Re: [asterisk-users] Snom870 sidecar

[Christian Stredicke]

The sidecar is not in the market yet. Just some information... It has
its own CPU, Ethernet port and it is able to run applications (for
example, Asterisk).

 

CS

 

Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Olivier
Gesendet: Sonntag, 18. Oktober 2009 12:46
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Snom870 sidecar

 

Hi,

Watching Snom 870's video (http://www.youtube.com/watch?v=9e8hPxX0oDU),
you can see a new sidecar (phone extension) which seem very interesting.
Has someone details on this extension ?
Any release date or/and data sheet ?

Regards

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Re: [asterisk-users] Snom Phones Registration/Failover Feature

[Christian Stredicke]

Check out the "Failover Identity" ("Ersatz Identität") in the identity 
settings. Works a little bit different, but you can achieve the same effect 
with this.

CS

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Raimund Sacherer
Gesendet: Donnerstag, 13. August 2009 16:16
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Snom Phones Registration/Failover Feature

Hello Mailinglist,

i was reading a paper regarding a Asterisk clustering solution and  
they where pretty excited about a feature in polycom phones:

You can add a registration to a primary asterisk server
You can add a registration to a secondary asterisk server

The polycom phones will talk to the primary server as long as all goes  
well, If they have a problem with an INVITE, they automatically  
register to the secondary asterisk server and start using them. Every  
few seconds (I think it was 30) the phone tries again to register on  
the primary server, if this succeeds, it uses the primary again.

This is in my oppinion a pretty decent way of doing failover (reminds  
me of radius). It beats using Heartbeat and IP Takeover and all the  
hassle you (could) have with this solution.

I was reading in the documentation about the SNOM phones (mainly 300)  
but I did not find anything in the users-pdf's or on there  
knowledgebase/website which would tell me if this is possible, there  
is something for failover configuration but it is not explained at all.

It's highly appreciated if someone with insight could explain to me or  
point me to the right documentation on how/if this works with SNOM's.

Thank you,
best regards

-- 
Raimund Sacherer
-
RunSolutions
Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares


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Re: [asterisk-users] IP phone recommendation

[Christian Stredicke]

Check out the snom 300 or the snom 820...

CS

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Rilawich Ango
Gesendet: Mittwoch, 3. Juni 2009 09:45
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] IP phone recommendation

Hi all,
  Any good recommendation of IP phone in term of sound quality and
price (reasonable) using with asterisk?
ango

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Re: [asterisk-users] Rusting Snoms?

[Christian Stredicke]

I would not do a firmware upgrade from such an old version. That will probably 
mean a lot of (more) trouble. "Never touch a running system"... New firmware 
might have the option to set the link speed, but IMHO it is easier to do that 
on the switch.

CS

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Tim Panton
Gesendet: Dienstag, 19. Mai 2009 15:46
An: Christian Stredicke
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Rusting Snoms?

It isn't POE - its using the original power brick that came with the phone.

Swapping to a dumb hub (without uplink-autosensing) seems to fix it.

Would a firmware upgrade (from 3,56m 6154) help ?

Tim.

On 19 May 2009, at 13:28, Christian Stredicke wrote:

> With cheap PoE devices Ethernet can easily get "on the edge" - or over 
> the edge. If you have another switch/different model, a quick try will 
> help isolating the problem.
>
> CS
>
> -Ursprüngliche Nachricht-
> Von: Tim Panton [mailto:t...@westhawk.co.uk]
> Gesendet: Dienstag, 19. Mai 2009 13:46
> An: Asterisk Users Mailing List - Non-Commercial Discussion; Christian 
> Stredicke
> Betreff: Re: [asterisk-users] Rusting Snoms?
>
> On further investigation - it may well be that the switch doesn't like 
> the phones (or vice-versa) I tried daisy-chaining one phone off the 
> second port of the other and got distinctly better audio.
>
> It's a new netgear fvs 318 with autosensing 100/10 ports.
>
> Any clues ?
>
> Thanks.
> Tim
>
> On 9 May 2009, at 11:04, Christian Stredicke wrote:
>
>> Because the phone is a digital system, I would suspect that it is a 
>> problem with the switch. Run a quick PCAP trace to see where the 
>> jitter comes from. Depending on the firmware version, you can do that 
>> from the web interface.
>>
>> CS
>>
>> -Ursprüngliche Nachricht-
>> Von: asterisk-users-boun...@lists.digium.com 
>> [mailto:asterisk-users-boun...@lists.digium.com
>> ] Im Auftrag von Tim Panton
>> Gesendet: Samstag, 9. Mai 2009 11:46
>> An: Asterisk Users Mailing List - Non-Commercial Discussion
>> Betreff: [asterisk-users] Rusting Snoms?
>>
>> This is a bit off topic, because I 'think' it isn't an Asterisk 
>> problem.
>> However I'm not sure and anyhow I'm hoping someone may recognize the 
>> symptom.
>>
>> We moved offices a month ago. Our trusty SNOM190s (all between 3 and
>> 5 years old) were packed up for the move, then unpacked a couple of 
>> weeks later.
>>
>> On unpacking them and connecting them to the new network, several of 
>> them didn't work well. The symptom is that outgoing RTP audio is 
>> garbled - like the packets are pulsed. Inbound is fine. This isn't 
>> true for all of the phones, just some of them. (The all run the same 
>> SNOM firmware)
>>
>> To be fair, they are on a new network, so it could be the cables or 
>> new 1Gb switches, except that the problem moves with the phone if you 
>> relocate it from one desk to another.
>>
>> I've tried a fresh asterisk install, but that didn't help either.
>>
>> So I am forced to conclude that something went 'bad' in those
>> (old) phones while they were switched off. Has anyone got any clues 
>> for me?
>>
>> Thanks!
>>
>> Tim.
>>
>> Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk
>>
>>
>>
>>
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>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>  http://lists.digium.com/mailman/listinfo/asterisk-users
>
> Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk
>
>
>

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




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Re: [asterisk-users] Rusting Snoms?

[Christian Stredicke]

With cheap PoE devices Ethernet can easily get "on the edge" - or over the 
edge. If you have another switch/different model, a quick try will help 
isolating the problem.

CS

-Ursprüngliche Nachricht-
Von: Tim Panton [mailto:t...@westhawk.co.uk] 
Gesendet: Dienstag, 19. Mai 2009 13:46
An: Asterisk Users Mailing List - Non-Commercial Discussion; Christian Stredicke
Betreff: Re: [asterisk-users] Rusting Snoms?

On further investigation - it may well be that the switch doesn't like  
the phones (or vice-versa)
I tried daisy-chaining one phone off the second port of the other and  
got distinctly better audio.

It's a new netgear fvs 318 with autosensing 100/10 ports.

Any clues ?

Thanks.
Tim

On 9 May 2009, at 11:04, Christian Stredicke wrote:

> Because the phone is a digital system, I would suspect that it is a  
> problem with the switch. Run a quick PCAP trace to see where the  
> jitter comes from. Depending on the firmware version, you can do  
> that from the web interface.
>
> CS
>
> -Ursprüngliche Nachricht-
> Von: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com 
> ] Im Auftrag von Tim Panton
> Gesendet: Samstag, 9. Mai 2009 11:46
> An: Asterisk Users Mailing List - Non-Commercial Discussion
> Betreff: [asterisk-users] Rusting Snoms?
>
> This is a bit off topic, because I 'think' it isn't an Asterisk  
> problem.
> However I'm not sure and anyhow I'm hoping someone may recognize the  
> symptom.
>
> We moved offices a month ago. Our trusty SNOM190s (all between 3 and  
> 5 years old) were packed up for the move, then unpacked a couple of  
> weeks later.
>
> On unpacking them and connecting them to the new network, several of  
> them didn't work well. The symptom is that outgoing RTP audio is  
> garbled - like the packets are pulsed. Inbound is fine. This isn't  
> true for all of the phones, just some of them. (The all run the same  
> SNOM firmware)
>
> To be fair, they are on a new network, so it could be the cables or  
> new 1Gb switches, except that the problem moves with the phone if  
> you relocate it from one desk to another.
>
> I've tried a fresh asterisk install, but that didn't help either.
>
> So I am forced to conclude that something went 'bad' in those
> (old) phones while they were switched off. Has anyone got any clues  
> for me?
>
> Thanks!
>
> Tim.
>
> Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk
>
>
>
>
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

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www.westhawk.co.uk




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Re: [asterisk-users] Rusting Snoms?

[Christian Stredicke]

It could be that the PCAP resolution is 10 ms. The phones don't have a 
real-time clock... But 50 ms is definitively not okay.

CS

-Ursprüngliche Nachricht-
Von: Tim Panton [mailto:t...@westhawk.co.uk] 
Gesendet: Dienstag, 19. Mai 2009 13:46
An: Asterisk Users Mailing List - Non-Commercial Discussion; Christian Stredicke
Betreff: Re: [asterisk-users] Rusting Snoms?

Christian, thanks, I'd never run pcap in a phone before - cool.

The trace shows jitter - but in a weird way. some of the packets have  
delta's of
 > 20 ms but always a multiple of 10 so 50 and 30 occur, as do 10 and 0.

Is that normal ?

Tim.


On 9 May 2009, at 11:04, Christian Stredicke wrote:

> Because the phone is a digital system, I would suspect that it is a  
> problem with the switch. Run a quick PCAP trace to see where the  
> jitter comes from. Depending on the firmware version, you can do  
> that from the web interface.
>
> CS
>
> -Ursprüngliche Nachricht-
> Von: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com 
> ] Im Auftrag von Tim Panton
> Gesendet: Samstag, 9. Mai 2009 11:46
> An: Asterisk Users Mailing List - Non-Commercial Discussion
> Betreff: [asterisk-users] Rusting Snoms?
>
> This is a bit off topic, because I 'think' it isn't an Asterisk  
> problem.
> However I'm not sure and anyhow I'm hoping someone may recognize the  
> symptom.
>
> We moved offices a month ago. Our trusty SNOM190s (all between 3 and  
> 5 years old) were packed up for the move, then unpacked a couple of  
> weeks later.
>
> On unpacking them and connecting them to the new network, several of  
> them didn't work well. The symptom is that outgoing RTP audio is  
> garbled - like the packets are pulsed. Inbound is fine. This isn't  
> true for all of the phones, just some of them. (The all run the same  
> SNOM firmware)
>
> To be fair, they are on a new network, so it could be the cables or  
> new 1Gb switches, except that the problem moves with the phone if  
> you relocate it from one desk to another.
>
> I've tried a fresh asterisk install, but that didn't help either.
>
> So I am forced to conclude that something went 'bad' in those
> (old) phones while they were switched off. Has anyone got any clues  
> for me?
>
> Thanks!
>
> Tim.
>
> Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk
>
>
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users

Tim Panton - Web/VoIP consultant and implementor
www.westhawk.co.uk




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Re: [asterisk-users] Rusting Snoms?

[Christian Stredicke]

Because the phone is a digital system, I would suspect that it is a problem 
with the switch. Run a quick PCAP trace to see where the jitter comes from. 
Depending on the firmware version, you can do that from the web interface.

CS

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Tim Panton
Gesendet: Samstag, 9. Mai 2009 11:46
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Rusting Snoms?

This is a bit off topic, because I 'think' it isn't an Asterisk problem.
However I'm not sure and anyhow I'm hoping someone may recognize the symptom.

We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years 
old) were packed up for the move, then unpacked a couple of weeks later.

On unpacking them and connecting them to the new network, several of them 
didn't work well. The symptom is that outgoing RTP audio is garbled - like the 
packets are pulsed. Inbound is fine. This isn't true for all of the phones, 
just some of them. (The all run the same SNOM firmware)

To be fair, they are on a new network, so it could be the cables or new 1Gb 
switches, except that the problem moves with the phone if you relocate it from 
one desk to another.

I've tried a fresh asterisk install, but that didn't help either.

So I am forced to conclude that something went 'bad' in those
(old) phones while they were switched off. Has anyone got any clues for me?

Thanks!

Tim.

Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk




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Re: [asterisk-users] Asterisk Security

[Christian Stredicke]

Check out http://ucsniff.sf.net. You can run it on the PC of your choice in the 
network (e.g. your PC) and then record the conversations.

Recording calls in the LAN is a lot more interesting than recording random 
calls that run over the Internet. Examples:

* Your boss intends to fire you and wants to talk it through with HR.

* Your customer is calling your boss and complains about you.

* Your colleague wants to get your job and develops a strategy to make you look 
stupid.

* Two colleagues have an office love affair and nobody should know about it.

This list can be extended.

I believe this makes clear that security should not be an afterthought in your 
next VoIP installation.

CS

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von SIP
Gesendet: Montag, 6. April 2009 14:31
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Asterisk Security

If that someone is between you and the other endpoint (like between you
and the switch, or using port-mirroring on a router somewhere), then
yes. The conversations can be recorded. In the US, the ability to be
able to do this is required by law. You've little to worry about random
hackers coming in off the Internet for this sort of thing. It's usually
something to do with having physical access to the network in which you
or the other end is connected.

There's ARP poisoning and the like which could make this possible in a
local network environment on either side, but for the most part, you'll
know who's on your local net, and they likely have physical access to
your phones as well. A listening device would be easier to plant in the
mic pickup of your phone if they REALLY wanted to listen in on your calls.

There are all sorts of levels one can to to find out what you're doing,
and preventing against them can involve a great deal of creativity.

That said, the answer is yes. You could use a VPN tunnel from one end to
the other, and many people do just that to help ensure the privacy of
their connections (both data and voice).

N.

