Re: [asterisk-users] SIP phone supporting more than 10 extension with a call transfer command

2007-03-16 Thread Sven Fischer (support)
snom320, snom360 and snom370 are supporting 12 different SIP identities.

Regards,
Sven

On Friday 16 March 2007 10:57, younss azzayani wrote:
 Hi every body,
 can someone please tell me about a SIP phone that support more than 10
 extension (free or not free ;) ) wich will be used in my company, i've
 bought a SNOM but it just support 5 sip extension
 Kind regards
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Handelsregistereintrag/Register of Corporations entry: 
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Re: [Asterisk-Users] OT: Snom 320, displaying text on the screen from *

2006-03-10 Thread Sven Fischer (support)
BTW regarding AOC you may want to have a look into this:

http://www.snom.com/wiki/index.php/Advice_of_charge_%28AOC%29_in_SIP

On Friday 10 March 2006 04:37, Dofear wrote:
 Can this feature be used to display total balance left (for a phone) on the
 display of the phone?

   _

 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Colin
 Anderson Sent: Friday, March 10, 2006 2:33 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] OT: Snom 320, displaying text on the screen
 from *


 try sipsak -M -O desktop -B foo -s sip:user@registrar -H ip of
 registrar

 the trick is to specify the -O desktop parameter + the -H ip of
 registrar parameter. Sipsak fakes the host-header of the registrar so
 that the Snom thinks it is coming from your Asterisk server, then lets the
 message through to the desktop (the display of the phone)

 I wasn't kidding about obscure syntax, sipsak is a PITA

 -Original Message-
 From: Sean Kennedy [mailto:[EMAIL PROTECTED]
 Sent: Thursday, March 09, 2006 5:01 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] OT: Snom 320, displaying text on the screen
 from *


 I have that set, but for some reason I get errors when I try sipsak, and
 nothing comes through to the phone:



 sipsak -M -B test -s sip:[EMAIL PROTECTED]
 timeout after 500ms
 timeout after 500ms...


 Some debugging info:


 [EMAIL PROTECTED] root]# sipsak -vvv -M -B test -s sip:[EMAIL PROTECTED]
 warning: ignoring -i option when in usrloc mode
 fqdnhostname: 192.168.1.1
 our Via-Line: Via: SIP/2.0/UDP
 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;alias

 New message with Via-Line:
 MESSAGE sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;alias
 To: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 MESSAGE
 Content-Type: text/plain
 Max-Forwards: 70
 User-Agent: sipsak 0.9.5
 From: sip:[EMAIL PROTECTED]:34213;tag=7c8bd47f
 Content-Length: 4

 test
 sending message ...

 request:
 MESSAGE sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP 192.168.1.1:34213;branch=z9hG4bK.105fb86e;rport;alias
 To: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 1 MESSAGE
 Content-Type: text/plain
 Max-Forwards: 70
 User-Agent: sipsak 0.9.5
 From: sip:[EMAIL PROTECTED]:34213;tag=7c8bd47f
 Content-Length: 4

 test
 send to: UDP:192.168.1.67:5060

 ignoring MESSAGE retransmission
 timeout after 500 ms


 So I am at a bit of a loss.

 Thanks for your help though, I apprecaite it.  :)

 Colin Anderson wrote:


 Trick with Sipsak is you have to change the network port to 5060 or sipsak
 messages never hit the right port. In the web interface, Advaced  Avanced
 Network  Network identity (port): change that to 5060 and you should be
 good assuming you can figure out sipsak's nasty syntax. hth.

-- 
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Whitepapers at:  http://www.snom.com/white_papers.html
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Re: [Asterisk-Users] Announcement: Snom 360 with integrated XML O bjects

2006-01-30 Thread Sven Fischer (support)
The XML minibrowser is available in a special image only. 

http://www.snom.com/minibrowser/firmware/

In future images it will be in by default.

On Friday 27 January 2006 16:15, Colin Anderson wrote:
 Aha, I see it's 4.1, cool. So I just have to do a straight upgrade to 5.0
 and I have this new toy to play with, correct?

 -Original Message-
 From: Sven Fischer (support) [mailto:[EMAIL PROTECTED]
 Sent: Friday, January 27, 2006 5:27 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Announcement: Snom 360 with integrated XML
 O bjects


 Hi.

 Use massdeployment for putting the licenses on to your phones.

 There is a setting called license_url you can use like the firmware
 update

 URL, the macro {mac} will be replaced by the MAC address of the phone. So
 if

 you provide the setting like this:

 license_url: http://yourwebserver/{mac}.txt

 all phones will download and install their licenses from this directory
 automatically.

