencing
the same problem as I was. The trick is the *order* in which you put your
hint priorities in your dialplan. My non-working sidecar dialplan had all
the hint priorities grouped together:

exten => 12345,hint,SIP/12345
exten => 12346,hint,SIP/12346

Which would register the hint, but it wouldn't work on the Snom. The way to
make it work, for sure, is to make sure your hint priority is the last
priority underneath the *related* priority for the extension. So, this will
work:

exten => 12345,1,Dial(SIP/12345)
exten => 12345,2,Voicemail(u12345)
exten => 12345,hint,SIP/12345

exten => 12346,1,Dial(SIP/12346)
exten => 12346,2,Voicemail(u12346)
exten => 12346,hint,SIP/12346

But this won't:

exten => 12345,hint,SIP/12345
exten => 12346,hint,SIP/12346

exten => 12345,1,Dial(SIP/12345)
exten => 12345,2,Voicemail(u12345)

exten => 12346,1,Dial(SIP/12346)
exten => 12346,2,Voicemail(u12346)

Also, you will get hooped if you lower-case your "SIP" statements. So
SIP/12345 will work but sip/12345 won't. 

As long as you follow these two tricks above, hint-ing is very
straightforward and painless. If you don't, it's really frustrating to get
going (as I found out today after a couple of hours of swearing) 
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[Asterisk-Users] Snom 360 NOTIFY syntax

[Patrick Friedel]

I'm rolling out an installation with snom 360s in the near future.  
Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a 
snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002.  I 
have the 360's set up to subscribe and notify for the line use lights, 
which works like a charm for interoffice calling (between the 360's, 
anyway.  The IAXy, 200 and, softphone will be used by less phone 
dependant types) but what I can't figure out from the Wiki is if it's 
possible to have the ZAP lines notify for the outbound lines so we can 
see how many lines are in use.


 My configuration looks something like this:

sip.conf:
[mjg]
type=friend
username=mjg
context=sip
callerid="Masuo" <6001>
secret=
host=dynamic
defaultip=199.242.227.227
canreinvite=no
mailbox=6001
subscribecontext=sip

[pjf]
type=friend
username=pjf
context=sip
callerid="Patrick" <6003>
secret=
host=dynamic
defaultip=199.242.227.227
canreinvite=no
mailbox=6003
subscribecontext=sip

360 configuration:
fkey6!: dest 
fkey7!: dest 

extensions.conf:
[macro-oneline]
exten => s,1,Dial(${ARG1},20,t)
exten => s,2,Voicemail(u${MACRO_EXTEN})
exten => s,3,Hangup
exten => s,102,Voicemail(b${MACRO_EXTEN})
exten => s,103,Hangup

exten => 6001,hint,SIP/mjg
exten => 6001,1,Macro(oneline,${MJG})

exten => 6003,hint,SIP/pjf
exten => 6003,1,Macro(oneline,${PJF})

 Is there any convenient way to monitor the status of my FXO lines from 
the phones?  Or do I have to set up the interested parties with gastman?


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[Asterisk-Users] SNOM 360 and parking

[Patrick Friedel]

OK, last showstopper that I just can't puzzle my way through - parking 
calls with the snom phones.  I get the two phones connected, I hit 
transfer on one, the other phone goes to MOH and the first phone gives 
me DT, so I dial 700 and hit the OK button.  Call transferred, the SNOM 
hangs up before I have a chance to hear which extension it parked to.  
Is there a way to make the SNOM phones stay off hook until you 
explicitly hang up during a transfer?  (my only complaint about these 
phones - occasionally they're just too darn smart for their own good.)


I can live without actual snom-style orbits at this time (handy though 
they might be), since the current system involves parking the call on an 
external line and walking over to another office to say that they have a 
call.  I imagine that down the road it'll usually just be an attended 
transfer, but we do park calls around phones a fair bit as we brainstorm 
issues.


(Actually, I can't get attended transfers working, either.  All 
transfers are blind.  Related?)


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[Asterisk-Users] snom 360 audio garbled

[Michael George]

I have a new snom 360 on an internal net to my * box.  When putting a call on
hold and taking it off, the audio will usually be broken and not
understandable.

Sometimes this happens on incoming calls and almost always on outgoing calls.

Anyone run into this before?

Thx!

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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[Asterisk-Users] Snom 360 record button?

[Patrick Friedel]

Sorry if this is an obvious question, but I haven't seen an obvious 
answer on the wiki that I remember.  Has anyone managed to make the 
record button on the snom 360 fire off the Monitor() application?  I 
don't see a bounty, and googling for "snom 360 record button asterisk" 
returns tons of product specification pages. (Joy!) I don't see a bounty 
for it, and the only mention I _see_ on the wiki is "one touch RECORD 
button usuable only with special PBX support via SIP INFO method" which 
isn't much of an answer.


I assume the answer is no because of this, but I'm hoping against hope 
this is just because I don't have anything set up for it:


SIP Debugging Enabled
voip*CLI>
*[I hit the record button at this point]*
Sip read:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-q6cqmwneki97;rport
From: "Patrick" ;tag=1pvw4rlq7s
To: ;tag=as51ba7d7b
Call-ID: [EMAIL PROTECTED]
CSeq: 4 INFO
Max-Forwards: 70
Contact: 
User-Agent: snom360/3.60r
Record: on
Content-Length: 0


11 headers, 0 lines
Receiving DTMF!
Jul 27 17:24:52 WARNING[26025]: chan_sip.c:6166 receive_info: Unable to 
parse INFO message from [EMAIL PROTECTED] Content

Transmitting (no NAT):
SIP/2.0 415 Unsupported media type
Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-q6cqmwneki97
From: "Patrick" ;tag=1pvw4rlq7s
To: ;tag=as51ba7d7b
Call-ID: [EMAIL PROTECTED]
CSeq: 4 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


to 10.0.1.213:2051

 I assume the first one is the snom requesting the PBX to start 
recording, and the second is asterisk reminding the snom that it doesn't 
allow the INFO method and to get bent?



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[Asterisk-Users] SNOM 360 locked up

[Steven Ringwald]

Hello all!

I was trying to get the dial-string setup for my regular usage, and the 
phone locked up in the middle of dialing. Basically, I put the following 
line in, hit save, and got as far as dialing '9', and the phone froze.


|^(9[0-9]{10}|sip:[EMAIL PROTECTED]|d

Now the phone boots up to the SNOM splash screen and hangs there. I can 
ping it, but cannot get to the web-interface and cannot reset to factory 
defaults using the web-gui.


Any idea how I can reset the phone to factory w/o using the GUI? Or am I 
completely hosed?


Steve




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[Asterisk-Users] Snom 360 and hints

[Paul Hales]

I am setting up a snom 360, and the lights come on OK when the mapped
user makes an outgoing call, but when the user takes an incoming call
the light does not come on.

I do not want to install the bristuff patch if possible.
(although I can see that with the devstate command I can make the lights
do whatever I want)

Any ideas?

regards,

Paul Hales
Melbourne, Australia

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[Asterisk-Users] Snom 360 hold problem

[Michael George]

Hello,

I have a customer who said that their Snom 360 is joining calls by accident.

The situation is that they had one call on the line and another call came in.
She pressed the hold button on the phone and the two calls were joined
together.

I do have "Call join on Xfer" set to yes, but I thought that would only come
into play when doing a transfer, not putting someone on hold.

The phone is at firmware 4.1, and there are no new updates, so that shouldn't
be it.

Anyone else experience this behavior on the phones, or know if I need to turn
off Call Join on Xfer?

Thanks!

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] Snom 360 problem

[Nils Ohlmeier]

Did you tried to turn off te option "Filter Packets from Registrar" on the 
Advanced page?
That is the only case, when a snom sends a 403 (= Forbidden).

Regards
  Nils Ohlmeier

On Friday 02 September 2005 09:11, altus wrote:
> Good day all
> I have asterisk on a box with one network card
> I have a 2 companies setup on the system.
> To keep all apart I binded a different ip to the interface,i,o,w eth0
> 192.168.0.254 and eth0:1 192.168.1.254
> And in sip.conf I took the bind setting out
> So each company's phones are on a different ip range,and all worked well
> So we decide to pull the snom190 out and exchange it with a snom360
> This company is on the virtual interface(eth0:1)
> The 360 register and can make outgoing calls fine
> But when you try to make a call to it it does not work?
> I gives this error in the cli
> "Forbidden - wrong password on authentication for INVITE to '"301"
> ;tag=as3405ec0a"
> But its 301 calling the snom360(user 310)???
> BUT
> If I change the phone's ip and tell it to connect to eth0,not eth0:1 it
> works,same account settings same everything?
> The snom190 worked this way
> Any Ideas why?

-- 
snom technology AGGradestr. 46D-12347 Berlin
Nils Ohlmeier
mailto:[EMAIL PROTECTED]  http://www.snom.com
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[asterisk-users] Snom 360 Function Keys

[Alessandro De Filippo]

I have a Snom 360 phone and I'm configuring it for use with Asterisk 
1.2.9 and Freepbx 2.1.1


On my PBX there are:
1) Some SIP phones
2) One digium quadri primary ISDN interface (TE410P)
3) Two Rhino Channel Banks
4) 25 Analogue Phones on every channel bank

How I can configure function keys on my SNOM 360 for monitoring analogue 
phone status?


Configure sip phones is very simple (just put in function keys panel the 
SIP URI of every phone) but I have same problems with analogue phones!


Someone have the same problem?
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Re: [Asterisk-Users] Snom 360 Opinions

[Remco Barende]

I'm not too pleased with the phones, I have about 40 of them, some of the 
displays tend to die and the dial pad feels to 'mushy' IMHO, just like the 
keys on a good old ZX80 computer


Also I'm having some issues with sound quality on some phones, but I still 
need to switch some phones to see if that is really an issue of the phone.


Also if you want to use * call files, with the 360 you will run into a big 
where the call is being redialled as if it failed while in fact the call 
is ongoing. Annoying and haven't found out if that is an * bug or Snom 
bug. The Snom 190's do not have this problem.