Tom wrote:
> Since we are talking about security, if I am using * to talk to a cisco
> gateway via SIP, is there some sort of encryption you can use?  Like a 
> vpn tunnel?  
>
> Can someone capture packets and re-assemble to make out a conversation?
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
> Sent: Saturday, April 04, 2009 7:20 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk Security
>
> Lets not be that paranoid. If you have these ports open to the internet then
> from time to time someone will check if your default unsecured context
> can dial out to PSTN...
>
> with sip.conf you can add
>
> allowguest=no
>
> With IAX2 there's no allowguest but I believe you have to have a guest
> username in iax.conf with no password to access
> the unsecured context.
>
> Martin
>
> On Sat, Apr 4, 2009 at 3:42 PM, Todd Reese  wrote:
>   
>> Hi All,
>>
>> Coming in to day, the logs on the asterisk server showed several entries
>> such as:
>>
>> [Apr  4 15:25:16] NOTICE[9280]: chan_sip.c:14627 handle_request_invite:
>> Call from '' to extension '9810380487965419' rejected because extension
>> not found.
>>
>> This has gotten me to thinking about security of this box.
>>
>> 1. Currently the box sits behind a firewall with iax and sip ports
>> pointing to it for the ip phones that are offsite.  There isn't any
>> other access through the firewall to this box.
>> 2. All devices have an extension assigned to them in sip.conf and
>> extensions.conf.  i.e. supra ATA, Grandstream GXP-2000
>> 3. The box is fed via Les.net and Voicepluse.  All other feeds are
>> shutoff when not active.
>>
>> I'm looking for ideas to tighten up on the security so that this won't
>> happen again.
>>
>> TIA,
>>
>> Todd Reese
>>
>>
>>
>>
>>
>>
>>
>>
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> No virus found in this incoming message.
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> Version: 8.0.238 / Virus Database: 270.11.41/2040 - Release Date: 04/04/09
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Re: [asterisk-users] building a phone

[Christian Stredicke]

I agree, the intersting part is adding what is not included in the standard 
firmware.

Regarding documentation... On the one hand the phone is running a "regular" 
embedded Linux, I think that does not require additional documentation. The API 
to the phone is a different topic. It will really depend what content we are 
talking about.

Many applications can be done using the mini-browser. The software does not 
even have to run on the phone for that. Maybe a concrete example of an 
application that has to run locally on the phone would be useful.

CS

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Paul Chambers
Gesendet: Mittwoch, 4. März 2009 05:15
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] building a phone

It may not be necessary to replace Snom's firmware to add interesting 
functionality to the product. Though that was not the original poster's 
premise, admittedly.

As to the 'loose ends', they usually remain so until someone is 
motivated to drive them to closure. Absence of a suitable hardware 
platform is guaranteed to perpetuate that situation :)

The big question for me is whether Snom would provide some documentation 
to those prepared to invest their time. With GPL'd software likely to be 
part of the mix, such information couldn't be covered by a 
non-disclosure or some restrictive developer agreement.

One of the things that helps to kick-start a developer community is to 
sell 'developer kits' (like Digium did). Single-unit quantities with a 
'not-for-resale' provision, perhaps with membership of some developer 
program.

Paul

Christian Stredicke wrote:
> To be honest: I am not very optimistic regarding this project. 
>
> The WRT is really a case where you essentially use stuff that is already 
> available and which is very very stable (e.g. Linux). There is nothing really 
> special for the WRT. 
>
> For a phone, the picture looks different. There are so many components 
> necessary that are either not available or not very stable. There is a 
> tremendous risk of ending up with a project that has a lot of loose ends. 
>
> But if someone wants to give it a try, sure. We have nothing to lose! Those 
> who know embedded Linux will easily feel like home on the phone once they are 
> logged in.
>
> Definitively an interesting topic for our Asterisk developer meeting that we 
> want to run this month in Berlin.
>
> Maybe for starters we just compile an Asterisk and run it on the phone. That 
> will be fun!
>
> CS
>
> -Ursprüngliche Nachricht-
> Von: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Michael Graves
> Gesendet: Sonntag, 1. März 2009 18:30
> An: Asterisk Users Mailing List - Non-Commercial Discussion
> Betreff: Re: [asterisk-users] building a phone
>
> On Sun, 1 Mar 2009 18:14:18 +0100, Christian Stredicke wrote:
>
>   
>> I have influential contacts inside snom...
>>
>> CS
>> 
>
> So you do! What do you think? Would snom be interested in selling
> hardware into a group of users who would be loading community developed
> application firmware?
>
> It makes me wonder how many little routers Cisco sells that actually
> get loaded with WRT-DD and the like?
>
> Michael
>
>   
>> -Ursprüngliche Nachricht-
>> Von: asterisk-users-boun...@lists.digium.com 
>> [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Paul Chambers
>> Gesendet: Sonntag, 1. März 2009 01:30
>> An: Asterisk Users Mailing List - Non-Commercial Discussion
>> Betreff: Re: [asterisk-users] building a phone
>>
>> Michael Graves wrote:
>> 
>>> On Sat, 28 Feb 2009 14:59:23 -0800, Paul Chambers wrote:
>>>   
>>>> Michael Graves wrote:
>>>> 
>>>>> Witness the fact that the old Pingtel phones ran Java, and they were 
>>>>> incredibly lame.
>>>>>
>>>>> I think part of what this thread misses is that DSP is a god chunk of 
>>>>> what SIP phones need. A general purpose CPU is not the right tool for 
>>>>> the task. A cheap DSP is better suited to compression, transcoding, etc.
>>>>>
>>>>> OTOH, presuming that the snom phones are Linux on a suitable platform 
>>>>> soomeone could develop a custom software load for them and OEM the 
>>>>> hardware.
>>>>>   
>>>> I'm surprised that no-one has mentioned Astfin. Basically uClinux and 
>>>> asterisk running on an Analog Devices Bla

Re: [asterisk-users] building a phone

[Christian Stredicke]

To be honest: I am not very optimistic regarding this project. 

The WRT is really a case where you essentially use stuff that is already 
available and which is very very stable (e.g. Linux). There is nothing really 
special for the WRT. 

For a phone, the picture looks different. There are so many components 
necessary that are either not available or not very stable. There is a 
tremendous risk of ending up with a project that has a lot of loose ends. 

But if someone wants to give it a try, sure. We have nothing to lose! Those who 
know embedded Linux will easily feel like home on the phone once they are 
logged in.

Definitively an interesting topic for our Asterisk developer meeting that we 
want to run this month in Berlin.

Maybe for starters we just compile an Asterisk and run it on the phone. That 
will be fun!

CS

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Michael Graves
Gesendet: Sonntag, 1. März 2009 18:30
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] building a phone

On Sun, 1 Mar 2009 18:14:18 +0100, Christian Stredicke wrote:

>I have influential contacts inside snom...
>
>CS

So you do! What do you think? Would snom be interested in selling
hardware into a group of users who would be loading community developed
application firmware?

It makes me wonder how many little routers Cisco sells that actually
get loaded with WRT-DD and the like?

Michael

>-Ursprüngliche Nachricht-
>Von: asterisk-users-boun...@lists.digium.com 
>[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Paul Chambers
>Gesendet: Sonntag, 1. März 2009 01:30
>An: Asterisk Users Mailing List - Non-Commercial Discussion
>Betreff: Re: [asterisk-users] building a phone
>
>Michael Graves wrote:
>> On Sat, 28 Feb 2009 14:59:23 -0800, Paul Chambers wrote:
>>   
>>> Michael Graves wrote:
>>> 
>>>> Witness the fact that the old Pingtel phones ran Java, and they were 
>>>> incredibly lame.
>>>>
>>>> I think part of what this thread misses is that DSP is a god chunk of 
>>>> what SIP phones need. A general purpose CPU is not the right tool for 
>>>> the task. A cheap DSP is better suited to compression, transcoding, etc.
>>>>
>>>> OTOH, presuming that the snom phones are Linux on a suitable platform 
>>>> soomeone could develop a custom software load for them and OEM the 
>>>> hardware.
>>>>   
>>> I'm surprised that no-one has mentioned Astfin. Basically uClinux and 
>>> asterisk running on an Analog Devices Blackfin DSP. There's also some 
>>> 'open source' hardware that's available - the IP04 and friends. I'm 
>>> using an Edgepbx FX08, and they also have a two-port version (FX02). 
>>> Atcom has a single-port one, the IP01.
>>>
>>> Though if I were going to prototype an 'open' SIP phone, I'd probably 
>>> start with a beagle board (TI OMAP3530 - dual-core ARM+DSP). It's a 
>>> pretty powerful SOC - its brother (3430) powers the Palm Pre.
>>>
>>> Just another datapoint :)
>>> 
>>
>> Yeah, that'd be great hardware to select. 
>>
>> What I was thinking is that this thread seems to be driven by those of
>> a software bent. For that group perhaps there's an opportunity to write
>> code for something like a snom 820. It's a solid solid hardware basis
>> for the project. Snom would be foolish not to sell it for such use,
>> even price it attractively. That way the hardware work would be done,
>> and the software geeks could work their magic.
>>   
>I'm a card-carrying (embedded linux) software geek, and I know I'd be 
>interested :)
>
>Anyone got some influencial contacts inside Snom? or Aastra, for that 
>matter, their hardware also seems good quality from what people have said.
>
>Another possibility is talking to Atcom (or other VoIP ODMs), they seem 
>to have done pretty well from the IP04 and derivatives. They've 
>experienced the benefits of an open development model, perhaps they'd be 
>interested. Not sure what the quality of their existing handset hardware 
>is like.
>
>Anyone on the list have the contacts to get the ball rolling?
>
>Paul
>
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Re: [asterisk-users] building a phone

[Christian Stredicke]

I have influential contacts inside snom...

CS

-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Paul Chambers
Gesendet: Sonntag, 1. März 2009 01:30
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] building a phone

Michael Graves wrote:
> On Sat, 28 Feb 2009 14:59:23 -0800, Paul Chambers wrote:
>   
>> Michael Graves wrote:
>> 
>>> Witness the fact that the old Pingtel phones ran Java, and they were 
>>> incredibly lame.
>>>
>>> I think part of what this thread misses is that DSP is a god chunk of 
>>> what SIP phones need. A general purpose CPU is not the right tool for 
>>> the task. A cheap DSP is better suited to compression, transcoding, etc.
>>>
>>> OTOH, presuming that the snom phones are Linux on a suitable platform 
>>> soomeone could develop a custom software load for them and OEM the 
>>> hardware.
>>>   
>> I'm surprised that no-one has mentioned Astfin. Basically uClinux and 
>> asterisk running on an Analog Devices Blackfin DSP. There's also some 
>> 'open source' hardware that's available - the IP04 and friends. I'm 
>> using an Edgepbx FX08, and they also have a two-port version (FX02). 
>> Atcom has a single-port one, the IP01.
>>
>> Though if I were going to prototype an 'open' SIP phone, I'd probably 
>> start with a beagle board (TI OMAP3530 - dual-core ARM+DSP). It's a 
>> pretty powerful SOC - its brother (3430) powers the Palm Pre.
>>
>> Just another datapoint :)
>> 
>
> Yeah, that'd be great hardware to select. 
>
> What I was thinking is that this thread seems to be driven by those of
> a software bent. For that group perhaps there's an opportunity to write
> code for something like a snom 820. It's a solid solid hardware basis
> for the project. Snom would be foolish not to sell it for such use,
> even price it attractively. That way the hardware work would be done,
> and the software geeks could work their magic.
>   
I'm a card-carrying (embedded linux) software geek, and I know I'd be 
interested :)

Anyone got some influencial contacts inside Snom? or Aastra, for that 
matter, their hardware also seems good quality from what people have said.

Another possibility is talking to Atcom (or other VoIP ODMs), they seem 
to have done pretty well from the IP04 and derivatives. They've 
experienced the benefits of an open development model, perhaps they'd be 
interested. Not sure what the quality of their existing handset hardware 
is like.

Anyone on the list have the contacts to get the ball rolling?

Paul

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Re: [asterisk-users] Snom - we are puzzled

[Christian Stredicke]

I would get a PCAP trace from the phone to see what is going on "on the cable".

CS 

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Ronald 
Wiplinger (Lists)
Gesendet: Dienstag, 28. Oktober 2008 23:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Snom - we are puzzled

we have installed asterisk and snom with PUBLIC IPs (IP/25) on one DSL line we 
have for our office a different ADSL with one IP shared.

Two identical setup snom 360 (except the user name) with two public IP 
addresses are connected at the hub to the server / DSL line

phone A can call B, B cannot call A, because A is not registered!!!

We disconnect A and setup a softphone (on the ADSL line with stun) and it works.

How can I track down this problem.

bye

R.

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Re: [asterisk-users] Sporadic One Way Audio

[Christian Stredicke]

We have seen cases where an IP address conflict caused something like this.

You can take Wireshark traces on the PC (possibly run them in a loop so that 
you have a pretty long context) and if you have one-way audio be quick to log 
on to the web interface of the phone and also take a wireshark (PCAP) trace.

There are a couple of tools available that may help to track such problems 
down: http://manageengine.adventnet.com/products/vqmanager, 
http://palladion.net, www.networkinstruments.de, and www.voipfuture.com. I know 
some of them offer a 14-days demo, and it tremendeously helped on of our 
clients to fix network problems. You can also use SNMP tools to poll if the 
phone has any blackouts regaring network availbility (see 
http://wiki.snom.com/SNMP).

Also the phone sends a statistics at the end of each call. Check the BYE 
message, there is a counter of received and transmitted packets. Those numbers 
should be roughtly the same.