 Ok, you need to send us a list of all MAC addresses and we will send you
 the

 needed licenses.

 BTW which firmware is already on the phones ? If it is 4.something and they
 are working, the license is already on the phones. With firmware 4.0 you
 need
 to have a license on the phone.

 Best regards,

 Sven

 On Friday 27 January 2006 06:01, Colin Anderson wrote:
  Is there any plans for a site license or some way to deploy the license a
  little more elegantly? I have a lot of 360's!
 
  I'm excited about this feature - it enables me to deploy some solutions
  that I have been promising to my endusers. The two I have in mind are
  Outlook calendar push to the display, and Outlook contact pull to the
  directory. Some other ones will involve caller ID lookup to our CRM. If I
  make progress along these lines, I will post results to the list.
 
  -Original Message-
  From: Christian Stredicke [mailto:[EMAIL PROTECTED]
  Sent: Thursday, January 26, 2006 9:14 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: RE: [Asterisk-Users] Announcement: Snom 360 with integrated XML
  Objects
 
 
  As far as the licence is concerned that is something that we introduced
  in the 4.0 image and this is not against our customers (which would be
  stupid). It shall protect us from clones.
 
  The jump to the 5.0 is not about this licensing stuff, we just changed
  the ramdisk and freed up more memory. I know this is not very pleasant,
  and we cross fingers that this is the last time we have to do something
  like this. But running out of memory is also not very pleasant!
  Especially when new cool features ask for more memory!
 
  CS
 
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] On Behalf Of
   Colin Anderson
   Sent: Thursday, January 26, 2006 2:16 PM
   To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   Subject: RE: [Asterisk-Users] Announcement: Snom 360 with
   integrated XML Objects
  
   that is *very* cool. However, I am somewhat concerned about
   being forced to license the firmware (even if it is free) can
   you comment for the list the rationale behind forcing a
   license and how this might affect Snom users who, say, want
   to DOWN grade their firmware?
  
   ps is there a timetable for supported, formally released
   firmware version?
  
   -Original Message-
   From: Hirosh Dabui [mailto:[EMAIL PROTECTED]
   Sent: Thursday, January 26, 2006 11:02 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: [Asterisk-Users] Announcement: Snom 360 with
   integrated XML Objects
  
  
   -BEGIN PGP SIGNED MESSAGE-
   Hash: RIPEMD160
  
   Dear user,
  
   the new snom 360 is able to use services from standard web servers.
   Users can deploy customized client services with snom 360 and
   interact with other users via the keypad. The snom 360 will
   use HTTP protocol from standard web servers, like Apache.
   Typical services are:
  
   ~   1. To-do lists
   ~   2. Stock Information
   ~   3. Weather
   ~   4. Provisioning
   ~   5. Agenda
   ~   6. Telephone directory
  
  
   For further information go to
   http://snom.com/wiki/index.php/Xmlobjects
  
   Note: *That is a pre-release, probably the software is still unstable*
  
   Best regards,
  
   Hirosh Dabui
  
   - --
   snom technology AG
   Dipl.-Ing. Hirosh Dabui
  
   PGP Key-ID: 0x30A34758
   mailto:[EMAIL PROTECTED]
  
   http://snom.com
  
  
   -BEGIN PGP SIGNATURE-
   Version: GnuPG v1.4.2 (GNU/Linux)
  
   iD8DBQFD2Q6YAO47/DCjR1gRA6REAJ4iSyot8OhFVDt0/C2I7KFoRCP18ACeNGau
   FCXMUdN9loiwy948EO8th9U=
   =Qntp
   -END PGP SIGNATURE-
  
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Re: [Asterisk-Users] Announcement: Snom 360 with integrated XML O bjects

2006-01-27 Thread Sven Fischer (support)
Hi.

Use massdeployment for putting the licenses on to your phones.

There is a setting called license_url you can use like the firmware update 
URL, the macro {mac} will be replaced by the MAC address of the phone. So if 
you provide the setting like this:

license_url: http://yourwebserver/{mac}.txt

all phones will download and install their licenses from this directory 
automatically.

Ok, you need to send us a list of all MAC addresses and we will send you the 
needed licenses.

BTW which firmware is already on the phones ? If it is 4.something and they 
are working, the license is already on the phones. With firmware 4.0 you need 
to have a license on the phone.

Best regards,

Sven

On Friday 27 January 2006 06:01, Colin Anderson wrote:
 Is there any plans for a site license or some way to deploy the license a
 little more elegantly? I have a lot of 360's!