Just my $0.02 (which is really not a lot these days!) :)

On Sat, 12 Nov 2005, Curren C. Calhoun wrote:


I¹m looking to add in some Snom 360 phones, could anyone give thoughts or
opinions about the speakerphone, general quality... Also the phone would
need to be powered over Ethernet...

I like some of the listed features and the expandability of the phone but am
open to any other suggestions as well...

Thanks


Curren
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Re: [Asterisk-Users] Snom 360 Opinions

[Andrew Latham]

Cleaner looking, 12 line apperances, affordable sidecar, runs on
linux, developing XML services, very programable, buttons are firm in
a good way, simple layout for users

installing 30 right now...


On 11/12/05, Remco Barende <[EMAIL PROTECTED]> wrote:
> I'm not too pleased with the phones, I have about 40 of them, some of the
> displays tend to die and the dial pad feels to 'mushy' IMHO, just like the
> keys on a good old ZX80 computer
>
> Also I'm having some issues with sound quality on some phones, but I still
> need to switch some phones to see if that is really an issue of the phone.
>
> Also if you want to use * call files, with the 360 you will run into a big
> where the call is being redialled as if it failed while in fact the call
> is ongoing. Annoying and haven't found out if that is an * bug or Snom
> bug. The Snom 190's do not have this problem.
>
> Just my $0.02 (which is really not a lot these days!) :)
>
> On Sat, 12 Nov 2005, Curren C. Calhoun wrote:
>
> > I¹m looking to add in some Snom 360 phones, could anyone give thoughts or
> > opinions about the speakerphone, general quality... Also the phone would
> > need to be powered over Ethernet...
> >
> > I like some of the listed features and the expandability of the phone but am
> > open to any other suggestions as well...
> >
> > Thanks
> >
> >
> > Curren
> >
>
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>


--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
---
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Re: [Asterisk-Users] Snom 360 Opinions

[Curren C. Calhoun]

Thanks that's the type of info I'm looking for...

I've heard some early grumblings but wanted to see if anything else has come
up...


> From: Remco Barende <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Sat, 12 Nov 2005 09:50:01 +0100 (CET)
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [Asterisk-Users] Snom 360 Opinions
> 
> I'm not too pleased with the phones, I have about 40 of them, some of the
> displays tend to die and the dial pad feels to 'mushy' IMHO, just like the
> keys on a good old ZX80 computer
> 
> Also I'm having some issues with sound quality on some phones, but I still
> need to switch some phones to see if that is really an issue of the phone.
> 
> Also if you want to use * call files, with the 360 you will run into a big
> where the call is being redialled as if it failed while in fact the call
> is ongoing. Annoying and haven't found out if that is an * bug or Snom
> bug. The Snom 190's do not have this problem.
> 
> Just my $0.02 (which is really not a lot these days!) :)
> 
> On Sat, 12 Nov 2005, Curren C. Calhoun wrote:
> 
>> I¹m looking to add in some Snom 360 phones, could anyone give thoughts or
>> opinions about the speakerphone, general quality... Also the phone would
>> need to be powered over Ethernet...
>> 
>> I like some of the listed features and the expandability of the phone but am
>> open to any other suggestions as well...
>> 
>> Thanks
>> 
>> 
>> Curren
>> 


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Re: [Asterisk-Users] Snom 360 Opinions

[Omar A. Sabek]

Curren,

Can you tell us a little more about the environment you are deploying these phones in, how many phones and what kind of setup?

Omar A. SabekOn 11/12/05, Curren C. Calhoun <[EMAIL PROTECTED]> wrote:
Thanks that's the type of info I'm looking for...I've heard some early grumblings but wanted to see if anything else has comeup...> From: Remco Barende <
[EMAIL PROTECTED]>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion> <asterisk-users@lists.digium.com>> Date: Sat, 12 Nov 2005 09:50:01 +0100 (CET)
> To: Asterisk Users Mailing List - Non-Commercial Discussion> <asterisk-users@lists.digium.com>> Subject: Re: [Asterisk-Users] Snom 360 Opinions
>> I'm not too pleased with the phones, I have about 40 of them, some of the> displays tend to die and the dial pad feels to 'mushy' IMHO, just like the> keys on a good old ZX80 computer>
> Also I'm having some issues with sound quality on some phones, but I still> need to switch some phones to see if that is really an issue of the phone.>> Also if you want to use * call files, with the 360 you will run into a big
> where the call is being redialled as if it failed while in fact the call> is ongoing. Annoying and haven't found out if that is an * bug or Snom> bug. The Snom 190's do not have this problem.>
> Just my $0.02 (which is really not a lot these days!) :)>> On Sat, 12 Nov 2005, Curren C. Calhoun wrote:>>> I¹m looking to add in some Snom 360 phones, could anyone give thoughts or
>> opinions about the speakerphone, general quality... Also the phone would>> need to be powered over Ethernet...>>>> I like some of the listed features and the expandability of the phone but am
>> open to any other suggestions as well...>>>> Thanks>>>>>> Curren>>___--Bandwidth and Colocation sponsored by 
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[Asterisk-Users] Snom 360 Passsword Issue

[Edward de Zeeuw]

I have multiple (20+) Snom 360 phones communicating with asterisk
1.2.7.1.  Almost regularly (daily) and in some cases ongoing 9every 10
minutes the phones ask for password and id the account they are seeking
the password for.  If I hit the X key the phone continues operating
normally.  Has anyone else come across a similar issue?

Edward
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RE: [Asterisk-Users] Snom 360 problems

[Guido Hecken]

> Anyone have a Snom they're happy with?   How did you manage that?  :)
> 
> I have a system of:
> 
> Asterisk 1.2.3
> 2 Wildcard TDM400P  Rev I and E/F
> 1 Snom 360 + sidecar
> ~15 Sipura/Linsys SPA-841
> ~15 Grandstream 101
> 
> Everything (currently) is on the same network, not a router to be seen
> between any two.  Also everything, except the snom, is working sweetly.
> 
> The main problem is ECHO.. awful echo and only on the Snom.  When using
> a Zap line or to another sip phone.  I've tweaked the * for echo and
> managed to only create echo and piss everyone else off, pounded the
> settings in the Snom trying to find anything, and updated the firmware
> to Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2
> v3.36 after noticing a changelog that sounded like it may have related
> to echo.  Not even a slight reduction in echo so far.
> 
> A second serious problem is Call join.   Even with "Call join on Xfer (2
> calls)" OFF if the user is doing a transfer of one call when a second
> starts ringing the 2 callers get bridged, no transfer.  Really nice, now
> I have two customers talking to each other with no clue what's going on
> and neither gets who they were trying to reach.
> 
> Any ideas on what I can try next?

This firmware works well for us: snom360-SIP 4.1 available here:
http://snom.com/download/share/snom360-4.1-SIP-j.bin
No echo and overall voice quality is excellent.

Did you check the codecs on the snom and on asterisk (sip.conf)?
Is Silence Suppression off on the snom?
If you would post your config (under settings on the snom) we could have a
closer look in the problem.

Regards, 

Guido
 
gwsNetTech
Guido Hecken

Quirrenbacher Str. 36
53639 Königswinter
Germany

fon +49(2244) 870663
fax +49(2244) 870664
mobil  +49(179) 1267353
web http://www.gwsnettech.de
mailto:[EMAIL PROTECTED]


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Re: [Asterisk-Users] Snom 360 problems

[Brian Kennedy]

Guido Hecken wrote:

  
Anyone have a Snom they're happy with?   How did you manage that?  :)

I have a system of:

Asterisk 1.2.3
2 Wildcard TDM400P  Rev I and E/F
1 Snom 360 + sidecar
~15 Sipura/Linsys SPA-841
~15 Grandstream 101

Everything (currently) is on the same network, not a router to be seen
between any two.  Also everything, except the snom, is working sweetly.

The main problem is ECHO.. awful echo and only on the Snom.  When using
a Zap line or to another sip phone.  I've tweaked the * for echo and
managed to only create echo for everyone else, pounded the
settings in the Snom trying to find anything, and updated the firmware
to Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2
v3.36 after noticing a changelog that sounded like it may have related
to echo.  Not even a slight reduction in echo so far.

A second serious problem is Call join.   Even with "Call join on Xfer (2
calls)" OFF if the user is doing a transfer of one call when a second
starts ringing the 2 callers get bridged, no transfer.  Really nice, now
I have two customers talking to each other with no clue what's going on
and neither gets who they were trying to reach.

Any ideas on what I can try next?

  
  
This firmware works well for us: snom360-SIP 4.1 available here:
http://snom.com/download/share/snom360-4.1-SIP-j.bin
No echo and overall voice quality is excellent.

Did you check the codecs on the snom and on asterisk (sip.conf)?
Is Silence Suppression off on the snom?
If you would post your config (under settings on the snom) we could have a
closer look in the problem.

  

I believe I'll try the 5.5.1b firmware as suggested in another response
for the second problem.  
yes, yes and attached. 