CS 

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Brent Davidson
Gesendet: Freitag, 24. Oktober 2008 18:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Sporadic One Way Audio

I'm having an unusual problem at one of my branch offices.  Every now and then 
they will make a call and the person they call is unable to hear them, but they 
are able to hear the person.  The Asterisk server has only one ethernet 
interface and is on the same physical network as the 2 snom 300 phones and is 
connected to the PSTN lines with a  Rhino R4FXO-EC card.  Usually hanging up 
and calling back solves the problem, but it is still aggravating to the 
customer that has been called.  
Normally I'd suspect that something was only passing packets in one direction, 
but there is no firewall between the asterisk server and the phones and no 
iptables or anything like that running on the Asterisk server and sifting 
through sip debug logs to try to find one call out of maybe 50 has so far 
proven fruitless.

Are there any common issues that might cause this?

Thanks,
Brent Davidson



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Re: [asterisk-users] Snom phones and P-asserted

[Christian Stredicke]

snom supports from-change (RFC 4916), maybe that helps.

CS

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Loic Didelot
Gesendet: Dienstag, 16. September 2008 16:31
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Snom phones and P-asserted

Hello,
does anyone know if Snom supports the P-asserted header for pickups and 
transferts...

I tried the following branch and it works for Linksys phones
http://bugs.digium.com/view.php?id=8824


Maybe someone on this list has a better contact to Snom and can push things or 
at least provide deadlines.

Best regards,
Loic Didelot.


--
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
[EMAIL PROTECTED]
http://www.mixvoip.com


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Re: [asterisk-users] BLF and Snom phones

[Christian Stredicke]

This just adds a new drop down called "BLF" in the function key area. This is a 
mix of the already existing "Extension" (which displays dialog-state 
information) and "Speed Dial". The LED is controlled by the dialog state like 
in the Extension mode, while the key is controlled by the Speed Dial nature. 

In other words, no matter what the LED is doing, pushing that button will 
always dial the speed dial number. If you set it to something like "*7123", 
then this button will always pick up the call for 123.

CS 

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Steve Davies
Gesendet: Mittwoch, 26. März 2008 14:01
An: Asterisk Users Mailing List - Non-Commercial Discussion; Christian Stredicke
Betreff: Re: [asterisk-users] BLF and Snom phones

Hi,

Could you explain for the benefit of the list what you have changed in the snom 
image that will benefit this ticket? I am already receiving your current beta 
images, through our distributor, up-to about 2008-13-19, and am not aware of 
any changes that affect BLF behaviour or short-dials...

NOTE to list: User beware - The last few versions of beta firmware I tested 
break re-invites. Of course this may be fixed by now.

Regards,
Steve

On 26/03/2008, Christian Stredicke <[EMAIL PROTECTED]> wrote:
>
>
> Anyone who is willing to try out an image please send me a private email.
>
> CS
>
>  
>  Von: Christian Stredicke
> Gesendet: Sonntag, 23. März 2008 11:56
> An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Betreff: AW: [asterisk-users] BLF and Snom phones
>
>
>
> I agree with Russel that vendor specific things should be the 
> exception. The RFC was not written for features like call pickup, and 
> the way snom interpreted it years ago (even my snom 100 already 
> supported dialog state!) was just because we wanted to avoid 
> additional provisioning. If there should be something in the snom 
> phones that needs to be done, then we can take a look into this. Looking at 
> the ticket, it seems to be simple.
>
> CS [from snom]
>
>  
>  Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Im Auftrag von Rob 
> Hillis
> Gesendet: Sonntag, 23. März 2008 11:01
> An: Asterisk Users Mailing List - Non-Commercial Discussion
> Betreff: Re: [asterisk-users] BLF and Snom phones
>
>
> Bill Hackensack wrote:
>
>
>
> On Sat, Mar 22, 2008 at 7:17 AM, Philipp Kempgen 
> <[EMAIL PROTECTED]>
> wrote:
>
> >
> >
> >  http://bugs.digium.com/view.php?id=5014
> >
> >
>
> The response on that issue from Russell is the kind of response that 
> really ticks me off.  No, no, no, we don't want any real features that 
> users want, we want basic, boring features.  Asterisk is a call center 
> system, not for regular, everyday business users.
>
> It could be so much more, though...
>
> Works great as an advanced IVR as a front end to a real phone system, 
> though.
> While his basic point makes sense (we want to get away from channel 
> specific implementation stuff) what he seems to be ignoring is that 
> this patch actually provides no benefit at all to non-SIP channels, 
> since the Snom phones don't support any technology other than SIP.
>
> I still think Asterisk is more than just a front-end to a "rea"l phone 
> system.  What you can achieve with Asterisk is vastly beyond anything 
> you can achieve with most other PABX systems without spending an utter 
> fortune.
>

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Re: [asterisk-users] BLF and Snom phones

[Christian Stredicke]

Anyone who is willing to try out an image please send me a private email.
 
CS



Von: Christian Stredicke 
Gesendet: Sonntag, 23. März 2008 11:56
An: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Betreff: AW: [asterisk-users] BLF and Snom phones


I agree with Russel that vendor specific things should be the exception. The 
RFC was not written for features like call pickup, and the way snom interpreted 
it years ago (even my snom 100 already supported dialog state!) was just 
because we wanted to avoid additional provisioning. If there should be 
something in the snom phones that needs to be done, then we can take a look 
into this. Looking at the ticket, it seems to be simple.  
 
CS [from snom]



Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Rob Hillis
Gesendet: Sonntag, 23. März 2008 11:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] BLF and Snom phones


Bill Hackensack wrote: 



On Sat, Mar 22, 2008 at 7:17 AM, Philipp Kempgen <[EMAIL PROTECTED]> 
wrote:



 
http://bugs.digium.com/view.php?id=5014



The response on that issue from Russell is the kind of response that 
really ticks me off.  No, no, no, we don't want any real features that users 
want, we want basic, boring features.  Asterisk is a call center system, not 
for regular, everyday business users.
 
It could be so much more, though...
 
Works great as an advanced IVR as a front end to a real phone system, 
though.


While his basic point makes sense (we want to get away from channel specific 
implementation stuff) what he seems to be ignoring is that this patch actually 
provides no benefit at all to non-SIP channels, since the Snom phones don't 
support any technology other than SIP.

I still think Asterisk is more than just a front-end to a "rea"l phone system.  
What you can achieve with Asterisk is vastly beyond anything you can achieve 
with most other PABX systems without spending an utter fortune.


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Re: [asterisk-users] BLF and Snom phones

[Christian Stredicke]

I agree with Russel that vendor specific things should be the exception. The 
RFC was not written for features like call pickup, and the way snom interpreted 
it years ago (even my snom 100 already supported dialog state!) was just 
because we wanted to avoid additional provisioning. If there should be 
something in the snom phones that needs to be done, then we can take a look 
into this. Looking at the ticket, it seems to be simple.  
 
CS [from snom]



Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Rob Hillis
Gesendet: Sonntag, 23. März 2008 11:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] BLF and Snom phones


Bill Hackensack wrote: 



On Sat, Mar 22, 2008 at 7:17 AM, Philipp Kempgen <[EMAIL PROTECTED]> 
wrote:



 
http://bugs.digium.com/view.php?id=5014



The response on that issue from Russell is the kind of response that 
really ticks me off.  No, no, no, we don't want any real features that users 
want, we want basic, boring features.  Asterisk is a call center system, not 
for regular, everyday business users.
 
It could be so much more, though...
 
Works great as an advanced IVR as a front end to a real phone system, 
though.


While his basic point makes sense (we want to get away from channel specific 
implementation stuff) what he seems to be ignoring is that this patch actually 
provides no benefit at all to non-SIP channels, since the Snom phones don't 
support any technology other than SIP.

I still think Asterisk is more than just a front-end to a "rea"l phone system.  
What you can achieve with Asterisk is vastly beyond anything you can achieve 
with most other PABX systems without spending an utter fortune.


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Re: [asterisk-users] Snom 370 buton Recordings

[Christian Stredicke]

BTW I would recommend to move to 7.1.30, this is much better than
7.0.17.

CS



Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Dovid B
Gesendet: Mittwoch, 2. Januar 2008 21:31
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Snom 370 buton Recordings


Have a look here: http://www.voip-info.org/wiki/view/Snom+Phones and at
features.conf

- Original Message - 
From: voip crazy   
To: asterisk-users@lists.digium.com 
Sent: Friday, December 21, 2007 2:13 PM
Subject: [asterisk-users] Snom 370 buton Recordings

Hello all,

I am using the Snom 370 phone  with firmware Snom370-SIP 7.0.17
connected to an asterisk 1.2.14 and I can't record any calls using the
"Recording" button on this phone. The extension I configured on this
phone has the values "Recording on demand", an the voicemail enabled. I
am using FreePBX to manage my PBX. 

How should I configure the "Function keys" to make this work?
Anybody have made this button works on this phone? How?

Any clue will be welcomed.

Thanks in advance.

Voipcrazy











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Re: [asterisk-users] VPN Client with the IP Phone and what its VPNServer

[Christian Stredicke]

The snom 370 used a OpenVPN client. 

See http://en.wikipedia.org/wiki/OpenVPN and 
http://wiki.snom.com/Networking/VPN (that link contains a slash, but is also 
linked on http://wiki.snom.com/Main_Page). 

CS

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von bilal ghayyad
Gesendet: Dienstag, 11. Dezember 2007 13:16
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] VPN Client with the IP Phone and what its VPNServer

Hi Oliver;

Thanks  alot for your reply.
What the needed VPN server? Cisco, Juniper, Planet ?
Does it use IPSec or PPTP?

Regards
Bilal
-
Snom

2007/12/11, bilal ghayyad <[EMAIL PROTECTED]>:
>
> Hi All;
>
> Is there an IP Phones working with Asterisk that
come
> built in with VPN Client? And what the VPN server it works with it 
> fine?
>
> Regards
> Bilal



  

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Re: [asterisk-users] Registration of Snom 320phonewithAsterisk 1.4.13

[Christian Stredicke]

What you can still to is setting the port on the phone to port 5060 - just as a 
little dirty workaround until there is a better solution available.

CS

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jason White
Gesendet: Montag, 29. Oktober 2007 10:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Registration of Snom 320phonewithAsterisk 1.4.13

On Mon, Oct 29, 2007 at 09:22:21AM +0100, Christian Stredicke wrote:
> Well, the response should go to the port number provided in the Via header.
> If there is a rport set, then to that port. Everything looks good in 
> the log, the only problem is that the response is sent to the wrong port.

I tried inserting
nat=never
into sip.conf but that didn't help.

Is there a configuration option that will fix this? If not, what's the prospect 
of having it corrected for the next release of Asterisk?

I can test a patch if that would help.

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Re: [asterisk-users] Registration of Snom 320 phonewithAsterisk 1.4.13

[Christian Stredicke]

Well, the response should go to the port number provided in the Via header. If 
there is a rport set, then to that port. Everything looks good in the log, the 
only problem is that the response is sent to the wrong port.

The Contact port will be used later when the server wants to send a request 
(not a response) to the phone.

CS

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jason White
Gesendet: Montag, 29. Oktober 2007 09:16
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Registration of Snom 320 phonewithAsterisk 1.4.13

On Mon, Oct 29, 2007 at 08:17:20AM +0100, Christian Stredicke wrote:
> I guess the problem is that * sends the response to port 5060, while 
> the phone listens on port 2xxx for an answer.

That could be the problem.

The phone specifies port 2048 in its "contact" field. Is there a way to 
configure Asterisk to respond on whichever port the phone specifies?

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Re: [asterisk-users] Registration of Snom 320 phone withAsterisk 1.4.13

[Christian Stredicke]

I guess the problem is that * sends the response to port 5060, while the phone 
listens on port 2xxx for an answer.

CS

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Jason White
Gesendet: Montag, 29. Oktober 2007 07:46
An: asterisk-users@lists.digium.com
Betreff: Re: [asterisk-users] Registration of Snom 320 phone withAsterisk 1.4.13

Here are more details:

The phone and the Asterisk box are behind the same router (the Asterisk machine 
is 192.168.0.2 and the phone is 192.168.0.4).

A ping command works:

[EMAIL PROTECTED]:~$ ping -c 10 192.168.0.4
PING 192.168.0.4 (192.168.0.4) 56(84) bytes of data.
64 bytes from 192.168.0.4: icmp_seq=1 ttl=64 time=0.500 ms
64 bytes from 192.168.0.4: icmp_seq=2 ttl=64 time=0.491 ms
64 bytes from 192.168.0.4: icmp_seq=3 ttl=64 time=0.493 ms
64 bytes from 192.168.0.4: icmp_seq=4 ttl=64 time=0.495 ms
64 bytes from 192.168.0.4: icmp_seq=5 ttl=64 time=0.495 ms
64 bytes from 192.168.0.4: icmp_seq=6 ttl=64 time=0.493 ms
64 bytes from 192.168.0.4: icmp_seq=7 ttl=64 time=0.493 ms
64 bytes from 192.168.0.4: icmp_seq=8 ttl=64 time=0.495 ms
64 bytes from 192.168.0.4: icmp_seq=9 ttl=64 time=0.505 ms
64 bytes from 192.168.0.4: icmp_seq=10 ttl=64 time=0.492 ms

--- 192.168.0.4 ping statistics ---
10 packets transmitted, 10 received, 0% packet loss, time 9005ms rtt 
min/avg/max/mdev = 0.491/0.495/0.505/0.014 ms [EMAIL PROTECTED]:~$

However, the phone never appears to receive the responses from Asterisk to its 
register requests. The error on the phone is:
[2]29/10/2007 17:02:59: Transport Error: Pending packet 1046807: generating fake
[2]29/10/2007 17:02:59: Registrar [EMAIL PROTECTED] timed out

>From /etc/asterisk/sip.conf:

[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to A
realm=192.168.0.2
context = default ;Default for incoming calls [5549] disallow=all allow=ulaw 
allow=alaw allow=gsm type=friend ;(inbound and outbound calls accepted) 
secret=localphone ; obvious password for testing host=dynamic callerid=Jason 
White <5549> dtmfmode=auto
mailbox=5549 ;(Asterisk VM-system's mailbox #)

The output from sip set debug is attached, as captured earlier by the script 
command.