 I'm excited about this feature - it enables me to deploy some solutions
 that I have been promising to my endusers. The two I have in mind are
 Outlook calendar push to the display, and Outlook contact pull to the
 directory. Some other ones will involve caller ID lookup to our CRM. If I
 make progress along these lines, I will post results to the list.

 -Original Message-
 From: Christian Stredicke [mailto:[EMAIL PROTECTED]
 Sent: Thursday, January 26, 2006 9:14 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Announcement: Snom 360 with integrated XML
 Objects


 As far as the licence is concerned that is something that we introduced
 in the 4.0 image and this is not against our customers (which would be
 stupid). It shall protect us from clones.

 The jump to the 5.0 is not about this licensing stuff, we just changed
 the ramdisk and freed up more memory. I know this is not very pleasant,
 and we cross fingers that this is the last time we have to do something
 like this. But running out of memory is also not very pleasant!
 Especially when new cool features ask for more memory!

 CS

  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  Colin Anderson
  Sent: Thursday, January 26, 2006 2:16 PM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] Announcement: Snom 360 with
  integrated XML Objects
 
  that is *very* cool. However, I am somewhat concerned about
  being forced to license the firmware (even if it is free) can
  you comment for the list the rationale behind forcing a
  license and how this might affect Snom users who, say, want
  to DOWN grade their firmware?
 
  ps is there a timetable for supported, formally released
  firmware version?
 
  -Original Message-
  From: Hirosh Dabui [mailto:[EMAIL PROTECTED]
  Sent: Thursday, January 26, 2006 11:02 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [Asterisk-Users] Announcement: Snom 360 with
  integrated XML Objects
 
 
  -BEGIN PGP SIGNED MESSAGE-
  Hash: RIPEMD160
 
  Dear user,
 
  the new snom 360 is able to use services from standard web servers.
  Users can deploy customized client services with snom 360 and
  interact with other users via the keypad. The snom 360 will
  use HTTP protocol from standard web servers, like Apache.
  Typical services are:
 
  ~   1. To-do lists
  ~   2. Stock Information
  ~   3. Weather
  ~   4. Provisioning
  ~   5. Agenda
  ~   6. Telephone directory
 
 
  For further information go to
  http://snom.com/wiki/index.php/Xmlobjects
 
  Note: *That is a pre-release, probably the software is still unstable*
 
  Best regards,
 
  Hirosh Dabui
 
  - --
  snom technology AG
  Dipl.-Ing. Hirosh Dabui
 
  PGP Key-ID: 0x30A34758
  mailto:[EMAIL PROTECTED]
 
  http://snom.com
 
 
  -BEGIN PGP SIGNATURE-
  Version: GnuPG v1.4.2 (GNU/Linux)
 
  iD8DBQFD2Q6YAO47/DCjR1gRA6REAJ4iSyot8OhFVDt0/C2I7KFoRCP18ACeNGau
  FCXMUdN9loiwy948EO8th9U=
  =Qntp
  -END PGP SIGNATURE-
 
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Re: [Asterisk-Users] OT:Snom 360 prompt for registration pwd?

2006-01-23 Thread Sven Fischer (support)
I recommend to use the mass deployment feature to maintain your phones.

http://www.snom.com/wiki/index.php/Massdeployment_Firmware_Release_5

Besides this setting you cannot expect each setting is set by default to 
exactly match your needs. Different environments different setup.

Best regards,

Sven

On Saturday 21 January 2006 19:57, Colin Anderson wrote:
 I can confirm that this is the issue. I now have to toggle it off manually
 on 120 phones. I can tell you, in the real world, you don't hand out
 passwords to users for their phones, they will not understand why you need
 a password for a phone. You may want to consider changing the default
 settings.

 Thanks for the info. Quick and accurate responses like yours are why I am a
 Snom fan.

 -Original Message-
 From: Christian Stredicke [mailto:[EMAIL PROTECTED]
 Sent: Saturday, January 21, 2006 8:51 AM
 To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
 Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] OT:Snom 360 prompt for registration pwd?


 The idea was that passwords will not be provisoned automatically, you
 must enter them manually on the phone. Which makes sense in scenarios
 where you completely automatically provision phones and hand out the
 password to the users.

 But maybe you are right, we should turn this off by default. I also had
 some pain with it!

 CS

  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of
  The VoIP Connection
  Sent: Saturday, January 21, 2006 10:46 AM
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
  Subject: RE: [Asterisk-Users] OT:Snom 360 prompt for registration pwd?
 