Thanks for your help.


language!: English
redirect_number!: 
redirect_busy_number!: 
redirect_event!: none
redirect_time!: 
phone_type!: 
codec_tos!: 160
mac&: 000413231FA6
setting_server!:
subscribe_config!: off
ip_adr!: 10.11.10.100
netmask!: 255.255.0.0
update_server!: 
dns_domain!: cincinnati
dns_server1!: 10.11.0.1
dns_server2!: 
dhcp!: off
gateway!: 10.11.0.1
phone_name!: 
utc_offset!: -18000
ntp_server!: 10.12.0.2
lcserver1!: 
ring_sound!: Ringer4
http_proxy!: 
http_port!: 80
http_user!: 
http_pass!: 
http_scheme!: off
https_port!: 443
webserver_type!: http_https
webserver_cert!: 
dst!: 3600 04.01.07 02:00:00 10.05.07 02:00:00
timezone!: USA-5
contrast!: 16
sip_retry_t1!: 500
session_timer!: 3600
network_id_port!: 
max_forwards!: 70
user_phone!: off
active_line!: 1
outgoing_identity!: 1
challenge_response!: on
refer_brackets!: off
sip_proxy!: 
register_http_contact!: off
cmc_feature!: off
filter_registrar!: off
challenge_reboot!: off
challenge_checksync!: off
action_dnd_on_url!: 
action_dnd_off_url!: 
action_redirection_on_url!: 
action_redirection_off_url!: 
action_incoming_url!: 
action_outgoing_url!: 
action_setup_url!: 
action_offhook_url!: 
action_onhook_url!: 
action_missed_url!: 
action_connected_url!: 
action_disconnected_url!: 
aoc_amount_display!: off
aoc_pulse_currency!: $
aoc_cost_pulse!: 1
rtp_port_start!: 49152
rtp_port_end!: 65534
preselection_nr!: 
auto_dial!: 5
dtmf_payload_type!: 101
dnd_mode!: on
privacy_in!: off
privacy_out!: off
admin_mode_login!: 
admin_mode_password!: 
admin_mode_password_confirm!: 
admin_mode!: on
tone_scheme!: USA
vol_speaker!: 1
vol_ringer!: 7
vol_handset!: 15
vol_headset!: 10
vol_speaker_mic!: 0
vol_handset_mic!: 1
vol_headset_mic!: 0
log_level!: 5
auto_connect_type!: auto_connect_type_handsfree
auto_connect_indication!: on
logon_wizard!: on
guess_number!: on
guess_start_length!: 4
friends_ring_sound!: Ringer4
family_ring_sound!: Ringer2
colleagues_ring_sound!: Ringer6
vip_ring_sound!: Ringer4
break_key!: false
publish_presence!: off
edit_alpha_mode!: 123
display_method!: display_name
call_waiting!: on
cw_dialtone!: on
disable_speaker!: off
no_dnd!: off
mute!: off
dirty_host_ttl!: 
headset_device!: none
update_policy!: never_update
conf_hangup!: on
enum_suffix!: e164.arpa
mwi_notification!: silent
vlan!: 
vlan_id!: 
vlan_qos!: 
block_url_dialing!: off
release_sound!: off
deny_all_feature!: off
transfer_on_hangup!: on
ethernet_replug!: nothing
mwi_dialtone!: stutter
support_idna!: off
custom_melody_url!: 
ringer_headset_device!: speaker
dtmf_speaker_phone!: off
presence_timeout!: 15
require_prack!: off
offer_gruu!: on
offer_mpo!: off
firmware_status!: 
firmware_interval!: 
firmware!: http://snom.com/download/snom360-ramdiskToJffs2-3.36-br.bin
bootloader!: 
update_filename!: 
update_host_b!: 
update_host_f!: 
sip_port!: 2051
web_language!: English
call_completion!: off
callpickup_dialoginfo!: on
use_backlight!: on
reset_settings!: 
date_us_format!: on
time_24_format!: off
call_join_xfer!: off
alert_info_playback!: on
ringing_time!: 60
silence_compression!: off
syslog_server!: 
screen_saver_timeout!: 60
intercom_enabled!: off
with_flash!: on
snmp_trusted_addresses!: 
snmp_port!: 161
short_form!: off
audio_device_indicator!: on
license_data&: Mac:000413231F

Re: [Asterisk-Users] Snom 360 problems

[asterisk]

On Fri, 24 Mar 2006, Brian Kennedy wrote:

Anyone have a Snom they're happy with?   How did you manage that?  :)


I would be happier if snom fixed the US indications and the giant 3000 
point font they use for everything.


-Dan
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Re: [Asterisk-Users] Snom 360 problems

[Brian Kennedy]

I may need a consultant that can help with some problems.

We had 2 smaller satellite offices on the same asterisk systems with no 
trouble.  We've just upgraded the main office and hit troubles.  There's 
no going back because we've entirely outgrown our old system.


Any * consultants around Cincinnati want a peek?

Brian Kennedy wrote:


Anyone have a Snom they're happy with?   How did you manage that?  :)

I have a system of:

Asterisk 1.2.3
2 Wildcard TDM400P  Rev I and E/F
1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841
~15 Grandstream 101

Everything (currently) is on the same network, not a router to be seen 
between any two.  Also everything, except the snom, is working sweetly.


The main problem is ECHO.. awful echo and only on the Snom.  When 
using a Zap line or to another sip phone.  I've tweaked the * for echo 
and managed to only create echo and piss everyone else off, pounded 
the settings in the Snom trying to find anything, and updated the 
firmware to Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 
jffs2 v3.36 after noticing a changelog that sounded like it may have 
related to echo.  Not even a slight reduction in echo so far.


A second serious problem is Call join.   Even with "Call join on Xfer 
(2 calls)" OFF if the user is doing a transfer of one call when a 
second starts ringing the 2 callers get bridged, no transfer.  Really 
nice, now I have two customers talking to each other with no clue 
what's going on and neither gets who they were trying to reach.


Any ideas on what I can try next?

Thanks...
...Brian
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[asterisk-users] Snom 360 flickering screen

[Nick Hoffman]

Hi guys. I just bought and configured a Snom 360 and have noticed that the 
LCD is constantly flickering at a rate of ~10-15Hz (that's a guess). 
Either way, it's very distracting. Has anyone else encountered this 
before? Any solutions?

Cheers,
-- Nick
E: [EMAIL PROTECTED]
P: +61 7 5591 3588
F: +61 7 5591 6588

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RE: [asterisk-users] snom 360 auto answer

[Klaverstyn, David C]

This is my code (that I copied form somewhere) for paging a group of
phones.  By dialling 99 it will page phones 2101, 2102 and 2105.

 

Just include the context ext-paging in your dial plan and modify the
extension numbers and all should be good.

 

This works on Linksys Phones but should also work on Snoms.

 

I hope this helps you.

 

 

[ext-paging]

exten => PAGE2101,1,GotoIf($[ ${CALLERID(number)} = 2101 ]?skipself)

exten => PAGE2101,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)

exten => PAGE2101,n,Set(__ALERT_INFO=Ring Answer)

exten => PAGE2101,n,Set(__SIP_URI_OPTIONS=intercom=true)

exten => PAGE2101,n,Dial(SIP/2101,5)

exten => PAGE2101,n(skipself),Noop(Not paging originator)

 

exten => PAGE2102,1,GotoIf($[ ${CALLERID(number)} = 2102 ]?skipself)

exten => PAGE2102,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)

exten => PAGE2102,n,Set(__ALERT_INFO=Ring Answer)

exten => PAGE2102,n,Set(__SIP_URI_OPTIONS=intercom=true)

exten => PAGE2102,n,Dial(SIP/2102,5)

exten => PAGE2102,n(skipself),Noop(Not paging originator)

 

exten => PAGE2105,1,GotoIf($[ ${CALLERID(number)} = 2105 ]?skipself)

exten => PAGE2105,n,Set(__SIPADDHEADER=Call-Info: \;answer-after=0)

exten => PAGE2105,n,Set(__ALERT_INFO=Ring Answer)

exten => PAGE2105,n,Set(__SIP_URI_OPTIONS=intercom=true)

exten => PAGE2105,n,Dial(SIP/2105,5)

exten => PAGE2105,n(skipself),Noop(Not paging originator)

 

 

exten => Debug,1,Noop(dialstr is
LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]
aging)

exten =>
99,1,Page(LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]&LOCAL/PAGE
[EMAIL PROTECTED])

 

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Kim
Sent: Monday, 8 January 2007 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] snom 360 auto answer

 

Hi,

 

I'm testing paging using snom 360.

Can someone correct my dialplan?

 

Regards,

Jason.

 

==

;exten => _99,1,SIPAddHeader(Call-Info:

Answer-After=0)

;exten => _99,n,SIPAddHeader(Call-Info:

\;answer-after=0)

;exten => _99,n,Dial(SIP/${EXTEN:2})

 

exten => _99,1,Set(__SIPADDHEADER=Call-Info:

answer-after=0)

exten =>

_99,n,Set(__SIP_URI_OPTIONS=intercom=true)

exten => _99,n,Set(__ALERT_INFO=Ring Answer)

exten => _99,n,Dial(SIP/${EXTEN:2})

 

 

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RE: [asterisk-users] snom 360 auto answer

[Jason Kim]

Thankyou David,

It works for Linksys,but not for snom 360.
Do I need to change someting using web UI ?

--- "Klaverstyn, David C"
<[EMAIL PROTECTED]> wrote:

> This is my code (that I copied form somewhere) for
> paging a group of
> phones.  By dialling 99 it will page phones 2101,
> 2102 and 2105.
> 
>  
> 
> Just include the context ext-paging in your dial
> plan and modify the
> extension numbers and all should be good.
> 
>  
> 
> This works on Linksys Phones but should also work on
> Snoms.
> 
>  
> 
> I hope this helps you.
> 
>  
> 
>  
> 
> [ext-paging]
> 
> exten => PAGE2101,1,GotoIf($[ ${CALLERID(number)} =
> 2101 ]?skipself)
> 
> exten => PAGE2101,n,Set(__SIPADDHEADER=Call-Info:
> \;answer-after=0)
> 
> exten => PAGE2101,n,Set(__ALERT_INFO=Ring Answer)
> 
> exten =>
> PAGE2101,n,Set(__SIP_URI_OPTIONS=intercom=true)
> 
> exten => PAGE2101,n,Dial(SIP/2101,5)
> 
> exten => PAGE2101,n(skipself),Noop(Not paging
> originator)
> 
>  
> 
> exten => PAGE2102,1,GotoIf($[ ${CALLERID(number)} =
> 2102 ]?skipself)
> 
> exten => PAGE2102,n,Set(__SIPADDHEADER=Call-Info:
> \;answer-after=0)
> 
> exten => PAGE2102,n,Set(__ALERT_INFO=Ring Answer)
> 
> exten =>
> PAGE2102,n,Set(__SIP_URI_OPTIONS=intercom=true)
> 
> exten => PAGE2102,n,Dial(SIP/2102,5)
> 
> exten => PAGE2102,n(skipself),Noop(Not paging
> originator)
> 
>  
> 
> exten => PAGE2105,1,GotoIf($[ ${CALLERID(number)} =
> 2105 ]?skipself)
> 
> exten => PAGE2105,n,Set(__SIPADDHEADER=Call-Info:
> \;answer-after=0)
> 
> exten => PAGE2105,n,Set(__ALERT_INFO=Ring Answer)
> 
> exten =>
> PAGE2105,n,Set(__SIP_URI_OPTIONS=intercom=true)
> 
> exten => PAGE2105,n,Dial(SIP/2105,5)
> 
> exten => PAGE2105,n(skipself),Noop(Not paging
> originator)
> 
>  
> 
>  
> 
> exten => Debug,1,Noop(dialstr is
>
LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]
> aging)
> 
> exten =>
>
99,1,Page(LOCAL/[EMAIL PROTECTED]&LOCAL/[EMAIL PROTECTED]&LOCAL/PAGE
> [EMAIL PROTECTED])
> 
>  
> 
>  
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
> Behalf Of Jason Kim
> Sent: Monday, 8 January 2007 2:30 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] snom 360 auto answer
> 
>  
> 
> Hi,
> 
>  
> 
> I'm testing paging using snom 360.
> 
> Can someone correct my dialplan?
> 
>  
> 
> Regards,
> 
> Jason.
> 
>  
> 
> ==
> 
> ;exten => _99,1,SIPAddHeader(Call-Info:
> 
> Answer-After=0)
> 
> ;exten => _99,n,SIPAddHeader(Call-Info:
> 
> \;answer-after=0)
> 
> ;exten => _99,n,Dial(SIP/${EXTEN:2})
> 
>  
> 
> exten => _99,1,Set(__SIPADDHEADER=Call-Info:
> 
> answer-after=0)
> 
> exten =>
> 
> _99,n,Set(__SIP_URI_OPTIONS=intercom=true)
> 
> exten => _99,n,Set(__ALERT_INFO=Ring Answer)
> 
> exten => _99,n,Dial(SIP/${EXTEN:2})
> 
>  
> 
>  
> 
> __
> 
> Do You Yahoo!?
> 
> Tired of spam?  Yahoo! Mail has the best spam
> protection around 
> 
> http://mail.yahoo.com 
> 
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Re: [asterisk-users] SNOM 360 Rejecting Calls

[Philipp Kempgen]

Hi,

Is this a real question? I'm asking because it seems to be a
babelfish translation of a post in a german forum
http://www.ip-phone-forum.de/archive/index.php/t-99696.html
from over a year ago. (is their date wrong?)

Dovid B wrote:

>  Does someone know coincidentally the cause for the error message specified 
> in the Subject?

Probably misconfiguration. ;)

> The following scenario: Snom 360 behind one rout (wiederrum on a DSL line 
> with static IP address hangs). The Snom has a private IP, routs accomplishes 
> NAT. STUN and ICE are activated, as SIP haven 5060/udp are firmly used. 
> Detailed packages passed on on haven 5060/udp of rout to the Snom.
> 
> The telephone registers itself as expected, and outgoing telephone calls can 
> be led problem-free. Detailed telephone calls however do not function (a call 
> is signaled, which rings then however after 3 time on the mailbox is sent).
> 
> The SIP log shows that the telephone sees the INVITE of the Registrar, it 
> however in principle with "486 Busy here" answered (that is then also the 
> reason, why the detailed call is sent on the mailbox). The message mentioned 
> "Denying call appears contemporaneous id=X reason=unconditional" in the log, 
> whereby X was so far always a negative, one-digit number.
> 
> Does someone have an idea?

Sounds like a bug, I'd suggest filing a bug report on both the
Snom and the Digium issue trackers with severity set to major
(since you can't make any calls at all!). ;)

No, seriously now:

I would suspect you have activated some kind of call forwarding rule
or DND on the Snom (have a look at the prefs.htm page on the built in
web server). Which version of the firmware are you using? Did you
try to reset the phone to it's factory defaults?

Does ist work without NAT? How does your sip.conf and
extensions.conf look like? What's the output of
asterisk -vvvr
?

Regards
  Philipp

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
 Let's use IT to solve problems and not to create new ones.
   Asterisk? -> http://www.das-asterisk-buch.de

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Re: [asterisk-users] SNOM 360 Rejecting Calls

[Dovid B]

I was super super tired when I sent it. I had the same problem as the posted 
from http://www.ip-phone-forum.de/archive/index.php/t-99696.html and he had 
no response to his issue. So I reposted here. Resetting the phone took care 
of the issue. After being up for 4 days straight you have virtually no brain 
cells left. Thanks for the idea to rest it (don't know why I didn't think of 
that).


- Original Message - 
From: "Philipp Kempgen" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Friday, May 11, 2007 2:19 AM
Subject: Re: [asterisk-users] SNOM 360 Rejecting Calls


Hi,

Is this a real question? I'm asking because it seems to be a
babelfish translation of a post in a german forum
http://www.ip-phone-forum.de/archive/index.php/t-99696.html
from over a year ago. (is their date wrong?)

Dovid B wrote:

 Does someone know coincidentally the cause for the error message 
specified in the Subject?


Probably misconfiguration. ;)

The following scenario: Snom 360 behind one rout (wiederrum on a DSL line 
with static IP address hangs). The Snom has a private IP, routs 
accomplishes NAT. STUN and ICE are activated, as SIP haven 5060/udp are 
firmly used. Detailed packages passed on on haven 5060/udp of rout to the 
Snom.


The telephone registers itself as expected, and outgoing telephone calls 
can be led problem-free. Detailed telephone calls however do not function 
(a call is signaled, which rings then however after 3 time on the mailbox 
is sent).


The SIP log shows that the telephone sees the INVITE of the Registrar, it 
however in principle with "486 Busy here" answered (that is then also the 
reason, why the detailed call is sent on the mailbox). The message 
mentioned "Denying call appears contemporaneous id=X reason=unconditional" 
in the log, whereby X was so far always a negative, one-digit number.


Does someone have an idea?


Sounds like a bug, I'd suggest filing a bug report on both the
Snom and the Digium issue trackers with severity set to major
(since you can't make any calls at all!). ;)

No, seriously now:

I would suspect you have activated some kind of call forwarding rule
or DND on the Snom (have a look at the prefs.htm page on the built in
web server). Which version of the firmware are you using? Did you
try to reset the phone to it's factory defaults?

Does ist work without NAT? How does your sip.conf and
extensions.conf look like? What's the output of
asterisk -vvvr
?

Regards
 Philipp

--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [Asterisk-Users] SNOM 360 extension lights

[Olle E. Johansson]

Ross Kevlin wrote:
> I recently got a SNOM 360 and have been trying to get the extension
> lights to work. I can see the subscriptions with sip show subscriptions
> but I don't see any notifies when a call is made. I must be missing
> something because I've tried looking to see if anyone else has had this
> problem but the only solutions I've seen have been to put hints in and I
> have those. Any suggestions?
> 
There is a page on the wiki that subscribes this. If you have a proper
subscription and you have the hints, it "should" work as we usually say
in this business when we don't know what the problem is. If you are
using CVS head, you can limit the context reachable for subscriptions
with the "subscribecontext" setting. I haven't tried with the 360 yet,
so I might have to power up that phone.

For those of you that haven't explored the subscription support in the
Asterisk SIP channel:

The key to get device state notification in SIP subscriptions and the AMI,
the Asterisk manager interface, is the use of a "hint" priority. The
phone subscribe to an extension, but in order for the PBX to know which
phone that is connected to an extension, you need to tell Asterisk what
relationship you have between an extension and a device.

exten => 3000,hint,SIP/olle
exten => 3000,dial(SIP/olle,30)

This extension in your dialplan tells Asterisk that if anyone subscribes
to extension 3000, they want to know the status of SIP/olle. Without the
hint, the extension will always be available and there's no notification
at all.

In CVS head, you can do this with IAX2 as well. I have a patch that has
been in the bugtracker for a few months that adds a bit more. If you
apply a call limit with the incominglimit, you will see that the notify
function will tell you not only if the device is available or not, but
also if they're on a call. This works beautifully with Xten's Eyebeam.

I am close to securing funding for development of another addition to
chan_sip during the summer. This will be the "shared line apperances"
solution developed by Broadsoft and now implemented in phones from many
manufacturers like Sipura, Aastra and I also suspect Grandstream and
Polycom. When we have this working, you will be able to get Asterisk
with one of these SIP phones to behave much more like a key system.

A lot of text that really did not answer the question, sorry.

/Olle


Astricon - the Asterisk User's conference - Madrid June 15-17
http://www.astricon.net/europe/ - Register today!
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Re: [Asterisk-Users] SNOM 360 extension lights

[Matias G.]

http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom
 
remember always to search the wiki it has tons of 
info...
 
I'm using a 360 and the lights work fine. Pay 
attention to the hint stuff in the dialplan.
 
bye,
M.