Asterisk version 1.4.13, Debian GNU/Linux Sid (up to date); this phone has 
successfully registered with external Asterisk servers.

Suggestions are much appreciated.



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AW: [asterisk-users] WiFi SIP phones

[Christian Stredicke]

Try a Nokia E61/E62... Version 3 supports SIP and WiFi and they have a big 
battery that allows long talking and standby times.

CS

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Justin Moore
Gesendet: Donnerstag, 24. Mai 2007 10:16
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] WiFi SIP phones

On 5/23/07, Michael Graves <[EMAIL PROTECTED]> wrote:
>  I must say that I've VERY happy with my Aastra 4801 CT phones. I 
> think that they're DECT. Each can have up to six cordless handsets. 
> Technically its a 9 line phone, but if you use G.729 you can only 
> sustain two calls at once. I can have a call on the portable and easily take 
> another on the base.

I am also an extremely happy user of an Aastra 480i CT. Awesome phone.
However, I was under the impression that the OP was looking for a WiFi phone 
that could be carried from place to place, but I may be wrong...

--
Justin Moore
aka wantmoore
---
www.wantmoore.com
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AW: [asterisk-users] Snom 320 echo

[Christian Stredicke]

Most of the cases can easily be solved by setting the handset mic gain
to 2 (out of 1..8). The gain is usually much to high - optimal for
whispering voices. If the other side talks loud the echo of the cable
will be amplified too much.
 
CS



Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Mike
Hammett
Gesendet: Dienstag, 23. Januar 2007 16:16
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Snom 320 echo


Has anyone ever encountered an echo on the IP phone side of a call?  It
is an echo of the user's own voice.  I believe that no one else in the
office is experiencing this problem.  The phone itself is a Snom 320.
I've asked Snom for assistance since my source no longer carries Snom,
but unlike previous times they've been slow to respond.
 
 
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
 
 
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AW: [asterisk-users] Which IP phones have best voice quality, preferably under $150

[Christian Stredicke]

snom 300 :">

CS

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Kristian 
Kielhofner
Gesendet: Mittwoch, 1. November 2006 12:01
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Which IP phones have best voice 
quality,preferably under $150

Zeeshan Zakaria wrote:
> Hi all,
>  
> I have to buy some IP phones. Previously I have used Grandstream 
> GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems 
> with sound quality with all of them, and I was always of the opinion 
> that it were the phones which were not good. In GXP-2000 deployment of 
> about 50 phones, some work good, some have sound problems like words 
> missing, clicking sounds when talking, and some don't work at all 
> (probably defective).
>  
> What good phone are out there which will work perfectly and will not 
> be expensive. Should be $150 or maximum $200.
> 
> --
> Zeeshan A Zakaria
> 

Zeeshan,

Anything from Polycom - IP 301, IP 430.

--
Kristian Kielhofner
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AW: [asterisk-users] NAT issue ? [More Info]

[Christian Stredicke]

Your router might have a problem if there are several 
devices behind NAT with the same port number. Either explicitly set the ports on 
the phone (SIP, RTP, and risk that other ports like DNS, NTP, ... will have 
the same problem) or buy another router that implements NAT/PAT 
properly.
 
CS


Von: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] Im Auftrag von Dovid 
BGesendet: Dienstag, 31. Oktober 2006 22:31An: 
asterisk-users@lists.digium.comBetreff: [asterisk-users] NAT issue ? 
[More Info]

Also when I do sip show peers I get
 
sip show 
peersName/username  
Host    Dyn Nat 
ACL Port Status    
sipmedia/XX   
69.1.236.33 
5060 
Unmonitored10307/10307    
65.8.212.215 D   
N  60414    OK (147 
ms)10305/10305    
(Unspecified)    D   N  
0    UNKNOWN   
10306/10306    
65.8.212.215 D   
N  60414    OK (135 
ms)10320/10320    
(Unspecified)    D   N  
0    UNKNOWN   
10325/10325    
(Unspecified)    D   N  
0    UNKNOWN   
10315/10315    
(Unspecified)    D   N  
0    UNKNOWN   
10310/10310    
69.33.224.23 D   
N  3120 OK (104 ms)8 
sip peers [4 online , 4 offline]
 
307 is the SNOM 300 and 306 is the SNOM 
360
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AW: [asterisk-users] Snom or Cisco Phones?

[Christian Stredicke]

I think one of the differences is: We do pay attention to Asterisk and this 
mailing list ;-)

CS 

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira
Gesendet: Dienstag, 31. Oktober 2006 13:47
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Snom or Cisco Phones?

Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom 
and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need to 
focus more in SIP and Asterisk compatibility and less in pricing (yes, I know 
the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these features 
important?
Thanks

Joao Pereira

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AW: [asterisk-users] PoE IP Phone

[Christian Stredicke]

Here comes the advertisement for snom phones: http://www.snom.com.

CS

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von bilal ghayyad
Gesendet: Donnerstag, 5. Oktober 2006 15:46
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] PoE IP Phone

Hi List;

I am looking to use an good IP Phone working with Asterisk and work based on 
PoE (so it takes the power via the ethernet cable, no need to connect for it 
separated power adaptor).

Can someone advise me for good one?

Regards
Bilal

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RE: [asterisk-users] SNOM 320 - 404 "Not Found"

[Christian Stredicke]

Try "Support broken Registrar" on the phone (line settings) - then the
phone accepts that the line parameter has been ignored.

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of oi geli
> Sent: Friday, September 22, 2006 7:47 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] SNOM 320 - 404 "Not Found" 
> 
> I am trying to get a SNOM 320 working with Asierisk.
> It does register and I can make outbound calls. But it would 
> not take inbound calls. This is what I get;
> 
> -- Executing Dial("Zap/2-1", "SIP/102|20|Tt") in new stack
> -- Called 102
> -- Got SIP response 404 "Not Found" back from
> 192.168.1.105
> -- SIP/102-cf47 is circuit-busy
> 
> 
> Here is the outbound call;
>   == Spawn extension (outgoing, 102, 102) exited non-zero on 
> 'SIP/105-5526'
> -- Executing Dial("SIP/102-fbb7",
> "Zap/g1/9729772921|90") in new stack
> -- Called g1/NX
> -- Zap/2-1 answered SIP/102-fbb7
> 
> Here is the sip.conf
> [102]
> 
> type=friend
> 
> username=102
> 
> secret=102
> 
> host=dynamic
> 
> context=outgoing
> 
> reinvite=no
> callwaiting=yes
> threewaycalling=yes
> 
> canreinvite=no
> 
> qualify=300
> 
> callerid="102" <102>
> 
> mailbox=102
> 
> Here is the sip registration;
> 
> localhost*CLI> sip show peers
> Name/usernameHost Mask
> Port Status
> 
> 102/102  192.168.1.105   (D)  255.255.255.255 
> 5060 OK (41 ms)
> 101/101  192.168.1.100   (D)  255.255.255.255 
> 5060 OK (40 ms)
> 
> I would highly appreciate the help to resolve the problem.
> 
> Thanks
> 
> 
> 
> __
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> around http://mail.yahoo.com 
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RE: [asterisk-users] IP phone with 2 ethernet jacks

[Christian Stredicke]

AFAIK snom does support layer 2 and layer 3 QoS. Is there any other QoS?

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Dovid Bender
> Sent: Friday, August 25, 2006 12:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IP phone with 2 ethernet jacks
> 
> SNOM is a good phone but dosent have QOS. The polycom does :)
> - Original Message -
> From: "Guido Hecken" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Friday, August 25, 2006 4:52 AM
> Subject: RE: [asterisk-users] IP phone with 2 ethernet jacks
> 
> 
> We like the SNOM 360 Phones. They have really good features.
> 
> Guido
> 
> > -Ursprüngliche Nachricht-
> > Von: Mindaugas Kuprys [mailto:[EMAIL PROTECTED]
> > Gesendet: Freitag, 25. August 2006 09:40
> > An: asterisk-users
> > Betreff: [asterisk-users] IP phone with 2 ethernet jacks
> >
> > Hi,
> > Can anyone suggest good quality IP phone with 2 Ethernet 
> jacks. I wanted
> > Sipura but they don't have such product.
> >
> > Thanks,
> > Mindaugas
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> 
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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 missed call.

[Christian Stredicke]

If the * server sends the following header in the CANCEL request, then
then snom phone does not count the call as missed:

Reason: SIP;cause=200;text="Call completed elsewhere"

See http://www.ietf.org/rfc/rfc3326.txt. Maybe someone can post an
example on how to insert this header.

Christian

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Franklin Webb
> Sent: Friday, July 21, 2006 4:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SNOM missed call.
> 
> Hi Thomas,
> 
> That setting is controlled by line.  Maybe you could setup 
> two seperate lines on the phones and direct the two different 
> call types accordingly.
> 
> Franklin Webb
> Assistant IT Project Leader
> Inter Medi@ Marketing Solutions
> 610-701-9670
> [EMAIL PROTECTED]
> 
> - Original Message -
> From: Thomas Laurids Pedersen <[EMAIL PROTECTED]>
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> 
> Sent: Thursday, July 20, 2006 2:19:14 PM GMT-0500
> Subject: [asterisk-users] SNOM missed call.
> 
> Hi All,
> 
> Using AAH 2.8.
> 
> I have configured a group to handle a common number for a 
> remote office.
> All phones in the office is in the group and they are ringing 
> with a seperate ringtone. All this is very well.
> 
> However all phones other than the one how answered the call 
> is recording a missed call. I know this is an option in the 
> SNOM phone, but is there some way to avoid this for this type 
> of calls ? or is there another way of doing this ?
> 
> Best regards
> 
> Thomas Laurids Pedersen
> 
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> 
> 
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RE: [asterisk-users] Snom 300 headset with static noise

[Christian Stredicke]

Title: FW: [asterisk-users] Snom 300 headset with static noise



There is a difference in the biasing circuit for the microphones in the 
headsets. Unfortunately there is no standard on the market. The snom phones 
190/320/360 (let’s say: type A) behave different than snom 300 (type B). So 
there is always the need to have different headsets or different cables (Quick 
Disconnect). Some headsets are just working with one type (those with extra 
amplifier) and other devices seem to work in both environments, but that’s not 
really true. The headsets are always working much better with just one type. So 
if someone has a headset designed for type A, he’ll have a bad quality while 
connecting it to type B phones although he is able to here something. 

 
Don’t forget to have a connection to an earth-signal (e. g. shielded 
Ethernet cable to PC/switch or earth-grounded power supply). 

 
Hope this helps, 
CS

  -Original Message-From: 
  [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] 
  On Behalf Of Koopmann, Jan-PeterSent: Sunday, July 16, 2006 1:31 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [asterisk-users] Snom 300 headset with static noiseOn Freitag, 14. 
  Juli 2006 10:13 Adrià Vidal wrote:> Someone using these phone Snom 
  300 with his own headset ?We used to but the quality was horrifying. 
  Since we changed to Plantronics Noise Cancelling headsets everything is 
  wounderful.> We got horrible static noise on them?Maybe the 
  article Michiel pointed out helps you still the overall voice quality of their 
  headsets (at least the ones they sold last year) is awful.Kind 
  regards,  
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RE: [Asterisk-Users] SNOM Softphone on windows 2000

[Christian Stredicke]

Well we do write to the registry... Sorry about that, but how would we
otherwise store the information that is needed for the phone?!

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Alexander Lopez
> Sent: Thursday, June 29, 2006 4:01 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] SNOM Softphone on windows 2000
> 
> W2K had problems with Security (Surprising huh?) You may need 
> to grant write access for the user to the Folder where SNOM 
> is installed. I don't think SNOM is writing to the registry 
> if so you will need to open permissions up on those keys in the hive.
> 
> 
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RE: *** Spam *** [Asterisk-Users] recommended telephones

[Christian Stredicke]

Check 
out http://www.digium.com/en/ecosystem/partners/interoppartners.php
 
CS

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  RicardoSent: Thursday, June 29, 2006 11:16 AMTo: 
  asterisk-usersSubject: *** Spam *** [Asterisk-Users] recommended 
  telephones
  Hello alli wondered what telephones should you recommend to 
  use with asterisk, sip compatible, that could use as many functions as 
  possible, like any modern digital phone with programable keys. It should have 
  leds that display who is busy at the moment, let transfer calls as simple as 
  possible, display who is calling, allow multiconference...Where can i 
  get that information??Thanks and pardon my bad 
EnglishRicardo.
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RE: [Asterisk-Users] best hardphone for Asterisk?

[Christian Stredicke]

snom 300 :-)
 
CS

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Crazy 
  BoySent: Friday, June 23, 2006 7:16 AMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] best 
  hardphone for Asterisk?
  Dear Friends,We have implemented "Asterisk" in our 
  organization. There are 150 members in our organization. At present all are 
  using softphones. Now, I want to buy hardphones for our staff. Can anybody 
  suggest me that what is the best hardphone for Asterisk with 
  low-cost?Thank 
  you.Regards,Chandra.
  
  
  Ring'em or ping'em. Make PC-to-phone 
  calls as low as 1¢/min with Yahoo! Messenger with 
Voice.
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RE: *** Spam *** [Asterisk-Users] Don't use CDRTool From AG-projescts

[Christian Stredicke]

This post cannot be left without comment. People who don't know you or Adrian 
might get a wrong impression.