  Christian,
 
  Why is this this setting on by default?  I don't understand
  why anyone would want this behavior. -Mike
 
  Michael Crown
  Managing Partner
  www.thevoipconnection.com
  321.989.6728 ext. 611
  sip:[EMAIL PROTECTED]
 
   -Original Message-
   From: Christian Stredicke [mailto:[EMAIL PROTECTED]
   Sent: Friday, January 20, 2006 8:05 PM
   To: Colin Anderson
   Cc: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: RE: [Asterisk-Users] OT:Snom 360 prompt for
 
  registration pwd?
 
   Did you try to turn Challenge Response on Phone off in
 
  the advanced
 
   settings on the web interface?
  
   CS
  
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
 
  Behalf Of Colin
 
Anderson
Sent: Friday, January 20, 2006 8:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] OT:Snom 360 prompt for registration pwd?
   
I have a whack of Snom 360's. Occasionally, *some* of them,
  
   prompt the
  
user, on the screen, for the registration password. You enter it,
everything's OK.
Next day, same thing. This is like on 5 or 6 phones out
 
  of a lot of
 
120.
   
It's *always* the same phones. I haven't drilled down to
  
   things like
  
firmware rev yet, since I ordered them all as one lot, but I'm
wondering if anyone knows under which circumstances a 360 would
forget it's reg password?
   
tia
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See our FAQs at: http://www.snom.com/faq0.html?L=1
Whitepapers at:  http://www.snom.com/white_papers.html
---
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Sven Fischer fax +49 30 39833111
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Re: [Asterisk-Users] controlling SIP subscriptions from SNOM phones

2006-01-09 Thread Sven Fischer (support)
On Saturday 07 January 2006 02:30, Philipp von Klitzing wrote:
 Hi!

  Now, one user, not the receptionist, has gone in and set his personal
  numbers to these function keys thinking that DESTINATION meant setting a
  number to dial out. So now I have a ton of SIP SUBSCRIBE messages for his
  numbers.

 Indeed this situation is not ideal. The first thing to do in my opinion
 is ask SNOM to provide a new type of DESTINATION option that does not
 issue subscribes.

This is already available with firmware release 5 for snom320/360. This new 
type is named speed dial.


 Secondly you need to be aware that if Asterisk doesn't find a matching
 hint in the subscribecontext it will look check in the default context!
 This is, btw, one good reason to not have your local phones in the
 default context unless you want everyone out there to be able to
 subscribe to everyone else...

 Finally: Have you tried to create a new context, set the user's
 subscribecontext to this and do a _.,hint,SIP/DoesNotExist or smth
 similar within that context (and nothing else)?

 Cheers, Philipp


Regards,

Sven


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Whitepapers at:  http://www.snom.com/white_papers.html
---
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Sven Fischer fax +49 30 39833111
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Re: [Asterisk-Users] GXP-2000 any good with * ?

2006-01-02 Thread Sven Fischer (support)
On Saturday 31 December 2005 01:57, Ross C wrote:
 ... and 2 Snom 320's (now discontinued I think).  

No, they are not discontinued !!! 

Regards,

Sven
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Re: [Asterisk-Users] GXP-2000 any good with * ?

2006-01-02 Thread Sven Fischer (support)
This doesn't seem to be correct, too...

Sven

On Monday 02 January 2006 17:43, Ross C wrote:
 Sorry!!
 Just discontinued @ voipsupply.com I guess.
 Thx for the correction.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Sven Fischer
 (support)
 Sent: Monday, January 02, 2006 2:48 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?

 On Saturday 31 December 2005 01:57, Ross C wrote:
  ... and 2 Snom 320's (now discontinued I think).

 No, they are not discontinued !!!

 Regards,

 Sven
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---
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Whitepapers at:  http://www.snom.com/white_papers.html
---
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Sven Fischer fax +49 30 39833111
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Re: [Asterisk-Users] SNOM 360 locked up

2005-12-29 Thread 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=saveuser_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] displaying a message on the Snom 320 using sipsak

2005-10-13 Thread Sven Fischer (support)
Hi,

for the snom360 it is working the same way. Use firmware version 4.3 and be 
aware that the message is send to a specific SIP line and the phone is 
displaying it only if this SIP line is the current active line (outgoing 
identity) symbolised by a black phone (snom360) or the text in brackets 
(snom320).

Regards,

Sven Fischer

On Wednesday 12 October 2005 20:20, Franklin Webb wrote:
 Greetings fellow list members,
 It seems like a lot of people have been having trouble getting
 indicators working on the Snom phones, myself included.  Recently I was
 able to get the desktop functionality of sipsak to work on my Snom320,
 and I thought I would share what I could with the list.  For those not
 familiar this will replace the standard display when you are not on a
 call (normally showing the registered extension) with a text message of
 your choosing.  Our intent is to update this when our agents log into,
 and out of, queues.  This will give a visual indicator for agents and
 supervisors in our call center as to whether or not the phone is logged
 in, which is a large concern for us, and probably any call center.