  - Original Message - 
  From: 
  Ross 
  Kevlin 
  To: asterisk-users@lists.digium.com 
  
  Sent: Friday, June 03, 2005 12:35 
PM
  Subject: [Asterisk-Users] SNOM 360 
  extension lights
  
  I contacted SNOM and they told me to change a 
  couple of options but still no lights, here is what they told me
   
  Line page SIP tab:o Long SIP-Contact (RFC3840) to "off"o 
  Support broken Registrar to "on"Advanced page:o Filter Packets 
  from Registrar to "off"
   
  And please ask the Asterisk community for help, I'm sure they solved 
  thatissue 100%, and we are not knowing so much about Asterisk.Your 
  snom support Teamhas anyone gotten a 360 to 
  work with the lights? what options and modifications to .conf files did you 
  have to make?
   
  here are the subscribe and notifies.
  it seems it terminates the subscription as soon 
  as its created. I don't think its a proxy authentication problem
  because it eventually sends the proxy 
  authentication information
   
  Using latest SUBSCRIBE request as basis 
  requestSending to 192.168.2.230 : 2051 (non-NAT)Found peer 
  '83'Transmitting (no NAT) to 192.168.2.230:2051:SIP/2.0 407 Proxy 
  Authentication RequiredVia: SIP/2.0/UDP 
  192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049nFrom: 
  ;tag=z6kvtd67buTo: 
  ;tag=as6c1cb2a5Call-ID: [EMAIL PROTECTED]CSeq: 
  1 SUBSCRIBEUser-Agent: MVC 001Allow: INVITE, ACK, CANCEL, OPTIONS, 
  BYE, REFER, NOTIFYContact: 
  Proxy-Authenticate: Digest realm="asterisk", 
  nonce="16747f76"Content-Length: 0
   
  ---Scheduling destruction of call '[EMAIL PROTECTED]' 
  in 15000 mssip1*CLI><-- SIP read from 
  192.168.2.230:2051:SUBSCRIBE sip:[EMAIL PROTECTED];user=phone 
  SIP/2.0Via: SIP/2.0/UDP 
  192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n;rportFrom: 
  ;tag=z6kvtd67buTo: 
  Call-ID: [EMAIL PROTECTED]CSeq: 
  1 SUBSCRIBEMax-Forwards: 70Contact: 
  Event: dialogAccept: 
  application/dialog-info+xmlExpires: 3600Content-Length: 0
   
  --- (12 headers 0 lines)---Ignoring this SUBSCRIBE 
  requestFound peer '83'Transmitting (no NAT) to 
  192.168.2.230:2051:SIP/2.0 407 Proxy Authentication RequiredVia: 
  SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049nFrom: 
  ;tag=z6kvtd67buTo: 
  ;tag=as6c1cb2a5Call-ID: [EMAIL PROTECTED]CSeq: 
  1 SUBSCRIBEUser-Agent: MVC 001Allow: INVITE, ACK, CANCEL, OPTIONS, 
  BYE, REFER, NOTIFYContact: 
  Proxy-Authenticate: Digest realm="asterisk", 
  nonce="16747f76"Content-Length: 0
   
  ---Scheduling destruction of call '[EMAIL PROTECTED]' 
  in 15000 mssip1*CLI><-- SIP read from 
  192.168.2.230:2051:SUBSCRIBE sip:[EMAIL PROTECTED];user=phone 
  SIP/2.0Via: SIP/2.0/UDP 
  192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x;rportFrom: 
  ;tag=z6kvtd67buTo: 
  Call-ID: [EMAIL PROTECTED]CSeq: 
  2 SUBSCRIBEMax-Forwards: 70Contact: 
  Event: dialogAccept: 
  application/dialog-info+xmlProxy-Authorization: Digest 
  username="83",realm="asterisk",nonce="16747f76",uri="sip:[EMAIL PROTECTED];user=phone",response="15d72104244317e2c0afa3499220e4ab",algorithm=md5Expires: 
  3600Content-Length: 0
   
  --- (13 headers 0 lines)---Using latest SUBSCRIBE request as 
  basis requestSending to 192.168.2.230 : 2051 (non-NAT)Found peer 
  '83'Looking for 117 in localusers-C2021-1Transmitting (no NAT) to 
  192.168.2.230:2051:SIP/2.0 200 OKVia: SIP/2.0/UDP 
  192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7xFrom: 
  ;tag=z6kvtd67buTo: 
  ;tag=as77c7b911Call-ID: [EMAIL PROTECTED]CSeq: 
  2 SUBSCRIBEUser-Agent: MVC 001Allow: INVITE, ACK, CANCEL, OPTIONS, 
  BYE, REFER, NOTIFYExpires: 3600Contact: 
  ;expires=3600Content-Length: 0
   
  ---Scheduling destruction of call '[EMAIL PROTECTED]' 
  in 361 msReliably Transmitting (no NAT) to 
  192.168.2.230:2051:NOTIFY sip:[EMAIL PROTECTED] SIP/2.0Via: SIP/2.0/UDP 
  192.168.2.252:5060;branch=z9hG4bK56396cd7;rportFrom: 
  ;tag=as77c7b911To: 
  ;tag=z6kvtd67buContact: 
  Call-ID: [EMAIL PROTECTED]CSeq: 
  102 NOTIFYUser-Agent: MVC 001Event: dialogContent-Type: 
  application/dialog-info+xmlContent-Length: 203
   
   entity="sip:[EMAIL PROTECTED]">terminated
   
  ---sip1*CLI><-- SIP read from 
  192.168.2.230:2051:SIP/2.0 200 OkVia: SIP/2.0/UDP 
  192.168.2.252:5060;branch=z9hG4bK56396cd7;rport=5060From: 
  ;tag=as77c7b911To: 
  ;tag=z6kvtd67buCall-ID: [EMAIL PROTECTED]CSeq: 
  102 NOTIFYContent-Length: 0
  
  

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Re: [Asterisk-Users] SNOM 360 extension lights

[David John Walsh]

Sorry Ross I must have missed your first postings, but what are you
trying to achive?

David

On 03/06/05, Ross Kevlin <[EMAIL PROTECTED]> wrote:
>  
> I contacted SNOM and they told me to change a couple of options but still no
> lights, here is what they told me 
>   
> Line page SIP tab:
> 
> o Long SIP-Contact (RFC3840) to "off"
> o Support broken Registrar to "on"
> 
> Advanced page:
> 
> o Filter Packets from Registrar to "off" 
>   
> And please ask the Asterisk community for help, I'm sure they solved that
> issue 100%, and we are not knowing so much about Asterisk.
> 
> Your snom support Team
> 
> has anyone gotten a 360 to work with the lights? what options and
> modifications to .conf files did you have to make? 
>   
> here are the subscribe and notifies. 
> it seems it terminates the subscription as soon as its created. I don't
> think its a proxy authentication problem 
> because it eventually sends the proxy authentication information 
>   
> Using latest SUBSCRIBE request as basis request
> Sending to 192.168.2.230 : 2051 (non-NAT)
> Found peer '83'
> Transmitting (no NAT) to 192.168.2.230:2051:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n
> From: ;tag=z6kvtd67bu
> To: ;tag=as6c1cb2a5
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 SUBSCRIBE
> User-Agent: MVC 001
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: 
> Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76"
> Content-Length: 0 
>   
> 
> ---
> Scheduling destruction of call
> '[EMAIL PROTECTED]' in 15000 ms
> sip1*CLI>
> <-- SIP read from 192.168.2.230:2051:
> SUBSCRIBE sip:[EMAIL PROTECTED];user=phone SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n;rport
> From: ;tag=z6kvtd67bu
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 SUBSCRIBE
> Max-Forwards: 70
> Contact: 
> Event: dialog
> Accept: application/dialog-info+xml
> Expires: 3600
> Content-Length: 0 
>   
> 
> --- (12 headers 0 lines)---
> Ignoring this SUBSCRIBE request
> Found peer '83'
> Transmitting (no NAT) to 192.168.2.230:2051:
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n
> From: ;tag=z6kvtd67bu
> To: ;tag=as6c1cb2a5
> Call-ID: [EMAIL PROTECTED]
> CSeq: 1 SUBSCRIBE
> User-Agent: MVC 001
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Contact: 
> Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76"
> Content-Length: 0 
>   
> 
> ---
> Scheduling destruction of call
> '[EMAIL PROTECTED]' in 15000 ms
> sip1*CLI>
> <-- SIP read from 192.168.2.230:2051:
> SUBSCRIBE sip:[EMAIL PROTECTED];user=phone SIP/2.0
> Via: SIP/2.0/UDP
> 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x;rport
> From: ;tag=z6kvtd67bu
> To: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 SUBSCRIBE
> Max-Forwards: 70
> Contact: 
> Event: dialog
> Accept: application/dialog-info+xml
> Proxy-Authorization: Digest
> username="83",realm="asterisk",nonce="16747f76",uri=
> "sip:[EMAIL 
> PROTECTED];user=phone",response="15d72104244317e2c0afa3499220e4ab",a
> lgorithm=md5
> Expires: 3600
> Content-Length: 0 
>   
> 
> --- (13 headers 0 lines)---
> Using latest SUBSCRIBE request as basis request
> Sending to 192.168.2.230 : 2051 (non-NAT)
> Found peer '83'
> Looking for 117 in localusers-C2021-1
> Transmitting (no NAT) to 192.168.2.230:2051:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x
> From: ;tag=z6kvtd67bu
> To: ;tag=as77c7b911
> Call-ID: [EMAIL PROTECTED]
> CSeq: 2 SUBSCRIBE
> User-Agent: MVC 001
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
> Expires: 3600
> Contact: ;expires=3600
> Content-Length: 0 
>   
> 
> ---
> Scheduling destruction of call
> '[EMAIL PROTECTED]' in 361 ms
> Reliably Transmitting (no NAT) to 192.168.2.230:2051:
> NOTIFY sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport
> From: ;tag=as77c7b911
> To: ;tag=z6kvtd67bu
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 NOTIFY
> User-Agent: MVC 001
> Event: dialog
> Content-Type: application/dialog-info+xml
> Content-Length: 203 
>   
> 
>  version="0" state="full"
>  entity="sip:[EMAIL PROTECTED]">
> 
> terminated
> 
>  
>   
> ---
> sip1*CLI>
> <-- SIP read from 192.168.2.230:2051:
> SIP/2.0 200 Ok
> Via: SIP/2.0/UDP
> 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport=5060
> From: ;tag=as77c7b911
> To: ;tag=z6kvtd67bu
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 NOTIFY
> Content-Length: 0 
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Re: [Asterisk-Users] SNOM 360 extension lights

[Olle E. Johansson]

Ross Kevlin wrote:
>  I have a snom 360 that im trying to get the extension lights working i can
> see the subscription being sent and a reply but the reply is a terminate.
Terminate being?