I know Adrian quite well and know that he is one of the real experts in this 
industry and he and his stuff does not deserve such a treatment. 

I would recommend that you change your attitude. It seems like you did not get 
what you want (for free) and you complain like a small child. An apology would 
be appropriate.

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: Thursday, June 22, 2006 3:16 PM
> To: Asterisk-users@lists.digium.com
> Cc: [EMAIL PROTECTED]
> Subject: *** Spam *** [Asterisk-Users] Don't use CDRTool From 
> AG-projescts
> 
> hello to all,
> 
> I advice you to not use CDRtool from ag-projects :
> Fisrt ag-projects talk about is product like a gpl software 
> however they don't provide at least some documentation for 
> non commercial users .
> 
> try to call them !!
> i'll offer you some money .
> 
> You can not Call them for some advices ...
> 
> It's really a bad product don't waste your time to setup it. 
> this enterprise must be  fogotten it's ag-projects .
> it's not a reliable society ... more and more 
> 
> projects around open(ser) asterisk and more are offered good 
> unliked projects cdrtool please do not use ag-projects products !
> 
> 
> Harry is not Harry Potter !
> 
> Regards
> 
>   
> 
> 
> 
> 
> 
> 
> 
>   
> 
>   
>   
> __
> _
> Yahoo! Mail réinvente le mail ! Découvrez le nouveau Yahoo! 
> Mail et son interface révolutionnaire.
> http://fr.mail.yahoo.com
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RE: [Asterisk-Users] Snom high SIP ping time

[Christian Stredicke]

If you ping on the SIP port the message has to go through 
the application layer - which takes some time considering it is an embedded 
system with a small CPU. That part should be ok.
 
It the phone becomes choppy, that problem is probably 
related to the RTP side. Maybe you have different packet sizes for incoming and 
outgoing traffic. You can get an Ethereal trace from the web interface of the 
phone which should show you the RTP jitter (PCAP trace). Or use a hub if you 
don't trust that trace. 6.1 is the latest version if you want to try the latest 
image (http://www.snom.com/wiki/index.php/Beta_Firmware). 

 
Hope that helps, CS

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Mike 
  HammettSent: Monday, June 12, 2006 3:01 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] Snom high 
  SIP ping time
  
  I don't know everything that's going on as 
  someone else has been working on the project, but it hasn't really been going 
  anywhere, so I had some questions.
   
  We've got some Snom 320s with Asterisk 1.2.9.1 (I 
  believe).  All was well (with a previous release), but the phones started 
  to get real choppy.  We are also running a softphone at this location and 
  it was fine.  The SIP qualify was returning ping times anywhere from 20 
  to 70 ms over a sparsely used LAN.  Command prompt (ICMP) pings were 
  under 1 ms.  No amount of different Asterisk versions or phone firmware 
  revisions seems to solve this.  All was well, then (as far as we know) 
  without changes, it crapped out.
   
  Any ideas?
   
   
  Mike HammettIntelligent Computing 
  Solutionshttp://www.ics-il.com
   
   
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RE: [Asterisk-Users] Phones that work well through NAT

[Christian Stredicke]

There are two approaches to get NAT working properly:

- Use UDP and send and receive from the same port. This is extremly
simple, however some phones do (by default) send and recieve from a
different ports. Then you have to tell explicity "no no, dont do that;
use the same port". There are even phones that send and receive from
different RTP ports. I would say they are extremly NAT unfriendly. And I
don't know why a phone vendor would do that. Anyway, the IETF specs
allow it. The problem with the UDP approach is the high keep-alive
traffic (every 15-20 secs you must refresh it) and the number of buggy
NAT implementations out there. I would say this approach works with 95 %
of the equipment.

- Use TCP/TLS and keep the TCP connection to the PBX open all the time.
This reduces and amound of keep-alive traffic and works with almost
anything on the market. Because a router that does not support https or
MS Exchange traffic will have a real hard time in the market place! TLS
has the advantage that "smart" routers cannot see the SIP traffic any
more and mess around with it. For example, there is a vendor out there
that does not understand the rport parameter in the Via and removes it
(but leave the ; standing there)!!! Especially when there are relatively
few user agents registered to the system (number of file descriptors),
this approach is superior. AFAIK the next * version will support this
approach; there are already systems available that support TCP and TLS.

Just my two cents.


Christian

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Andrew Kohlsmith
> Sent: Sunday, April 16, 2006 11:16 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Phones that work well through NAT
> 
> On Saturday 15 April 2006 22:37, C F wrote:
> > That is until you run into problems, while they do work, I wouldn't 
> > say that Polycoms work EXEPTIONALLY well, Cisco, and SPA 
> work *MUCH* 
> > better.
> 
> Can you detail some problems?  Just about any off-the-shelf 
> router seems to work with these.  There may be some cheap-ass 
> broken routers you can get for
> $5 which will not work, but all of the brand-name stuff I've 
> tried Just Works, which is why I say they work exceptionally well.
> 
> -A.
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RE: [Asterisk-Users] Receptionist Phones

[Christian Stredicke]

Well the problem with the sidecar is simple. Just try to light all
lights three times within one second. If you have 50 keys there is
already hell breaking loose. If you cascade side cars and say have 100
LED, this is a real Xmas tree. The CPU drowns in XML notifications. We
already had trouble, and we don't want to double it at this time. Good
work, IETF. 

BTW this is not only a problem if the phone. If the PBX has to supply 50
phones with 50 LED and e.g. they are going off hook at the same time, we
are talking about a burst of 50 * 50 = 2500 messages which will have
some impact of the PBX CPU as well. 

We need to do something about this first before we can start having 100
or 150 LED on a device.

Christian - yes I am from snom. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> mustardman29
> Sent: Tuesday, March 28, 2006 8:47 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Receptionist Phones
> 
> So how did the Polycom with sidecars work?  I like the idea 
> of a dedicated FOP display but not sure why you would need it 
> if you have a Polycom with sidecars.
> 
> > -Original Message-
> > From: Jerry Jones [mailto:[EMAIL PROTECTED]
> > Sent: Tuesday, March 28, 2006 7:28 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Receptionist Phones
> > 
> > We installed a snom with 3 sidecars. Kinda worked, but had so many 
> > quirks they had us replace with a Polycom. All their other 
> phones were 
> > of the poly variety. We installed a dedicated lcd running FOP for 
> > display. Receptionist was much happier.
> > 
> > One of the key problems was she like to set the handset on 
> her desk.  
> > But then the snom would not ring.
> > 
> > On Mar 28, 2006, at 9:01 AM, Bob McDowell wrote:
> > 
> > >
> > > Can you chain these to get more that 42 buttons?  I need 
> about 60...
> > >
> > >
> > > Bob McDowell
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf
> > Of Darrell
> > > Long
> > > Sent: Monday, March 27, 2006 4:32 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [Asterisk-Users] Receptionist Phones
> > >
> > > The 360 has an expansion unit. It adds 42 extensions.
> > >
> > > Darrell S. Long
> > > BestWeb Corporation
> > >
> > >   
> > >
> > >
> > >
> > > Daniel Hazelbaker wrote:
> > >> Hmm, which phone from Snom are you using for this? I've
> > looked around
> > >> their website and I can only find 3 VoIP phones, the 
> 300, 320 and 
> > >> 360.
> > >> The 360 by the looks of it only has 12 buttons you can assign to 
> > >> different extensions; am I missing something or is that
> > the phone and
> > >> you just do 12 per phone?
> > >>
> > >> Daniel
> > >>
> > >> On Mar 27, 2006, at 2:28 PM, <[EMAIL PROTECTED]> 
> > >> <[EMAIL PROTECTED]> wrote:
> > >>
> > >>> Yes - set up about 10 of them at a business last year.
> > >>>
> > >>> Monitoring is fine - picking up calls is a bit iffy at
> > the best of
> > >>> times.
> > >>> (that is, picking up a ringing call by pushing the
> > extension button.
> > >>> *8 works fine)
> > >>>
> > >>> Paul Hales
> > >>> Technical Manager
> > >>> AsteriskIT
> > >>
> > >> ___
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> > >> Asterisk-Users mailing list
> > >> To UNSUBSCRIBE or update options visit:
> > >> http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>
> > >
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> > >
> > >
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> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> > 
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RE: [Asterisk-Users] snom 320 MWI light

[Christian Stredicke]

Someone urged us to implement this behavior. I guess there was a large
company that told us that they were not able to send another MWI that
indicates that the messages were deleted... So far people could live
with this smart idea (it was not our idea).

CS (yes I am from snom)

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Nabeel Jafferali
> Sent: Friday, March 03, 2006 9:01 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] snom 320 MWI light
> 
> > I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the 
> > sip.conf entry, I have [EMAIL PROTECTED] and vmexten=*98.
> > 
> > The light on the snom 320 turns on when I have voicemail and the 
> > retrieve button dials the correct extensions.
> > 
> > However, the light turns off immediately after making the call to 
> > voicemail, even if I do not check the voicemail.
> 
> FYI Received the following from a vendor:
> 
> Currently there is not a way to keep the MWI light to stay on 
> after hitting retrieve button on the Snom.  The best option 
> at this point is to set
> checkmwi=1 in the general section of your sip.conf file.  
> This will cause the light to turn back on shortly if there 
> are un-checked messages waiting.
> 
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RE: [Asterisk-Users] What business IP phone to use

[Christian Stredicke]

The PCB has PoE "prepared" - if you open it you will 
see that there is a lot of space where you can solder all kinds of resistors and 
capacitors. Thats for PoE. However we decided that we don't place the necessary 
components because it would increase the price to the end customer by 25 USD - 
which would take us into a different pricing region. But apart from that we put 
everything else from the snom 320/360 there. And IMHO the audio quality is 
nothing less than the "high end" models, the handsfree mode probably even better 
(we avoided some mistakes we made in the other models). Even the 3-way 
conference is supported. 
 
Low use?! I would say at least 80 % of phone users today 
are "low use". A phone with great audio and mandatory (but not sexy) 
features like security for a mainstream price was missing for those users. 

 

And yes, I am from snom... (see my address!). Please excuse 
my excitement. 
 
 
Christian

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Joe 
  PukepailSent: Wednesday, February 22, 2006 8:31 AMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
  [Asterisk-Users] What business IP phone to use
  I like the specs on this, the only thing that it seems to be 
  missing is POE.  Anyone know if POE is going to be supported on the 
  300?  Looks nice and I could see it for low use areas, but would suck for 
  wall mounting if it can't do POE. 
  On 2/22/06, Cory 
  Andrews <[EMAIL PROTECTED]> wrote: 
  
Clint - Looks like your wish has been granted, 
and your love affair with Snom can continue.  They are soon releasing 
the new Snom 300, which has most of the features your are fond of in the 360 
and 320 models, and should be quite near, if not at, your $100 price point. 

 
Read up on it here -> http://www.snom.com/pressinformation_details.html?&tx_ttnews[tt_news]=354&tx_ttnews[backPid]=33&cHash=1bb97caf5c&L=1
 
Detailed specs here -> http://www.snom.com/snom300_voip_phone.html?&L=1 

 
Cory J AndrewsVOIPSupply.com454 Sonwil 
DriveBuffalo, NY 14225++voice - 716.630.1555 
X22email - [EMAIL PROTECTED]AIM - B2CORY

  - Original Message - 
  From: Clint Sharp 

  To: Asterisk Users 
  Mailing List - Non-Commercial Discussion 
  
  Sent: Wednesday, February 22, 2006 
  1:03 AM
  Subject: Re: [Asterisk-Users] What 
  business IP phone to use
   It's funny this thread has been coming up, because 
  I've been testing out phones at my office, and I just did a fairly 
  intensive quality test on them.1) Budgetones: Don't bother for a 
  business setting.  The speaker phone is basically useless (echo 
  problems) and the handset is horrible.  If you follow the suggestion 
  on the Wiki to drill out the handset, it improves things marginally, but 
  not much.  Users talking to you will constantly complain about you 
  sound muffled.  It's think it's a frequency response thing and not a 
  volume thing, I think it's just getting lower than a standard 8 khz sample 
  out of the microphone, because it's so cheap. 2) GXP-2000: Not 
  much better than the Budgetones, but at least the firmware is still in 
  active development.  Feature-wise it's pretty cool, but poor firmware 
  and poor handset hardware again make this a real problem for us.  We 
  lost one handset to static electricity yesterday (which was fixed by 
  adding in a microphone from an old business set, which actually improved 
  that phone's quality).  The speakerphone is useless due to echo 
  issues.  However, 4 line appearances is pretty cool for that price of 
  phone, and passthrough Ethernet at 100 mbs is pretty cool too.  
  Overall, I can't recommend them, because while they sound slightly better 
  than the budgetones, I still get many complaints about muffled calls. 
  3) Polycom: Of the 4 phone brands we're actively using (not 
  including the Wifi phone which rarely gets used), this was the best until 
  I got the Snom in today.  The handset is of good quality.  I 
  have an IP 301, but if the cheapest phone is this good, I'd definitely get 
  a 501 or 601 (and am considering ordering some, although I may order Snom 
  320s instead).  Their support policies do get on my nerves, I'd like 
  to not have to worry about what reseller I'm using, but it's a solid phone 
  with solid features, although the menus are cumbersome and I haven't 
  gotten MWI to work on it yet. 4) Snom 320: This is an excellent 
  phone based off one days testing.  Minimal configuration, 
  professional looking web interface, and the best sound quality of any of 
  the phones I tested.  THe speakerphone works great, and the handset 
  quality is outstanding, and tested the best with my callers that were 
  listening to me through the PST