 For the record I tried this with a Snom360 also and could not get it
 working.

 1.  Setup the phone in Asterisk as normal
 2.  Get and install sipsak.  It can be found at http://sipsak.org/
 http://sipsak.org/  (can be on any machine on your network, we used a
 Fedora Core 3 machine for this).
 3.  In the Snom320 Configuration, under the SIP tab of your extensions
 line (Line 1 for me) make sure Support Broken Registrar is set to on
 4.  In the Snom320 Configuration, under Advanced make sure Filter
 Packets from Registrar is set to off
 5.  In the Snom320 Configuration, under Advanced under Network
 identity (port): set it to 5060 (you might be able to use a different
 port in here and in the sipsak command, but this is what worked for me.
 6. Reboot the phone (just to be sure the settings take)

 Then use the following sipsak command:

 sipsak -vvv -M -O desktop -B Test Msg -r 5060 -s
 sip:[EMAIL PROTECTED]

 where:
 Test Msg is the message you want displayed.  To turn the message
 off just set it to empty string ().
 5060 is the port, you could try another port here if you set your
 phone to another port under Advanced
 6670 is the extension of the phone
 192.168.51.251 is the IP of the PHONE, not the Asterisk server.  It
 does not appear that you can use the IP of the Asterisk server.

 You can get a list of phones with IPs using the Asterisk command sip
 show peers.  Our intent is to build a simple database matching
 extension to IP and then execute sipsak commands from a script, probably
 in the manager API, when agents log in and out that will update the
 phone display accordingly.

 I hope this is helpful to some of you.

 Franklin Webb
 InterMedia Marketing Solutions

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Re: [Asterisk-Users] A good SIP receptionist phone

2005-05-19 Thread Sven Fischer (support)
On Monday 02 May 2005 20:10, Pedro wrote:
 What I did once was create an announcement that got played to the
 receptionist announcing who the call was for based on the number that
 was called.  This allowed the receptionist to know which greeting to
 recite.

Cool idea !


 On 5/2/05, Michael Welter [EMAIL PROTECTED] wrote:
  Chris Mason (Lists) wrote:
   The user name is the extension and the password is always the same. Not
   hard to configure.
 
  With the SNOM 220, you have five buttons/lamps that can be used as
  line appearances--these buttons can each register to a different SIP
  URL.
 
  Each sidecar has 20 buttons/lamps, and you may have up to three
  sidecars.  Using the hint priority in Asterisk, the buttons serve as
  extension busy lamps.  You can also use these buttons to transfer calls.
 
  I have an executive suites customer where each tenant is a separate
  business.  For an incoming call, the attendant needs to know which DID
  number is being called so she can answer with the proper greeting.
 
  I would like the sidecar buttons to be able to register to a SIP URL, so
  an incoming call would blink the tenants button, but that is not
  possible--I can only use the five buttons on the phone for that purpose,
  and there are more than five tenants.
 
  A suggestion was to alter the Called ID Name to the DID number.  This
  would work for the attendant, but the tenant would like to see the
  original Caller ID Name.
 
  I would rather not have to put a PC at the attendants position, but that
  is the way this is shaping up.  Does anyone have any suggestions?
 
  Thanks,
 
 
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Re: [Asterisk-Users] snom soft phone

2005-02-08 Thread Sven Fischer (support)
No,  is default for snom phones.

Sven

On Tuesday 08 February 2005 08:37, Altus Snyman wrote:
 Did you try 00
 That is what it is on the 220

 On Tue, 2005-02-08 at 09:36, Paradise Dove wrote:
  what is the password for Administrator in the softphone?
 
 
  On Tue, 8 Feb 2005 08:01:07 +0100, Christian Stredicke
 
  [EMAIL PROTECTED] wrote:
   Go to the web page, in Preferences there are two pull down menus for
   Audio Input and Autio Output.
  
   CS
  
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Juan J. Sierralta P.
Sent: Tuesday, February 08, 2005 2:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] snom soft phone
   
Hi,
   
 How do I change the default audio device ?
 I have one of those USB headset (which actually is another
soundcard) but the simulation insist in using my Soundblaster
Live card :(
   
   
--
Juanjo sin .sig :(
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Re: [Asterisk-Users] Re: Snom 190 - dhcp - settings_server

2004-11-22 Thread Sven Fischer (support)
On Monday 22 November 2004 16:01, Stefan Tichy wrote:
 On Sun, Nov 21, 2004 at 05:30:13PM +0200, Pertti Pikkarainen wrote:
  However I would use a more specific path for a web-server ;-)  Something
  like:
 
  option tftp-server-name http://192.168.0.9/snom/snom200.htm

 But for the snom 190 tftp-server-name in dhcp config will set
 update_server. The field/variable setting_server remains empty.