/O
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Re: [Asterisk-Users] SNOM 360 extension lights

[Olle E. Johansson]

Ross Kevlin wrote:
> the subscription is sent a reply and the reply has content that indicates
> its state is terminated
> 
> From: ;tag=as77c7b911
>  To: ;tag=z6kvtd67bu
>  Contact: 
>  Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
>  CSeq: 102 NOTIFY
>  User-Agent: MVC 001
>  Event: dialog
>  Content-Type: application/dialog-info+xml
>  Content-Length: 203
> 
>  
>version="0" state="full"
>   entity="sip:83 at 192.168.2.252">
>  
>  terminated
>  
>  
> 
> from what I understand the terminated state is the end of dialog and the
> subscription should end, but I still see the subscription in asterisk.
> 
> 
>>Ross Kevlin wrote:
>> I have a snom 360 that i'm trying to get the extension lights working I
> 
> can
> 
>>see the subscription being sent and a reply but the reply is a terminate.
>>Terminate being?
> 
Please show us a full packet and tell us who sends it where?
Asterisk currently does only SEND notifications, we do not parse
incoming notifications.

/O
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[Asterisk-Users] Snom 360 Message Waiting indicator

[Joe Pukepail]

I'm having problems with the Message waiting indicator on my Snom 360 that I'm using for testing.   I got the button and message waiting indicator working, the problem is : when I hit the voicemail button (or use the menu on the display to access voicemail) it seems to clear the message waiting indicator on the phone.  So even if I don't go in and delete any messages it clears the light and seems like the light isn't updated until I get another voicemail.

 
Anyone else run into this?  Anyone else get it working properly?  
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[Asterisk-Users] snom 360 incorrect US indications

[asterisk]

Anyone noticed the snom 360 indications are incorrect for US zone?

menu->preferences->tone scheme->usa

indications.conf:
[general]
country=us

extensions.conf:
exten => ,1,Answer
exten => ,n,Playtones(dial)
exten => ,n,Wait(30)

exten => ,1,Busy

exten => ,1,Answer
exten => ,n,Playtones(busy)
exten => ,n,Wait(30)


hit speakerphone on the snom 360. listen to the dialtone.
now dial  and compare to asterisk's dialtone.

hit speakerphone on the snom 360. dial .
now compare to the busy signal you get from .

in each case, snom tone is incorrect and asterisk is correct.

-Dan
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Re: [Asterisk-Users] Snom 360 NOTIFY syntax

[Michael George]

On Mon, Jul 11, 2005 at 01:16:08PM -0500, Patrick Friedel wrote:
> I'm rolling out an installation with snom 360s in the near future.  
> Simple SOHO configuration, 3 FXOs hanging off a TDM400B, 4 snom 360s, a 
> snom 200, some variant of IAX softphone, and an IAXy or Sipura 2002.  I 
> have the 360's set up to subscribe and notify for the line use lights, 
> which works like a charm for interoffice calling (between the 360's, 
> anyway.  The IAXy, 200 and, softphone will be used by less phone 
> dependant types) but what I can't figure out from the Wiki is if it's 
> possible to have the ZAP lines notify for the outbound lines so we can 
> see how many lines are in use.

I am by no means an expert at this, but I did some experimentation and it
appears that the NOTIFY will not get sent for the trunk lines, only for
extensions.  I also found that the SUBSCRIBE/NOTIFY sequence only works for
SIP and ZAP, I couldn't get it to work for IAX2.

I do not know why this would be, and it is possible I was doing something
wrong, but for what it's worth, that's my experience so far.

-- 
-M

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Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] SNOM 360 and parking

[Nils Ohlmeier]

On Tuesday 12 July 2005 23:29, Patrick Friedel wrote:
> OK, last showstopper that I just can't puzzle my way through - parking
> calls with the snom phones.  I get the two phones connected, I hit
> transfer on one, the other phone goes to MOH and the first phone gives
> me DT, so I dial 700 and hit the OK button.  Call transferred, the SNOM
> hangs up before I have a chance to hear which extension it parked to.
> Is there a way to make the SNOM phones stay off hook until you
> explicitly hang up during a transfer?  (my only complaint about these
> phones - occasionally they're just too darn smart for their own good.)
>
> I can live without actual snom-style orbits at this time (handy though
> they might be), since the current system involves parking the call on an
> external line and walking over to another office to say that they have a
> call.  I imagine that down the road it'll usually just be an attended
> transfer, but we do park calls around phones a fair bit as we brainstorm
> issues.
>
> (Actually, I can't get attended transfers working, either.  All
> transfers are blind.  Related?)

Press hold (R on the old phones) or the line key (the buttons on the right) of 
the line which you are currently using. The first call is on hold. Now call 
the transfer target. Talk to him. Now press the transfer button or Xfer soft 
button (below the screen). Depending on the configuration of your phone the 
two call will be joined immediately ('Call join on Xfer' on the advanced), or 
you have to select the target for transfer manually and press OK one more 
time (<- good protection against joining your girlfriend with your wife ;) ).
This is how attended transfer works the "snom way".

Regards
  Nils
-- 
snom technology AGGradestr. 46D-12347 Berlin
Nils Ohlmeier
mailto:[EMAIL PROTECTED]  http://www.snom.com
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Re: [Asterisk-Users] SNOM 360 and parking

[Steve Davies]

On 7/12/05, Patrick Friedel <[EMAIL PROTECTED]> wrote:
> OK, last showstopper that I just can't puzzle my way through - parking
> calls with the snom phones.  I get the two phones connected, I hit
> transfer on one, the other phone goes to MOH and the first phone gives
> me DT, so I dial 700 and hit the OK button.  Call transferred, the SNOM
> hangs up before I have a chance to hear which extension it parked to.
> Is there a way to make the SNOM phones stay off hook until you
> explicitly hang up during a transfer?  (my only complaint about these
> phones - occasionally they're just too darn smart for their own good.)

The 'Transfer' button on snom phones is for blind transfer. What you
describe above is the blind transfer of the caller to extension 700.

> I can live without actual snom-style orbits at this time (handy though
> they might be), since the current system involves parking the call on an
> external line and walking over to another office to say that they have a
> call.  I imagine that down the road it'll usually just be an attended
> transfer, but we do park calls around phones a fair bit as we brainstorm
> issues.

I assume from the lack of any other detail that "700" is how you park
your calls?

> (Actually, I can't get attended transfers working, either.  All
> transfers are blind.  Related?)

For attended transfer on the snom, use the 'hold' button instead of
the 'transfer' button. IIRC the default behaviour is to transfer on
hangup as you describe. We do not use call parking at-all here - it
only confused the "normal" users.

Regards,
Steve
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Re: [Asterisk-Users] SNOM 360 and parking

[Julian J. M.]

To do attended transfers with Snom 360, you need to put the current
call on hold, dial the dest extension, tell him/her something, and
press the Transfer button.

I don't think it'll work with asterisk call parking, though...

Julian J. M.

On 7/12/05, Patrick Friedel <[EMAIL PROTECTED]> wrote:
> OK, last showstopper that I just can't puzzle my way through - parking
> calls with the snom phones.  I get the two phones connected, I hit
> transfer on one, the other phone goes to MOH and the first phone gives
> me DT, so I dial 700 and hit the OK button.  Call transferred, the SNOM
> hangs up before I have a chance to hear which extension it parked to.
> Is there a way to make the SNOM phones stay off hook until you
> explicitly hang up during a transfer?  (my only complaint about these
> phones - occasionally they're just too darn smart for their own good.)
> 
> I can live without actual snom-style orbits at this time (handy though
> they might be), since the current system involves parking the call on an
> external line and walking over to another office to say that they have a
> call.  I imagine that down the road it'll usually just be an attended
> transfer, but we do park calls around phones a fair bit as we brainstorm
> issues.
> 
> (Actually, I can't get attended transfers working, either.  All
> transfers are blind.  Related?)
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Re: [Asterisk-Users] snom 360 audio garbled

[Scott]

Michael George wrote:
> I have a new snom 360 on an internal net to my * box.  When putting a call on
> hold and taking it off, the audio will usually be broken and not
> understandable.
> 
> Sometimes this happens on incoming calls and almost always on outgoing calls.
> 
> Anyone run into this before?
> 
> Thx!

You might try checking which codec is in use pre-hold and post-hold. On
our Snom 190s, g726 always seems to sound garbled, and the call may be
starting with one codec (like ulaw), then continuing with another (like
g726) after being taken off of hold.

Scott

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Re: [Asterisk-Users] snom 360 audio garbled

[Michael George]

On Mon, Jul 18, 2005 at 08:54:09PM -0400, Scott wrote:
> 
> You might try checking which codec is in use pre-hold and post-hold. On
> our Snom 190s, g726 always seems to sound garbled, and the call may be
> starting with one codec (like ulaw), then continuing with another (like
> g726) after being taken off of hold.

Do you mean which codec asterisk thinks it's using or which codec the phone is
using at a particular time?

I know that "sip show channels" will list the codecs used on a channel by *,
but how do I find out what codec the Snom is using on an ongoing call?

Also, if the snom went to a different codec, wouldn't the audio be completely
incomprehensible?  What we get is garbled and we can hear what sound like
syllables, but you cannot really make it out.

Thank you.

-- 
-M

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Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] snom 360 audio garbled

[Scott]

Michael George wrote:
> On Mon, Jul 18, 2005 at 08:54:09PM -0400, Scott wrote:
>>You might try checking which codec is in use pre-hold and post-hold. On
>>our Snom 190s, g726 always seems to sound garbled, and the call may be
>>starting with one codec (like ulaw), then continuing with another (like
>>g726) after being taken off of hold.
> 
> 
> Do you mean which codec asterisk thinks it's using or which codec the phone is
> using at a particular time?
> 
> I know that "sip show channels" will list the codecs used on a channel by *,
> but how do I find out what codec the Snom is using on an ongoing call?
> 
> Also, if the snom went to a different codec, wouldn't the audio be completely
> incomprehensible?  What we get is garbled and we can hear what sound like
> syllables, but you cannot really make it out.
> 
> Thank you.