RE: [Asterisk-Users] Asterisk and Snom 360

[Christian Stredicke]

Still beta, but we could not make it crash any more...: We would be
happy about the feedback from volunteers:-)

http://fox.snom.com/download/snom320-5.3.6a-beta-SIP-j.bin
http://fox.snom.com/download/snom320-5.3.6b-beta-SIP-j.bin
http://fox.snom.com/download/snom360-5.3.6a-beta-SIP-j.bin
http://fox.snom.com/download/snom360-5.3.6b-beta-SIP-j.bin

Release 5.3.6:
o LID: made sure audio channels are off in idle mode under all scenarios

Release 5.3.5:
o GUI: added cwi ringer indication
o GUI: fixed unnecessary dialog state switches on shared line offhook 
o GUI: status led for missed calls 
o SIP: RAck in PRACK was buggy 
o SIP: added call pickup for shared lines

Release 5.3.4:
o SIP: added +sip.rendering parameter for BLA hold/resume NOTIFYs 
o SIP: NOTIFYs with subscription-state: terminated remove the
subscription 

~~~ Christian

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Michael J. Liberatore
> Sent: Sunday, February 19, 2006 6:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Asterisk and Snom 360
> 
>  Ok here it is, just remember who hooked you up :)
> But I don't see anything about fixing a crashing problem that you
> described in 5.3
> I am running this on 1 phone and 5.3 on 3 others, the ones 
> with 5.3 seem
> perfect, the one with 5.3.3 actually locked up once doing a transfer.
> 
> Release 5.3.3:
> o GUI: fixed DND
> o GUI: fixed bug in displaying old voice mail messages
> o SIP: display local LED status for shared lines
> o WEB: "+" in settings value isn't anymore replaced by its 
> hex value on 
> settings dump web interface page
> o WEB: further enhanced french translation
> o SRTP: fixed bug with auto-answer
> o GUI: setting_server can be set manually via GUI menu (snom360)
> o GUI: ringer device should not switch to speaker if headset 
> is enabled
> o GUI: dkeys (e.g. Redial, Retrieve) are working in edit 
> number state, 
> too
> o SETTINGS: if setting_server is IP:port only, make a valid 
> URL out of 
> it
> o SIP: display local LED status for shared lines
> o GUI: Shared Lines can be mapped to LEDs
> o LID: random number generated from random audio data
> 
> 
> http://fox.snom.com/download/snom360-5.3.3b-SIP-j.bin
> 
> 
> -Mike
> Mike240se
> 
> 
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Olivier
> Krief
> Sent: Friday, February 17, 2006 1:20 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk and Snom 360
> 
> Indeed
> - Original Message -
> From: <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Wednesday, February 15, 2006 11:37 PM
> Subject: Re: [Asterisk-Users] Asterisk and Snom 360
> 
> 
> > On Wed, 15 Feb 2006, Olivier Krief wrote:
> >> Do not use 5.3 but 5.3.3 instead as major crashes occur with 5.3.
> >
> > http://www.snom.com/firmware.html#1641
> >
> > 5.3.3 is not available for public download...
> >
> > -Dan
> > ___
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> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users 
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> recipient(s) named 
> above. This message may include advisory, consultative and/or 
> deliberative material and, as such, would be privileged and 
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> and not a public document. Pursuant to 42 CFR, any 
> information in this 
> e-mail identifying a former, present, or potential client of 
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[Asterisk-Users] New version of snom soft phone

[Christian Stredicke]

Hey we have made a new version of our soft phone which fixes an
important bug in the SRTP SSRC part... It is compatible with our latest
version 5.3 of the hard phones.

http://www.snom.com/download/snom360-5.3.exe

Enjoy, Christian
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RE: [Asterisk-Users] changing displayed call info on snom 360

[Christian Stredicke]

That INFO must be inside the extsting dialog, maybe that was the
problem.

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Phil Blundell
> Sent: Monday, January 30, 2006 10:16 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] changing displayed call info on snom 360
> 
> Several of my SIP users are in the habit of diverting all 
> their calls to an assistant when they're out of the office.  
> When these calls ring on the assistant's phone, she wants to 
> be able to tell which number they've been forwarded from so 
> that she can say "Joe Blow's phone" or whatever when she 
> picks up the call.  The assistant's phone is a snom 360, 
> which normally just displays the number of the calling party 
> while it's ringing.
> 
> Snom's FAQ page at http://www.snom.com/wiki/index.php/FAQs 
> suggests that I can send a SIP INFO message to the phone to 
> change the displayed call information.  I did a few 
> experiments with a hacked chan_sip.c, but wasn't able to 
> produce any visible effect on the phone.
> 
> Does anybody have any experience making this snom feature 
> work with Asterisk, or know of any other way to influence the 
> information that's displayed on the phone?
> 
> Thanks
> Phil
> 
> 
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RE: [Asterisk-Users] english snom support forums ?

[Christian Stredicke]

What about starting such a thing on groups.yahoo.com?

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: Saturday, January 28, 2006 6:32 PM
> To: Asterisk-Users@lists.digium.com
> Subject: [Asterisk-Users] english snom support forums ?
> 
> Is there a forum for snom support in english?
> 
> There are some very active snom forums but they appear to be 
> entirely german language only.
> 
> -Dan
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RE: [Asterisk-Users] Announcement: Snom 360 with integrated XML Objects

[Christian Stredicke]

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?

[Christian Stredicke]

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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> > > Asterisk-Users mailing list
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> > > 
> > > 
> > > 
> > 
> > 
> 
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RE: [Asterisk-Users] OT:Snom 360 prompt for registration pwd?

[Christian Stredicke]

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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RE: [Asterisk-Users] Asterisk in SPA9000?

[Christian Stredicke]

snom uses snom!

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Michael Welter
> Sent: Friday, January 20, 2006 7:01 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk in SPA9000?
> 
> Andres wrote:
> > Did Linksys really use Asterisk for the SPA9000 software?
> > 
> 
> I certainly hope so.  Have you checked what SNOM uses for 
> their phones?
> 
> --
> Michael Welter
> Telecom Matters Corp.
> Denver, Colorado US
> +1.303.414.4980
> [EMAIL PROTECTED]
> www.TelecomMatters.net
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RE: [Asterisk-Users] GXP-2000 any good with * ?

[Christian Stredicke]

My understanding is that there is currently a shortage of phones at voipsupply 
(and also in other places). The 320 is selling pretty good :-) and we are 
making the biggest production run *ever* this month!

snom does not discontinue the 320! 

Christian

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Ross C
> Sent: Monday, January 02, 2006 12:46 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] GXP-2000 any good with * ?
> 
> http://www.voipsupply.com/product_info.php?cPath=95_114&produc
> ts_id=883
> am I misreading something?
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Sven Fischer
> (support)
> Sent: Monday, January 02, 2006 11:11 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] GXP-2000 any good with * ?
> 
> 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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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> --
> -
> See our FAQs at: http://www.snom.com/faq0.html?&L=1
> Whitepapers at:  http://www.snom.com/white_papers.html
> --
> -
> snom technology AG   Gradestraße 46 D-12347 Berlin
> Sven Fischer fax +49 30 39833111
> mailto:[EMAIL PROTECTED]   http://www.snom.com
> --
> -
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RE: [Asterisk-Users] SNOM 360 locked up SOLVED

[Christian Stredicke]

He had run into a deadlock situation where he entered an (illegal)
string for the dial plan that made the phone lock up right after reboot.
That bug was fixed in one of the early 4.x versions. The way out was a
little trick with the web browser.

Generally I think if people have a problem today they should move to
4.5. This version seems to be pretty stable, we did not get any
crash-complains or major problem reports from this version. 

For those who want to move on (feature-wise), it is time to jump on the
5.x train - the 5.0 version has been released a few days ago. We tried
our best to test this version as good as possible (including an
Asterisk-lab test), but from experience we know that new features always
take a certain time to stabilise. Therefore, I would today move to 5.0
only if it has a feature that the 4.5 does not have.

We tried to keep the release notes as informative as possible to make
this decision as easy as possible for you.

Happy New Year!


Christian

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Michael George
> Sent: Thursday, December 29, 2005 10:17 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] SNOM 360 locked up SOLVED
> 
> On Thu, Dec 22, 2005 at 03:58:07PM -0800, Steven Ringwald wrote:
> > Thank you so much for your help, Christian! Your suggestion worked 
> > perfectly, and the phones came back up without a problem.
> 
> What part of his suggestion?  Upgrading the firmware to 4.5 
> via the tftp server?
> 
> Please elaborate for the benefit of others who may run into 
> this problem.
> 
> Thank you.
> 
> --
> -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 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] Please recommend a phone

[Christian Stredicke]

Take a look at snom.com...

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jesse Keating
> Sent: Wednesday, October 19, 2005 5:31 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] Please recommend a phone
> 
> On Wed, 2005-10-19 at 16:39 -0400, Jesus Mogollon wrote:
> > 
> >I'm in need of a phone that would blink a led to let the callee 
> > know that there is an incoming call. The GXP-2000 does this 
> but I want 
> > an alternative to Grandstream. Any help is appreciated.
> 
> Polycom IP301s and 501s have a red LED that blinks when calls 
> are coming in.
> 
> --
> Jesse Keating
> GameHouse -- Systems Engineer
> 
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RE: [Asterisk-Users] OT: SIPSAK usage

[Christian Stredicke]

snom phones by default do not accept SIP messages from other
destinations that the registrar (in this case they send a error
response) and they dont listen on port 5060 by default. Reason:
SECURITY!!!

If you want to lower your security, you can manually specify the SIP
port to 5060 and manually disable the filering from the proxy/registrar.
But then dont complain if people make a fun out of themselves by making
your phone ring with funny SIPSAK requests!!!

I think the best practice on this is to send the requests to the proxy
which then will forward the packets depending on the proxy's security
policy. Replace proxy with Asterisk!

Christian

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Colin Anderson
> Sent: Friday, September 30, 2005 4:31 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] OT: SIPSAK usage 
> 
> I'm using sipsak to send messages to Snoms in my subnet. At 
> work, works
> fine:
> 
> sipsak -M -O desktop -B "foo" -s sip:[EMAIL PROTECTED] -H 
> 192.168.1.46
> 
> displays "foo" on the Snom display
> 
> On my home LAN (AAH 1.5, Snom 190 3.60s, switched 100, no 
> VLAN, no routing) the same command (modified for my LAN) 
> always yields:
> 
> (type: 3, code: 3): from 192.168.171.8 
> 
> at the console of the sending machine. Same if I use FQDN.  
> Type 3 Code 3 means "ICMP port unreachable" 
> 
> Doing a PCAP from the phone indicates that the Snom gets the 
> message, but nothing shows up in the SIP log. Doing tcpdump 
> on the originating machine yields something like "Reply from 
> 192.168.171.8 > 192.168.171.10 UDP port 5060 unreachable"
> 
> Same phones, same firmware rev, same version of sipsak, no 
> IPTABLES on the originating machine, DNS lookups work, call 
> behavior is normal, other SIP behavior like MWI works fine. I 
> got nuthin here, anyone got a tip?
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RE: [Asterisk-Users] SNOM 190 '486/Busy here' after upgrade to firmware 3.60s

[Christian Stredicke]

Looks like this phone has redirection or DND set. Anything on the
display? If it still a mystery send us the settings of the phone, then
it should become clear.

BTW if you have a snom trouble ticket, you can also go to
http://www.snom.com/onlinesupport.html (scroll down to set up an
account).

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Colin Anderson
> Sent: Friday, September 23, 2005 1:31 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] SNOM 190 '486/Busy here' after 
> upgrade to firmware 3.60s
> 
> I have a Snom 190 that refuses to accept calls after upgrade 
> to firmware 3.60s, latest. I get "SIP 486/Busy Here". No 
> change in the dialplan, nor settings in the phone. Calls out 
> fine. I did 30 other phones yesterday with 3.60s with no 
> problem, this is the only one. In the phone's log I get:
> 
> [5]22/9/2005 17:01:00: timeout::callback: Registering with 
> timeout of 0 ms
> [5]22/9/2005 17:05:45: Match challenge for user=9002, realm=asterisk
> [2]22/9/2005 17:07:31: Denying call id=-4 
> reason=unconditional <--wtf does this mean?
> [5]22/9/2005 17:07:31: Dialog -4/2 going to terminated
> [5]22/9/2005 17:07:31: timeout::callback: Registering with 
> timeout of 0 ms
> 
> Any Snom factory guys on the list care to comment?
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RE: [Asterisk-Users] Snom-320 badly garbled audio

[Christian Stredicke]

You can always take a PCAP (Ethereal) trace from the phone's web page
and analyze it with the RTP Statistics tool in Ethereal. That should
give you a hint whats up with jitter & Co.

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Darren Ellis
> Sent: Tuesday, September 20, 2005 10:01 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Snom-320 badly garbled audio
> 
> Hello,
> 
> I just bought a Snom-320 from ATAComm.  I plugged it into my 
> LAN, registered it with *, etc.  All my other SIP gear is 
> Sipura and works fine, both on the LAN and over the Internet.
> The new Snom seems like it can't process the audio from the 
> handset mic.  A steady tone is garbled, even on the LAN.  
> I've contacted ATAComm, Snom and the company representing 
> Snom in the US.  So far, ATAComm hasn't helped at all.  The 
> tech just said "I dunno," and referred me to the US Snom rep 