 The documentation suggests that dhcp data can be used to define
 setting_server. Just a bug in the snom 190 firmware ?

no, for the dhcp option 66 its ok like that ! Its internally handled 
correctly.

Sven



 Best regards

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Re: [Asterisk-Users] Snom200 strange sound problem

2004-10-29 Thread Sven Fischer (support)
Hi,

sounds for me like a typical NAT problem. I would recommend to use a more 
recent version like 3.52 or 3.56 you can find here

http://www.snom.com/download/share/

Regards,

Sven

On Friday 29 October 2004 08:12, Pamela Weis wrote:
 Hello group,

 I've a rather strange problem with my Snom200 telephone.
 I'm using it in combination with SER, asterisk and rtpproxy.
 The telephone is behind NAT and connects to SER. It can be called
 without any problem from any Client on asterisk or SER.
 But whenever I make a call to asterisk or other clients on asterisk I
 have no sound until I press the Transf and Esc button in sequence.
 With XLite and a Snom100 I don't have these problems. They work fine in
 all directions, therefore I don't believe that it's a configuration
 issue on asterisk or ser side but I maybe wrong.
 I have already tried to up- and downgrade the snom200 to another
 firmware (current version is now snom200-SIP 2.04g) but it didn't help.

 Any help on this would be greatly appreciated.

 Pamela

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Re: [Asterisk-Users] SNOM 190 - strange voice problems

2004-10-26 Thread Sven Fischer (support)
On Tuesday 26 October 2004 02:36, Ulexus wrote:
 Jeremy Rusnak wrote:
  Hi all,
 
 ...snip...
 
  We're running SIP and version 3.46 of the phone firmware.

 I don't know about this specific problem, but the latest firmware is
 3.55.  I'd try it.  http://www.snom.com/download/share

exactly ! Please update at least to 3.52e.

Regards,

Sven



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Re: [Asterisk-Users] Snom 190 VMail Soft Key

2004-10-21 Thread Sven Fischer (support)
Hi,

you can switch it of by setting

Dialtone during Hold:

to off on the advanced Webpage.

Regards,

Sven

On Wednesday 20 October 2004 19:57, Ronald Hartmann wrote:
 Problem fixed for the Vmail Soft Key

  I had the userfrom in the wrong context.

 Anyhelp on the hold button is still appreciated.


 Good day list,

 Need some assistance in setting up the snom190 with
 asterisk.

 My voicemail server is at extension *98.

 I have successfully been able to leave a message for an
 extension
 The MWI comes on and I can check messages by dialing *98 on
 the snom and all is great.

 HOWEVER, if I press the VMAIL button I get nowhere.

 Here is the trace that is sent to turn on the MWI

 Received from udp:192.168.3.11:5060 at 20/10/2004 12:22:48:890 (495
 bytes):
 NOTIFY sip:[EMAIL PROTECTED]:5060;line=g6az8l1z SIP/2.0
 Via: SIP/2.0/UDP 192.168.3.11:5060;branch=z9hG4bK7f8a0e8a
 From: Unknown sip:[EMAIL PROTECTED];tag=as234f5c67
 To: sip:[EMAIL PROTECTED]:5060;line=g6az8l1z
 Contact: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 NOTIFY
 User-Agent: Asterisk PBX
 Event: message-summary
 Content-Type: application/simple-message-summary
 Content-Length: 39

 Messages-Waiting: yes
 Voicemail: 1/0
 I would guess that somewhere I need to tell asterisk to send the from
 address as  mailto:[EMAIL PROTECTED] [EMAIL PROTECTED], however this is
 not clear to me how to do this.
 Have read the wiki on this and have tried to insert
 fromuser=*98 in the sip.conf file however no good.

 As a side note...

 If I place a caller on hold from the snom phone all is well
 the caller gets music on hold.  However the snom phone keeps giving me a
 stutter dial tone.  Is it possible to make the
 Snom 190 be quiet.  Maybe I just need to run and get a file to review
 and I do not like the phone to keep making that outrageous sound until I
 pick them back up.

 Thanks so much for helping out an asterisk newbie.