Yes, I mean running "sip show channels" both pre- and post-hold and
seeing if the codec listed for the channel is different. If it is, try
disallowing the second codec in sip.conf.

Scott
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Re: [Asterisk-Users] snom 360 audio garbled

[Michael George]

On Tue, Jul 19, 2005 at 02:17:07PM -0400, Scott wrote:
> > Do you mean which codec asterisk thinks it's using or which codec the phone 
> > is
> > using at a particular time?
> > 
> > I know that "sip show channels" will list the codecs used on a channel by *,
> > but how do I find out what codec the Snom is using on an ongoing call?
> > 
> > Also, if the snom went to a different codec, wouldn't the audio be 
> > completely
> > incomprehensible?  What we get is garbled and we can hear what sound like
> > syllables, but you cannot really make it out.
> > 
> > Thank you.
> 
> Yes, I mean running "sip show channels" both pre- and post-hold and
> seeing if the codec listed for the channel is different. If it is, try
> disallowing the second codec in sip.conf.

Okay, I'll check that.  Is there a way to disable all but the first codec from
the GUI?  I've not yet gotten to the mass-configuration yet...

Thanks!

-- 
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Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] snom 360 audio garbled

[Michael George]

On Tue, Jul 19, 2005 at 02:17:07PM -0400, Scott wrote:
> Michael George wrote:
> > On Mon, Jul 18, 2005 at 08:54:09PM -0400, Scott wrote:
> >>You might try checking which codec is in use pre-hold and post-hold. On
> >>our Snom 190s, g726 always seems to sound garbled, and the call may be
> >>starting with one codec (like ulaw), then continuing with another (like
> >>g726) after being taken off of hold.
> > 
> > 
> > Do you mean which codec asterisk thinks it's using or which codec the phone 
> > is
> > using at a particular time?
> > 
> > I know that "sip show channels" will list the codecs used on a channel by *,
> > but how do I find out what codec the Snom is using on an ongoing call?
> > 
> > Also, if the snom went to a different codec, wouldn't the audio be 
> > completely
> > incomprehensible?  What we get is garbled and we can hear what sound like
> > syllables, but you cannot really make it out.
> > 
> > Thank you.
> 
> Yes, I mean running "sip show channels" both pre- and post-hold and
> seeing if the codec listed for the channel is different. If it is, try
> disallowing the second codec in sip.conf.

According to the distributor, the lastest firmware revisions addressed this
problem.  I have now upgraded the firmware and I hope it helps.  I'll see
later this week.

Thank you.

-- 
-M

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Re: [Asterisk-Users] Snom 360 record button?

[Kevin P. Fleming]

Patrick Friedel wrote:

 I assume the first one is the snom requesting the PBX to start 
recording, and the second is asterisk reminding the snom that it doesn't 
allow the INFO method and to get bent?


Yep, you're right. That would be an interesting addition to Asterisk, 
but it will likely require a bounty to get someone to write support for it.

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Re: [Asterisk-Users] Snom 360 record button?

[Olle E. Johansson]

Patrick Friedel wrote:
> Sorry if this is an obvious question, but I haven't seen an obvious
> answer on the wiki that I remember.  Has anyone managed to make the
> record button on the snom 360 fire off the Monitor() application?  I
> don't see a bounty, and googling for "snom 360 record button asterisk"
> returns tons of product specification pages. (Joy!) I don't see a bounty
> for it, and the only mention I _see_ on the wiki is "one touch RECORD
> button usuable only with special PBX support via SIP INFO method" which
> isn't much of an answer.
> 
Is this during a call? Can you please send me a full SIP DEBUG of the call?

Brainstorming, maybe we could treat this as a transfer to a local
extension somehow and turn monitor on in the dial plan that way...

/O
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Re: [Asterisk-Users] Snom 360 record button?

[Patrick Friedel]

Olle E. Johansson wrote:


Patrick Friedel wrote:
 


Sorry if this is an obvious question, but I haven't seen an obvious
answer on the wiki that I remember.  Has anyone managed to make the
record button on the snom 360 fire off the Monitor() application?  I
don't see a bounty, and googling for "snom 360 record button asterisk"
returns tons of product specification pages. (Joy!) I don't see a bounty
for it, and the only mention I _see_ on the wiki is "one touch RECORD
button usuable only with special PBX support via SIP INFO method" which
isn't much of an answer.

   


Is this during a call? Can you please send me a full SIP DEBUG of the call?

Brainstorming, maybe we could treat this as a transfer to a local
extension somehow and turn monitor on in the dial plan that way...
 

 Yeah, that was in the middle of a call - the only other SIP debug 
information is the normal call build up and tear down.  I can generate 
it if you want, but it's nothing exciting, just the usual handshaking.  
But that was kind of what I was thinking would be a solution - Asterisk 
sees the Record INFO packet, and conferences the call to a local 
extension with Monitor() going.  I'm not 100% sure whether or not the 
Snom 360 expects anything _else_ (other than a simple acknowledgement) 
back from the PBX, as there doesn't appear to be a whitepaper for it it 
on Snom's site.  I would assume it would be fairly straightforward in 
chan_sip.c to bang in a new method, but whether there are any 
ramifications, well...

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Re: [Asterisk-Users] Snom 360 record button?

[Maik Schmitt]

> Is this during a call? Can you please send me a full SIP DEBUG of the call?
>
> Brainstorming, maybe we could treat this as a transfer to a local
> extension somehow and turn monitor on in the dial plan that way...

Hmm IMO the automon-feature would be better for this. It does exactly what we 
want (start recording during a call) and is configurable via Dial-Options. 
The only thing I don't know is how to activate it without sending the DTMF 
sequence.

Maik Schmitt
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RE: [Asterisk-Users] Snom 360 record button?

[Christian Stredicke]

It would be nice if the PBX can acknowlegdge the Record header - then it
would have the chance to paint a record icon on the screen.

In the next release.-)

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Patrick Friedel
> Sent: Thursday, July 28, 2005 7:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Snom 360 record button?
> 
> Olle E. Johansson wrote:
> 
> >Patrick Friedel wrote:
> >  
> >
> >>Sorry if this is an obvious question, but I haven't seen an obvious 
> >>answer on the wiki that I remember.  Has anyone managed to make the 
> >>record button on the snom 360 fire off the Monitor() 
> application?  I 
> >>don't see a bounty, and googling for "snom 360 record 
> button asterisk"
> >>returns tons of product specification pages. (Joy!) I don't see a 
> >>bounty for it, and the only mention I _see_ on the wiki is 
> "one touch 
> >>RECORD button usuable only with special PBX support via SIP INFO 
> >>method" which isn't much of an answer.
> >>
> >>
> >>
> >Is this during a call? Can you please send me a full SIP 
> DEBUG of the call?
> >
> >Brainstorming, maybe we could treat this as a transfer to a local 
> >extension somehow and turn monitor on in the dial plan that way...
> >  
> >
>   Yeah, that was in the middle of a call - the only other SIP 
> debug information is the normal call build up and tear down.  
> I can generate it if you want, but it's nothing exciting, 
> just the usual handshaking.  
> But that was kind of what I was thinking would be a solution 
> - Asterisk sees the Record INFO packet, and conferences the 
> call to a local extension with Monitor() going.  I'm not 100% 
> sure whether or not the Snom 360 expects anything _else_ 
> (other than a simple acknowledgement) back from the PBX, as 
> there doesn't appear to be a whitepaper for it it on Snom's 
> site.  I would assume it would be fairly straightforward in 
> chan_sip.c to bang in a new method, but whether there are any 
> ramifications, well...
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Re: [Asterisk-Users] Snom 360 record button?

[Patrick Friedel]

Maik Schmitt wrote:


Is this during a call? Can you please send me a full SIP DEBUG of the call?

Brainstorming, maybe we could treat this as a transfer to a local
extension somehow and turn monitor on in the dial plan that way...
   



Hmm IMO the automon-feature would be better for this. It does exactly what we 
want (start recording during a call) and is configurable via Dial-Options. 
The only thing I don't know is how to activate it without sending the DTMF 
sequence.
 

 Hmm, you're right, I wasn't aware of the automon feature - I don't 
know if it was in the original SIP trace, but the 360 is sending _some_ 
DTMF signalling, but I don't know what it's actually sending.  I'm 
currently at 9 for verbosity, let me amp that up and see if it will ever 
actually display the tones it's receiving.



voip*CLI> sip debug peer pjf
SIP Debugging Enabled for IP: 10.0.1.213:2051
voip*CLI> set verbose 255
Verbosity was 9 and is now 255
voip*CLI>
[boring build up that isn't new to anyone]

Sip read:
INFO sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-aytv2p1rs5ih;rport
From: "Patrick" ;tag=4rbnk4yyxd
To: ;tag=as3b19835c
Call-ID: [EMAIL PROTECTED]
CSeq: 3 INFO
Max-Forwards: 70
Contact: 
User-Agent: snom360/3.60r
Record: on
Content-Length: 0


11 headers, 0 lines
Receiving DTMF!
Jul 28 14:12:15 WARNING[26025]: chan_sip.c:6166 receive_info: Unable to 
parse INFO message from [EMAIL PROTECTED] Content

Transmitting (no NAT):
SIP/2.0 415 Unsupported media type
Via: SIP/2.0/UDP 10.0.1.213:2051;branch=z9hG4bK-aytv2p1rs5ih
From: "Patrick" ;tag=4rbnk4yyxd
To: ;tag=as3b19835c
Call-ID: [EMAIL PROTECTED]
CSeq: 3 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0

[teardown]

Hmm.  It's getting _something_, but the verbosity won't reveal it.  
Checking chan_sip.c, I don't see a mechanism for revealing the data, 
it'd be around line 7716.  It we could reveal it (or Nils just tells us. 
:), it might be as simple as editing features.conf and setting automon 
to whatever DTMF the Snom sends when you hit the record button.