> co.  Snom has replied, but the time zone differential and 
> language barrier is making the process tedious.
> 
> My * server is running Asterisk 1.0.9, zaptel 1.0.9.1, libpri 1.0.9.
> 
> The frustrating part of this is that the Sipura gear works 
> great.  So, I have a hard time accepting that it's an * or LAN issue.
> 
> Does anyone out there have Snom 320 phones in use?  Are you 
> experiencing garbled audio from the handset?  Audio in works 
> fine.  But nobody can understand what I say back to them.
> 
> I upgraded the Snom-320 to the latest firmware, v4.2, but 
> that did not clear the problem.
> 
> I've retooled so that I'm forcing ulaw, as I found that some 
> folks have had bad luck with GSM.  But I've tried both.
> 
> Thanks
> 
> Darren
> 
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RE: [Asterisk-Users] Snom 360 - Message waiting and conference keys

[Christian Stredicke]

You have two choices:

- Set the "Mailbox" setting for the line on the phone

- Send the "Message-Account" in the MWI body (should be done by
Asterisk)

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Paul Brock
> Sent: Wednesday, August 24, 2005 11:01 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Snom 360 - Message waiting and 
> conference keys
> 
> Hi,
> 
> Trying to set up these two buttons on a snom 360. The message 
> waiting key seems to send a call to it's own number, which is 
> obviously engaged and where you are prompted to leave another 
> message to yourself, and the conference key seems to do nothing.
> 
> Anyone manged to overcome these problems so that the 
> conference key actually conferences, and the message waiting 
> "retrieve" key actually retrieves the voice messages?
> 
> If so could you give me some pointers please??
> 
> Thanks
> Paul
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RE: [Asterisk-Users] codec gsm and cisco

[Christian Stredicke]

snom supports GSM.

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Rich Adamson
> Sent: Thursday, August 18, 2005 2:17 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] codec gsm and cisco
> 
> 
> > I would like to know if gsm codec is supported on the 79xx 
> series of 
> > Cisco Voip Phones.
> > 
> > It seems not to be supported, is it right ?
> 
> The Cisco 79X0's do not support gsm.
> 
> > I am looking for phones supporting that codec (AT 320 on example)
> 
> Don't know.
> 
> 
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RE: [Asterisk-Users] Load Testing

[Christian Stredicke]

Try this:

phone1=192.168.7.251
number1="1+0+1"
curl "http://$phone1/command.htm?key=$number1+ENTER"; >/dev/null
2>/dev/null
sleep 10
curl "http://$phone2/command.htm?key=CANCEL"; >/dev/null 2>/dev/null

Available keys:

#define KEY_CANCEL "CANCEL"
#define KEY_CLEAR "CLEAR"
#define KEY_ENTER "ENTER"
#define KEY_OFFHOOK "OFFHOOK"
#define KEY_ONHOOK "ONHOOK"
#define KEY_RIGHT "RIGHT"
#define KEY_LEFT "LEFT"
#define KEY_FUNCTION "FUNCTION" // below redial
#define KEY_MENU "MENU"
#define KEY_REDIAL "REDIAL"
#define KEY_ORG_F1 "F1"
#define KEY_ORG_F2 "F2"
#define KEY_ORG_F3 "F3"
#define KEY_ORG_F4 "F4"
#define KEY_SPEAKER "SPEAKER"
#define KEY_DISCONN "DISCONNECT"
#define KEY_RECALL "RECALL"
#define KEY_BREAK "BREAK"
#define KEY_TRANSFER "TRANSFER"
#define KEY_CONFERENCE "CONFERENCE"
#define KEY_HELP "HELP"
#define KEY_VOLUME_UP "VOLUME_UP"
#define KEY_VOLUME_DOWN "VOLUME_DOWN"
#define KEY_MUTE "MUTE"
#define KEY_HEADSET "HEADSET"
#define KEY_UP "UP"
#define KEY_DOWN "DOWN"
#define KEY_REC "REC"
#define KEY_RETRIEVE "RETRIEVE"
#define KEY_SETTINGS "SETTINGS"
#define KEY_PHONE_BOOK "PHONE_BOOK"
#define KEY_SNOM "SNOM"
#define KEY_DND "DND"

And yes, it is a good reason to set the password on your phone if you
dont want to use it only for testing! Giving everyone access to the web
server of the phone is not a good idea, not only for snom phones.

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Anton Krall
> Sent: Saturday, August 13, 2005 7:46 PM
> To: [EMAIL PROTECTED]; 'Asterisk Users 
> Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] Load Testing
> 
> Hi Michael.
> 
> Are there any script already made for doing this? Sending 
> calls from one asterisk to the one been tested? Something 
> that would simulate your 1 phone scenario?
> 
>  
> 
> |-Original Message-
> |From: [EMAIL PROTECTED]
> |[mailto:[EMAIL PROTECTED] On Behalf 
> Of The VoIP 
> |Connection
> |Sent: Viernes, 12 de Agosto de 2005 10:42 p.m.
> |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> |Subject: RE: [Asterisk-Users] Load Testing
> |
> |Anton,
> |
> |A great tool for "ghetto" call capacity testing is a single 
> snom phone.
> |There is no limit to how many calls a snom phone can make, 
> just put it 
> |on hold and dial again. So, with a single snom phone and a little 
> |imagination you can test any number of scenarios.  You can 
> approximate 
> |basic SIP capacity by creating an extension that plays the asterisk 
> |test message and dialing it repeatedly until quality starts 
> to degrade 
> |or asterisk gives up.
> |To simulate actual call throughput you really need another
> |(faster) machine to connect to, but you can use the same technique. 
> |
> |You can run "top" on the console while you are doing your 
> tests to see 
> |what resources you are using.  Check your logs when you are 
> done to see 
> |what errors were generated when it came unglued.  CPU is not 
> always the 
> |limiting resource, especially with Digium card interfaces 
> which tend to 
> |be bound by FSB speed, but echo cancellation and codec 
> conversion will 
> |burn a LOT of cycles.
> |
> |Michael Crown
> |Managing Partner
> |www.thevoipconnection.com
> |321.989.6728 ext. 611
> |sip:[EMAIL PROTECTED]
> |
> |> -Original Message-
> |> From: Anton Krall [mailto:[EMAIL PROTECTED]
> |> Sent: Friday, August 12, 2005 9:56 PM
> |> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> |> Subject: [Asterisk-Users] Load Testing
> |> 
> |> Guys.
> |> 
> |> How and which tools to use to load test an asterisk install? 
> |> Say for example, you need to see how many calls can be routed thru 
> |> before losing quality and making the cpu jump to the roof?
> |> 
> |> 
> |> 
> |
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RE: [Asterisk-Users] Snom 360 and firmware 4.0 problem

[Christian Stredicke]

Please take a look at http://www.snom.com/howto40.html. We tried to make
the upgrade procedure as smooth as possible, if you are having problems
please tell us and we will try to make it more simple. For example, if
you have a batch of phones give us an email and we will send you the
files in one go. 

New phones dont need that upgrade procedure. It is only necessary when
you are crossing the 4.0 version border. For example, all 320 already
have the certificate installed already, so for 320 there is no need to
go throught the procedure.

For release notes for 4.0, please check out
http://www.snom.com/snom360_release_notes.html.

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Michael George
> Sent: Friday, August 05, 2005 8:31 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Snom 360 and firmware 4.0 problem
> 
> I have a pair of snom 360s at a customer and they were giving 
> me Low Memory errors.  The distributor suggested updating the 
> firmware.  I did that, to the one just below 4.0 (which 
> wasn't released yet).  One of the phones is still giving the 
> Low Memory error every 3-4 days.  The other one had a broken 
> display that was just RMA'd, so it' hasn't been up long 
> enough to know if the error occurs on that one, too.
> 
> The distributor's latest suggestion was to go to the newest 
> firmware, 4.0.  I did that on the new 360 (from the RMA) and 
> with the same account settings as the one it was replacing, 
> it could not register with *.
> 
> Since I was in a pinch, I "updated" the firmware down to the 
> latest below 4.0 and the phone works just fine.
> 
> Does anyone with more knowledge than I know what might be 
> going on?  Maybe a new default setting in 4.0 that's breaking things?
> 
> Thank you.
> 
> --
> -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 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] IP Phone with Standard Power Ethernet

[Christian Stredicke]

After IEEE finally released 802.3af snom supports all three modes in the
320/360 models:

http://www.snom.com/whitepapers/faq-05-03-16-da.pdf (snom 320 = snom
360).

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> chris gamble
> Sent: Tuesday, July 12, 2005 5:17 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] IP Phone with Standard Power Ethernet
> 
> I am looking at phones for my asterisk system and seem to 
> have a problem.
> The only Power over Ethernet phones I can find that support 
> the IEEE standard are 3com. Cisco uses its own proprietary ( 
> and is expensive to boot ), snom has a different but equally 
> non-IEEE method, and i'm havent found another phone that I'm 
> confident can do the job for our office.
> 
> Whats a good high quality ip phone that uses IEEE power over 
> ethernet -- or is there a problem with IEEE power over ethernet??
> 
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RE: [Asterisk-Users] Pushing new firmware to Snom 190

[Christian Stredicke]

Take a look at http://www.snom.com/white_papers.html,
http://www.snom.com/whitepapers/FAQ-04-03-26-v3_4-sf.pdf and check out
DHCP option 66 and 67.

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Colin Anderson
> Sent: Monday, July 11, 2005 11:46 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Pushing new firmware to Snom 190
> 
> Anyone know how I can push a firmware update to a Snom 190 
> without using DHCP? In the web interface, I specify a path to 
> the Snom firmware, and it works, except I have to physically 
> press OK to get the update to download. I need to do it remotely...
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RE: [Asterisk-Users] secretary function

[Christian Stredicke]

You are looking for consultative Xfer and attempting a blind one. Gotta
put the first call on hold first and then join it with the second (line
to boss) using the Xfer key.

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Christian Hiller
> Sent: Friday, June 03, 2005 6:31 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] secretary function
> 
> Hello,
> 
> we got a SNOM 360 here and this gota TRANSFER button.
> With this i can transfer a call from my phone another one. 
> But when i push this Button and transfer the call to another 
> phone, i get kicked out.
> 
> Now, every secretary first asks the chief if he is available 
> or not - how can i implement this feature
> 
> thx for any ideas !
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RE: [Asterisk-Users] A good SIP receptionist phone

[Christian Stredicke]

Hi Michael,

you mean we should focus more on the usability (GUI) than the protocol
stuff? Maybe we should put the GUI programmer for a couple of days on
the receiptionists place and make sure he will have a lot of stress? :-)

Anyway, also for this usability stuff comments are welcome...

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> The VoIP Connection
> Sent: Tuesday, May 03, 2005 3:17 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] A good SIP receptionist phone
> 
> Christian,
> 
> The current snom scheme is great for most applications and 
> would probably work reasonably well for this user.  If you 
> read the original post, he indicates that he would be happy 
> with a snom if he could make it work, and I think this is the 
> main issue with the snom 220 - getting this setup to work can 
> be a little tricky.  We have found in the past that extension 
> monitoring and multiple registrations don't play well 
> together, which makes it hard to use for a lot of situations. 
>  This may be fixed now, I'm not sure when we last tested this.
> 
> Receptionists who are used to the usual key system "park and 
> page" routine can be trained pretty easily to transfer to 
> extensions if the system is set up right. In my experience, 
> most of these people are not stupid. Managing and routing an 
> endless stream of incoming calls is challenging and stressful 
> even under ideal circumstances. When a system doesn't work 
> the way it should it can be very frustrating.
> 
> I know this logic is kind of inside-out, but if you think of 
> a receptionist as a human auto-attendant/IVR and design a 
> phone that supports this role you will sell a lot of them.  A 
> lot of times the receptionist (i.e. office
> manager) is the decision-maker for phone system purchases.
> 
> Michael Crown
> Managing Partner
> The VoIP Connection
> 321.989.6728 ext. 611
> sip:[EMAIL PROTECTED]
> 
> 
> -Original Message-
> From: Christian Stredicke [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, May 03, 2005 1:53 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: Olle E. Johansson
> Subject: RE: [Asterisk-Users] A good SIP receptionist phone
> 
> We at snom would love to have a good LED integration with 
> Asterisk. The current state seems to be a good start, but can 
> use some improvements.
> What would be the best way to push this? Maybe sit together 
> for a few days and work on the integration (doing some dirty 
> hacks). Who would be the right person to talk to? Olle? 
> 
> CS 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Sean 
> > Kennedy
> > Sent: Monday, May 02, 2005 10:46 PM
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [Asterisk-Users] A good SIP receptionist phone
> > 
> > Adam Goryachev wrote:
> > 
> > >On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote:
> > >
> > >  
> > >
> > >>2) There isn't anything like what you want.  I know, I want
> > the same
> > >>thing.  There is no phone out there that will do this with any 
> > >>protocol that asterisk uses.  This is the one major failing of 
> > >>asterisk ( and voip in general.  I smell an oportunity 
> for a phone 
> > >>manufacture ), and what keeps it out of a lot of places.
> > >>
> > >>
> > >
> > >It's alright, you can come out from under your rock now
> > >
> > >The Polycom IP 600, Cisco 7960, and apparently the SNOM 
> (some model) 
> > >phones can all do what he wants. ie, have multiple lines
> > with blinking
> > >red lights when a call arrives on that line.
> > >
> > >The polycom ip600 and cisco 7960 both have 6 lines available.
> > >
> > >Regards,
> > >Adam
> > >
> > Ok, this is the first I've heard about it.  Will the lights 
> show call 
> > status?  As in, if the call is put on hold on one of those other 
> > extensions, it will flash?  Or go green ( or another color ) when a 
> > call is connected on another extension?
> > 
> > Basically a mimic of the partner ACS systems?
> > 
> > To my knowledge, there is no such thing.  Am I wrong?
> > 
> > Sean
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >

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

[Christian Stredicke]

We at snom would love to have a good LED integration with Asterisk. The
current state seems to be a good start, but can use some improvements.