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Re: [Asterisk-Users] Re: Dial group continues to ring after answer -SNOM phones and solution

2004-10-19 Thread Sven Fischer (support)
Hi,

On Monday 11 October 2004 19:12, Dave Cotton wrote:
 On Mon, 2004-10-11 at 11:51 -0500, Mike Meyer wrote:
  Someone pointed me here 
  
  http://www.snom.com/downloads/share

  http://www.snom.com/download/share

  !
  That where the SNOM support team sent me. Seems that they may be
  suggesting a different process or URL do update from. My concern is
  whether the latest version 3.54 has been tested and is an official
  release. I hate to put something out that hasn't been through a
  sufficient QA process. I don't want to risk getting my user's mad at me
  with a bad version of software.

 I've been working though the 3.5x series and haven't noticed any real
 nasties yet.

 Out of interest has anyone worked out how to use the Action URL
 settings?

Some few specific events on the phone can trigger web get requests to the 
configured URLs. Like lifting the handset is triggering there is some action 
going on on the phone etc...  

Regards,

Sven
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Re: [Asterisk-Users] Re: Dial group continues to ring after answer -SNOM phones and solution

2004-10-19 Thread Sven Fischer (support)
Hi,

no, it is

 http://www.snom.com/download/share

!!!

Sven

On Monday 11 October 2004 17:18, Alex Barnes wrote:
 Someone pointed me here 

 http://www.snom.com/downloads/share (had to guess at URL as the Snom
 site appears to be down or uber slow but if that's not it its damn close

 :-P )

 Which lists all versions of firmware for all their phones.  Handy if you
 have a specific version in mind but don't know the correct URL.  Tho I
 haven't had problems with the auto-update so far.

 HTH

 alex

 -Original Message-
 From: Mike Meyer [mailto:[EMAIL PROTECTED]
 Sent: 11 October 2004 16:12
 To: Asterisk Users Group
 Subject: [Asterisk-Users] Re: Dial group continues to ring after answer
 -SNOM phones and solution


 Asterisk Users;

  Just wanted to let you know I fixed my problem.

 To follow up on my own testing of the situation, I find that the
 continued ringing after pickup only occured on the SNOM phones in the
 group. The Grandstream phones stop ringing when another phone picks up.

 Having turned on SIP debugging I have verified that the cancel message
 is sent to the SNOM phone (and others in the group) when one of the
 group phones is picked up, as expected.

 It appears that the SNOMs don't handle the cancel message the same as
 the Grandstream. I was using SIP 2.03o firmware on the SNOM which is the
 latest official release.

 It seems that these phones even though they are set to do automatic
 update, they do not. Or perhaps that capability was broken in the
 firmware version I had last updated to.

 THE SOLUTION:
 To remedy the problem I upgraded to version 3.52 beta version. Also
 2.04g fixes this problem as well.

 I had to create my own internal TFTP server and flash update to 3.52.
 The standard update process did not work to go beyond 2.03y or 2.04g. I
 tried 2.05e  f and these would never come out of boot.

 MORAL TO THE STORY: Keep your SIP phone firmware up to date.

 SNOM support is telling me to upgrade to 3.54. I don't see this one
 listed on the standard update URL. I am a little leery about moving to
 that one.

 Now to upgrade my GrandStream's. They seem to be stuck at an old version
 as well.

 Thanks,
 Mike Meyer

 On Tue, 2004-10-05 at 16:47, Mike Meyer wrote:
  Asterisk Users:
 
   We have our * dial plan set up to ring 5 phones in the office

 and it

  delivers the call to the first that picks up their receiver.
   The problem is that after the pickup, everyone else's SIP phone

 keeps

  ringing at least once and sometimes twice. This interferes with the
  conversation, while others pick up the phone and get nothing.
 
   Does anyone else have similar problems or have a solution to

 stop the

  ring once answered? My dial statement looks like the following and has
 
  a timeout of 15 seconds.
 
  exten = MainTeam,1,Dial(${MainTeamChannels},15,tr)
  exten = MainTeam,2,Voicemail(u${MainTeam_EXT})
  ...
  note the variables MainTeamChannels define the SIP phone channels
  defined in sip.conf and MainTeam_EXT is the voicemail box for this
  group extension.
 
   As an alternative, I am considering doing a round robin on a

 call

  group or pickup group and implementing call pickup.
 
  Any ideas welcome.
 
  Mike Meyer

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 Dear Friends of Ubiquity Software:

 As you may have noticed, Ubiquity Software began using the web domain
 ubiquity.com earlier this year in addition to the previously established
 ubiquity.net for our website and email communications to you.  However,
 since that time, a dispute has emerged with respect to actual ownership of
 the ubiquity.com domain.