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Re: [Asterisk-Users] Snom 360 record button?

[Nils Ohlmeier]

On Thursday 28 July 2005 17:12, Patrick Friedel wrote:
>   Yeah, that was in the middle of a call - the only other SIP debug
> information is the normal call build up and tear down.  I can generate
> it if you want, but it's nothing exciting, just the usual handshaking.
> But that was kind of what I was thinking would be a solution - Asterisk
> sees the Record INFO packet, and conferences the call to a local
> extension with Monitor() going.  I'm not 100% sure whether or not the
> Snom 360 expects anything _else_ (other than a simple acknowledgement)
> back from the PBX, as there doesn't appear to be a whitepaper for it it
> on Snom's site.  I would assume it would be fairly straightforward in

Sorry for the missing whitepaper ;-)
The snom phone is not expecting anything else then any reply for the INFO 
request.

  Nils

> chan_sip.c to bang in a new method, but whether there are any
> ramifications, well...

-- 
snom technology AGGradestr. 46D-12347 Berlin
Nils Ohlmeier
mailto:[EMAIL PROTECTED]  http://www.snom.com
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Re: [Asterisk-Users] Snom 360 record button?

[Patrick Friedel]

Nils Ohlmeier wrote:


On Thursday 28 July 2005 17:12, Patrick Friedel wrote:
 


 Yeah, that was in the middle of a call - the only other SIP debug
information is the normal call build up and tear down.  I can generate
it if you want, but it's nothing exciting, just the usual handshaking.
But that was kind of what I was thinking would be a solution - Asterisk
sees the Record INFO packet, and conferences the call to a local
extension with Monitor() going.  I'm not 100% sure whether or not the
Snom 360 expects anything _else_ (other than a simple acknowledgement)
back from the PBX, as there doesn't appear to be a whitepaper for it it
on Snom's site.  I would assume it would be fairly straightforward in
   



Sorry for the missing whitepaper ;-)
The snom phone is not expecting anything else then any reply for the INFO 
request.
 

 Heh, figured that would draw you out of the woodwork.  It's good to 
know that information.

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Re: [Asterisk-Users] Snom 360 record button?

[Olle E. Johansson]

Christian Stredicke wrote:
> It would be nice if the PBX can acknowlegdge the Record header - then it
> would have the chance to paint a record icon on the screen.
> 
> In the next release.-)
> 
Right.

Is there another header for turning off recording?

Anyway, we should not send "unsupported media type"...

/O ;-)
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[Asterisk-Users] Snom 360 4.0 firmware issue

[Colin E. McDonald]

The new update seems to have cured my issue with calls intersecting and
Zap lines not being hung up after the user terminates the session but
now I am having sound issues with all of my phones. The sounds seems to
be very low on all of them and there is a definite change from the same
set when it was at 3.6j. The speaker also generates what appears to be
static but you can discern a scratchy sounding echo. This is also
occuring on all phones after the upgrade. I have genereated a support
ticket to Snom but I wanted to see if anyone on the list has run into
the same behavior. 


Thanks

Colin
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[Asterisk-Users] Snom 360 and 320 AutoAnswer

[Alvaro Parres]

Hi list..
 
   I want to make Snom 360 and 30 to autoanswer so i can have a paging sistem.
 
I tried tu send intercom=true with the little patch to chan_sip.c and it didm't work
 
any one have and idea of ow to do this.
 
 
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RE: [Asterisk-Users] SNOM 360 locked up

[Christian Stredicke]

Try loading
http://phone-ip-address/line_sip.htm?settings=save&user_dp_str1= (if
that was in the line 1) while the phone boots up (keep your finger on
the reload button). If that does not work, you need to do a tftp update.

Also consider moving to version 4.5
(http://www.snom.com/snom360_release_notes.html).

Christian

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Steven Ringwald
> Sent: Friday, December 23, 2005 5:16 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] SNOM 360 locked up
> 
> Hello all!
> 
> I was trying to get the dial-string setup for my regular 
> usage, and the phone locked up in the middle of dialing. 
> Basically, I put the following line in, hit save, and got as 
> far as dialing '9', and the phone froze.
> 
> |^(9[0-9]{10}|sip:[EMAIL PROTECTED]|d
> 
> Now the phone boots up to the SNOM splash screen and hangs 
> there. I can ping it, but cannot get to the web-interface and 
> cannot reset to factory defaults using the web-gui.
> 
> Any idea how I can reset the phone to factory w/o using the 
> GUI? Or am I completely hosed?
> 
> Steve
> 
> 
> 
> 
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RE: [Asterisk-Users] SNOM 360 locked up

[Steven Ringwald]

On Thu, 2005-12-22 at 23:34 +0100, Christian Stredicke wrote:


Try loading
http://phone-ip-address/line_sip.htm?settings=save&user_dp_str1= (if
that was in the line 1) while the phone boots up (keep your finger on
the reload button). If that does not work, you need to do a tftp update.



Yeah. The website address didn't work. (The phone, I think, is not far enough along to even start the webserver). I will try the tftp update method, and see what happens.

So far, though, it doesn't seem to be hitting the tftp server that I set up manually.




Also consider moving to version 4.5
(http://www.snom.com/snom360_release_notes.html).



Any idea how to do that? I think it is running 4.1. I have put the firmware image URL into the upgrade line before, and it didn't take. (Ended up going back to what it had previously had).

Thanks for the help!
Steve




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Re: [Asterisk-Users] SNOM 360 locked up

[Sven Fischer (support)]

On Friday 23 December 2005 00:39, Steven Ringwald wrote:
> On Thu, 2005-12-22 at 23:34 +0100, Christian Stredicke wrote:
> > Try loading
> > http://phone-ip-address/line_sip.htm?settings=save&user_dp_str1= (if
> > that was in the line 1) while the phone boots up (keep your finger on
> > the reload button). If that does not work, you need to do a tftp update.
>
> Yeah. The website address didn't work. (The phone, I think, is not far
> enough along to even start the webserver). I will try the tftp update
> method, and see what happens.
>
> So far, though, it doesn't seem to be hitting the tftp server that I set
> up manually.

A step by step description can be found here:

http://www.snom.com/wiki/index.php/Main_Page#Firmware_Update

>
> > Also consider moving to version 4.5
> > (http://www.snom.com/snom360_release_notes.html).
>
> Any idea how to do that? I think it is running 4.1. I have put the
> firmware image URL into the upgrade line before, and it didn't take.
> (Ended up going back to what it had previously had).
>
> Thanks for the help!
> Steve

Regards,

Sven

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Re: [Asterisk-Users] Snom 360 and hints

[Paul Hales]

To add to this - when I do a 'show hints' the phone shows state 0, which
I assume means not in use.

PaulH

On Thu, 2005-09-01 at 12:38 +1000, Paul Hales wrote:
> I am setting up a snom 360, and the lights come on OK when the mapped
> user makes an outgoing call, but when the user takes an incoming call
> the light does not come on.
> 
> I do not want to install the bristuff patch if possible.
> (although I can see that with the devstate command I can make the lights
> do whatever I want)
> 
> Any ideas?
> 
> regards,
> 
> Paul Hales
> Melbourne, Australia
> 
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Re: [Asterisk-Users] Snom 360 and hints

[Alessio Focardi]

Hello Paul,

Thursday, September 1, 2005, 4:38:42 AM, you wrote:

PH> I am setting up a snom 360, and the lights come on OK when the mapped
PH> user makes an outgoing call, but when the user takes an incoming call
PH> the light does not come on.

PH> I do not want to install the bristuff patch if possible.
PH> (although I can see that with the devstate command I can make the lights
PH> do whatever I want)

Same here, it think it depends on hint status: when you make a call
calling hint is set to 1, but called one stays 0.

Correct behaviour should be

put the hint of the caller to 1 (steady ligt) while calling

put the hint of the called to X (blinking light, cant remember which
state it is ) while phone is ringing, then to 1 if call is answered.

Unfortunately I dont know how to accomplish this 

Regards!


-- 
Best regards,
 Alessiomailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] Snom 360 and hints

[BJ Weschke]

 Issue #3644 has recently been committed to CVS-HEAD which allows for full device state notification via subscriptions for Snom 360 and other supporting phones w/o the need for additional patches.
On 9/1/05, Alessio Focardi <[EMAIL PROTECTED]> wrote:
Hello Paul,Thursday, September 1, 2005, 4:38:42 AM, you wrote:PH> I am setting up a snom 360, and the lights come on OK when the mapped
PH> user makes an outgoing call, but when the user takes an incoming callPH> the light does not come on.PH> I do not want to install the bristuff patch if possible.PH> (although I can see that with the devstate command I can make the lights
PH> do whatever I want)Same here, it think it depends on hint status: when you make a callcalling hint is set to 1, but called one stays 0.Correct behaviour should beput the hint of the caller to 1 (steady ligt) while calling
put the hint of the called to X (blinking light, cant remember whichstate it is ) while phone is ringing, then to 1 if call is answered.Unfortunately I dont know how to accomplish this Regards!
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Re: [Asterisk-Users] Snom 360 and hints

[Jeff Brownlee]

PH> I am setting up a snom 360, and the lights come on OK when the mapped
PH> user makes an outgoing call, but when the user takes an incoming call
PH> the light does not come on.

PH> I do not want to install the bristuff patch if possible.
PH> (although I can see that with the devstate command I can make the lights
PH> do whatever I want)

First, ensure that the 360 has "Filter Packets from Registrar" turned off (under 
Advanced).  Next, make sure you have hint priorities setup for each of the extensions you are 
trying to monitor.  With both of these in place, you should see an entry for each extension you are 
monitoring when you do "sip show subscriptions" from the * CLI.  If not, rinse and repeat 
the above steps.  Also, you may want to manually recycle power on the 360 if you happen to reset 
asterisk for any reason (reload extensions, etc), as it will lose all the subscriptions and have to 
wait until the phones resend the subscription.

-Jeff

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