 As an international software company founded over decade ago, you can
 always reach Ubiquity Software under the website www.ubiquity.net
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Re: [Asterisk-Users] Intercept HOLD of Snom phones

2004-10-19 Thread Sven Fischer (support)
Hi,

do a SIP trace or PCAP trace of the scenario via the webinterface and you will 
see exactly, what is going on...

Regards,

Sven

On Thursday 14 October 2004 21:53, Magnus Jungsbluth wrote:
 Hi,

 I'm running the 1.0 release of Asterisk an it is working nicely with our
 snom 105 phones. Hold puts the caller on hold, attended / unattended
 Transfer works directly with the snom buttons ...
 I have one question though: what does the snom exactly do to tell the *
 to put the call on hold (can I intercept this somewhere)?

 I would like to decide using the callerid which music on hold is tobe
 played: That is: play free music to calls from the outside but play
 copyrighted music if I put an internal call on hold (i.e. a co-worker).
 Is this possible ?

 regards,
 Magnus Jungsbluth
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Re: [Asterisk-Users] Snom Mass Deployment Config Problems

2004-10-19 Thread Sven Fischer (support)
Hi,

I'm sure a lot people can help you here, maybe I'm the first. See below 
inline:

On Tuesday 19 October 2004 11:20, Alex Barnes wrote:
 Hi all,

 I am hoping that someone out there is using the Snom phones
 configuration via HTTP server functionality.
 I have downloaded and read the FAQ many times but I am having trouble
 getting the settings to take effect.  Probably as I haven't formatted
 things correctly.  For example the fkey settings aren't taking effect.

 If someone is willing to email me (directly to save spamming the list is
 fine) a working settings file that would help me alot.

 thanks a lot for any help

 Alex

 [EMAIL PROTECTED]

 -

 html
 pre

 #Basic Settings
 phone_name: Snom 6107
 dhcp: true
 call_completion: true
 auto_dial: 10
 admin_mode_password: 

 #Line Settings

the brackets are wrong:

 user_realname[1]: Snom 200
 user_name[1]: snom
 user_host[1]: 172.16.0.217
 user_pass[1]: snom
 user_transport[1]: udp
 user_expiry[1]: 3600
 user_mailbox[1]: 8500
 user_outbound[1]: 172.16.0.217


it should be like:

user_realname1: Snom 200
user_name1: snom
user_host1: 172.16.0.217
user_pass1: snom
user_transport1: udp
user_expiry1: 3600
user_mailbox1: 8500
user_outbound1: 172.16.0.217

 #SIP Settings
 nat_detection: off
 tcp_threshold: udp
 publish_presence: true

 #Codec Settings
 dtmf_type_inband: false
 utc_offset: 0
 ntp_server: 193.195.52.24

 #Network Settings
 http_proxy: 193.195.52.26
 http_port: 8001

 #Update Settings
 setting_server: http://sqa5.sqa.net/test/snom200.htm

what is this ? remove it.

 http://sqa5.sqa.net/test/snom200.htm


 #Misc Settings
 tone_scheme: GBR

here again the brackets are wrong:

 fkey1: dest [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 fkey2: dest [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
 fkey[3]: dest [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
 fkey[4]: dest [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]


fkey1: dest sip:[EMAIL PROTECTED];user=phone
fkey2: dest sip:[EMAIL PROTECTED];user=phone
fkey3: dest sip:[EMAIL PROTECTED];user=phone
fkey4: dest sip:[EMAIL PROTECTED];user=phone


 /pre
 /html


BTW did you saw our FAQ regarding massdeployment ?

kind regards,

Sven Fischer





 Dear Friends of Ubiquity Software:

 As you may have noticed, Ubiquity Software began using the web domain
 ubiquity.com earlier this year in addition to the previously established
 ubiquity.net for our website and email communications to you.  However,
 since that time, a dispute has emerged with respect to actual ownership of
 the ubiquity.com domain.

 As an international software company founded over decade ago, you can
 always reach Ubiquity Software under the website www.ubiquity.net
 http://www.ubiquity.net/  and via email at @ubiquity.net.  However, we
 have also chosen to expand our domain to the more specific
 www.ubiquitysoftware.com http://www.ubiquitysoftware.com/  for web and
 @ubiquitysoftware.com for email communications.

 Please use either the historical ubiquity.net or begin to use the new
 ubiquitysoftware.com domain for all email communications to Ubiquity
 employees from now on.

 Thank you.

 Regards,

 Ubiquity Software
 www.ubiquitysoftware.com http://www.ubiquitysoftware.com/
 [EMAIL PROTECTED]

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