What would be the best way to push this? Maybe sit together for a few
days and work on the integration (doing some dirty hacks). Who would be
the right person to talk to? Olle? 

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Sean Kennedy
> Sent: Monday, May 02, 2005 10:46 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] A good SIP receptionist phone
> 
> Adam Goryachev wrote:
> 
> >On Sat, 2005-04-30 at 18:02 -0700, Sean Kennedy wrote:
> >
> >  
> >
> >>2) There isn't anything like what you want.  I know, I want 
> the same 
> >>thing.  There is no phone out there that will do this with any 
> >>protocol that asterisk uses.  This is the one major failing of 
> >>asterisk ( and voip in general.  I smell an oportunity for a phone 
> >>manufacture ), and what keeps it out of a lot of places.
> >>
> >>
> >
> >It's alright, you can come out from under your rock now
> >
> >The Polycom IP 600, Cisco 7960, and apparently the SNOM (some model) 
> >phones can all do what he wants. ie, have multiple lines 
> with blinking 
> >red lights when a call arrives on that line.
> >
> >The polycom ip600 and cisco 7960 both have 6 lines available.
> >
> >Regards,
> >Adam
> >
> Ok, this is the first I've heard about it.  Will the lights 
> show call status?  As in, if the call is put on hold on one 
> of those other extensions, it will flash?  Or go green ( or 
> another color ) when a call is connected on another extension?
> 
> Basically a mimic of the partner ACS systems?
> 
> To my knowledge, there is no such thing.  Am I wrong?
> 
> Sean
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RE: [Asterisk-Users] IP Softphone Recommendations

[Christian Stredicke]

Also try the snom soft phone: http://www.snom.com/snom360softphone.html. Sorry, 
Windows only:-( 

But at least its free!

Enjoy, CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Guillermo Salas M
> Sent: Wednesday, April 27, 2005 12:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] IP Softphone Recommendations
> 
> Ing CIP Alejandro Celi Mariátegui wrote:
> 
> >El mar, 26-04-2005 a las 09:42, Guillermo Salas M escribió:
> >
> >  
> >
> >>I´m using X-lite on windows and linux, looks pretty well.
> >>
> >>
> >
> >Do you have the link of the X-Lite Linux version? Not found in the 
> >xlite website.
> >
> >  
> >
> 
> Saludos desde Ecuador. 
> 
> Go to http://support.xten.com and register for an account. 
> Later, send an email to [EMAIL PROTECTED] requesting being a 
> beta tester.
> 
> For testing purposes, you can download the latest version from:
> http://xten.com/apps/xprolinuxbeta/xlite-linux-24.bz2
> 
> To installl:
> 
> run this command on the file donwloaded:
> >  bunzip2 xlite-linux-24.bz2
> 
> The result is a file xlite-linux-24, which is the executable, 
> you simply run it from the command line.
> 
> You may need to do a:
> 
> >  chmod +x xlite-linux-24
> 
> first to make it executable.
> 
> >Regards from PERU...
> >
> >  
> >
> 
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RE: [Asterisk-Users] SIP Softphone for testing with Asterisk

[Christian Stredicke]

Also try the snom 360 soft phone emulation:
http://snom.com/snom360softphone.html. 

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of raymond
> Sent: Thursday, April 07, 2005 6:01 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] SIP Softphone for testing with Asterisk
> 
> Hi all,
> 
> I had just set up my asterisk server.
> 
> Can anybody know that is there any sip softphone for testing 
> with asterisk?
> (I had download some from internet but I think all are 
> preconfig to certain server).
> 
> Cheers.
> 
> Raymond
> 
> 
> 
> 
> 
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RE: [Asterisk-Users] Difference between Snom 190 & Elmeg 290?

[Christian Stredicke]

There is a partnership between Elmeg and snom. We are using their
plastic (in the snom 190/200/220), they are using our hard- and software
(in the Elmeg 290). Elmeg have a long experience in making phones and we
have experience in making hard- and software for VoIP (as long as it can
be in the SIP-based industry). A good partnership!

We call it "snom inside"...

CS 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Torsten Krueger
> Sent: Saturday, March 05, 2005 2:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Difference between Snom 190 & Elmeg 290?
> 
> Hello Remco,
> 
> On Sat, 5 Mar 2005, Remco Barende wrote:
> 
> > Hi list!
> >
> > While looking for the Snom 190 I found another phone, the 
> Elmeg IP 290 
> > (www.elmeg.de).
> >
> > Looking at the pictures & the specs they seem to be very similar 
> > beasts but the firmware is supposedly not interchangeable.
> >
> > Does anyone know the difference between the 2, do they work 
> with Asterisk?
> >
> > The weird thing is that Elmeg has similar phones with the Snom look 
> > but they are ISDN only (no voip) while Snom has several 
> other models 
> > that are IP. Who's cloning who? I don't want to end up with 
> phones for 
> > which firmware support or update will disappear soon while the 
> > 'orginal' will continue to be supported?
> 
> Elmeg has been for a long time a manufacturer of ISDN Phones 
> and small to medium PBXes in germany. Snom uses the chassis 
> of the elmeg phones and puts their own electronics in them. 
> So it seems very likely that the elmeg IP-Phones are in fact 
> Snom phones. I do not wether the firmware can be changed 
> across elmeg and snom, but if there are no artificial 
> barriers in place that prevent this this could be possible.
> 
> Torsten
> >
> > Thx!
> > Remco
> > ___
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> >
> >
> 
> --
> Media Online Internet Services & Marketing GmbH
> Torsten Krueger   [EMAIL PROTECTED]
> fon: 49-231-5575100fax: 49-231-55751098
> Kurze Str. 10  D-44137 Dortmund
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RE: [Asterisk-Users] Anybody using X-Lite Softphone ? tryed toforwarda call to X-Lite....

[Christian Stredicke]

Try the snom soft phone! http://snom.com
CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Dave Chase
> Sent: Saturday, February 26, 2005 12:31 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - 
> Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Anybody using X-Lite Softphone 
> ? tryed toforwarda call to X-Lite
> 
> XLite does not support transfer... You have to buy their XPro
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Mateo Meier
> Sent: Tuesday, February 22, 2005 3:50 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Anybody using X-Lite Softphone ? 
> tryed to forwarda call to X-Lite
> 
> Hey Guys
> 
> Im trying to forward a call from the asterisk mainmenue to my 
> second computer with X-Lite installed..
> 
> What I've done so far is this:
> 
> Installed X-lite @my win PC.. 
> 
> X-Lite configuration: 
> Menu | System Settings | SIP Proxy | default Display Name: 
> mateo01 User Name & Authorization User: mateo01
> Password: 
> Domain/Realm: 192.168.1.**
> SIP Proxy: 192.168.1.**
> 
> 192.168.1.** = IP address of Asterisk 
> 
> and the sip.conf file looks like that:
> 
> [mateo01]
> type=friend
> username=mateo01
> callerid="mateo01" <1234>
> host=dynamic
> secret=
> disallow=all
> allow=gsm
> allow=ulaw
> allow=alaw
> context=sip
> nat=no
> 
> 
> Now, Im unsure what to do ? whats next ? and what do I type 
> in to extensions.conf  instead of the following:
> 
> exten=>2,1,Dial(capi/720:078***)
> 
> Thx for the help
> Mateo
> 
> 
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> 
> 
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RE: [Asterisk-Users] snom soft phone

[Christian Stredicke]

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] snom soft phone

[Christian Stredicke]

Sorry, in the beginning we want to focus on the OS which has the biggest
market share by the numbers. Please excuse this, but porting it to Linux
is not trivial

CS

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Maik Schmitt
> Sent: Sunday, February 06, 2005 11:47 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] snom soft phone
> 
> > It is available at 
> http://snom.com/download/snom360-3.57j.exe (Windows 
> > only). To use the free version, just leave the license code 
> field empty.
> 
> Unfortunately it does not run under wine/crossover 
> office/cedega. Will there be a Linux version or at least a 
> version that works with wine?
> I'd love to test it but I will not install Windows just for 
> testing a phone.
> 
> -- 
> Maik Schmitthttp://graphics.cs.uni-sb.de/VoIP
> 
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[Asterisk-Users] snom soft phone

[Christian Stredicke]

Some of you might already know that we are releasing a new phone, snom
360. To make the phone well-known and stable, we have made a soft phone
version out of it and offer it for trial or private use for free (for
more details, see the license conditions).

There are only few limitations to the phone. First of all, the audio
subsystem will work only work with an acceptable quality if you are
using a headset. We only offer G.711 and GSM in this build (G729A can be
enabled by an additional license). Other features like transfer,
Security and LEDs should work just like in the "real" phone. 

It is available at http://snom.com/download/snom360-3.57j.exe (Windows
only). To use the free version, just leave the license code field empty.
Please write your comments to mailto:[EMAIL PROTECTED] Please treat this
version as a beta version.

We are looking forward to a close integration with Asterisk!


Enjoy! Christian
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RE: [Asterisk-Users] Cost of IP Phones, or Isn't It Just Software?

[Christian Stredicke]

Well if you just take a look at the sand that is needed to make the chips
you even get better prices... 

Sand --> silicon --> chips --> PCB --> phone --> a lot of talking

It's not the material of the phone, it's the payroll of the people who make
the "-->" happen.-)

Never mind my rude simplification, CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Michael Sandee
> Sent: Wednesday, June 16, 2004 8:45 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Cost of IP Phones, or Isn't It Just
> Software?
> 
> >
> >
> >Am I dreaming?
> >
> Yes.
> 
> Community based development is too unreliable.
> 
> Just to refer to ongoing projects... Look at the farfon
> (www.farfon.com), It's an active project in the final stages of
> development.
> It offers the benefits (modular, programming of your own features,
> quality components, low price, corporate backing, designed by skilled
> engineers...) without the negative sides of your plan.
> 
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RE: [Asterisk-Users] RE: RE: snom reporting busy when it shouldn't

[Christian Stredicke]

I think it's obvious that there are two dialogs being set up (take a look at
the call-id and from-tag).  I think on the protocol level the behavior is
ok, although not beautiful. 

But I assume that * should send only one INVITE. Maybe there is a second
registration dangling and * is forking the request under a new call-id.

Christian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of nicolas
> Sent: Tuesday, May 25, 2004 7:49 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] RE: RE: snom reporting busy when it shouldn't
> 
> Christian,
> 
> they send two INVITE.
> Below the sip debug of an dial without doing an answer before.
> (There are 2 Phones (100/200) and a sipgate registrar)
> 
> INVITE from *
> RING from snom
> INVITE from *
> BUSY from snom
> CANCEL from *
> 
> if you want i can send a sip debug from the "call waiting indication"
> matter
> but is like above, without the 2. INVITE:
> 
> INVITE from *
> BUSY from snom
> CANCEL from *
> 
> Hope you can help.
> 
> nicolas
> 
> Reliably Transmitting:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK38407e55
> From: "410137463" ;tag=as6d950cc4
> To: 
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Tue, 25 May 2004 07:49:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 364
> 
> Sip read:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK38407e55
> From: "410137463" ;tag=as6d950cc4
> To: ;tag=l0ggp0vc1z
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> Contact: 
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO
> Allow-Events: talk, hold, refer
> Content-Length: 0
> 
> Reliably Transmitting:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK120fd9bf
> From: "0410137463" ;tag=as2a7e2d4f
> To: 
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Tue, 25 May 2004 07:49:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 364
> 
> Reliably Transmitting:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5832dc08
> From: "410137463" ;tag=as05d21741
> To: 
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Tue, 25 May 2004 07:49:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 364
> 
> Sip read:
> SIP/2.0 486 Busy Here
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK120fd9bf
> From: "0410137463" ;tag=as2a7e2d4f
> To: ;tag=1se6rz4cq8
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> Contact: 
> Content-Length: 0
> 
> Transmitting:
> ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK120fd9bf
> From: "0410137463" ;tag=as2a7e2d4f
> To: ;tag=1se6rz4cq8
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Content-Length: 0
> 
> Sip read:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5832dc08
> From: "410137463" ;tag=as05d21741
> To: ;tag=d2jhjs2gig
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> Contact: 
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO
> Allow-Events: talk, hold, refer
> Content-Length: 0
> 
> Reliably Transmitting:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK1e64da5d
> From: "0410137463" ;tag=as4981b9a8
> To: 
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Tue, 25 May 2004 07:49:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 364
> 
> Sip read:
> SIP/2.0 486 Busy Here
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK1e64da5d
> From: "0410137463" ;tag=as4981b9a8
> To: ;tag=1pnv8t8wys
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> Contact: 
> Content-Length: 0
> 
> Transmitting:CLI>
> ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK1e64da5d
> From: "0410137463" ;tag=as4981b9a8
> To: ;tag=1pnv8t8wys
> Contact: 
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Content-Length: 0
> 
> Sip read:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK38407e55
> From: "410137463" ;tag=as6d950cc4
> To: ;tag=l0ggp0vc1z
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 INVITE
> Contact: 
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO
> Allow-Events: talk, hold, refer
> Content-Length: 0
> 
> Sip read:
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 190.100.200.1:5060;branch=z9hG4bK5832dc08
> From: "410137463" ;tag=as05d21741
> To: ;tag=d2jhjs2gig
> Call-ID: [EMAIL PROTECTED]
> CSeq: 102 

RE: [Asterisk-Users] RE: snom reporting busy when it shouldn't

[Christian Stredicke]

Well, how many INVITE are being sent to the phone? If it receives the second
one, it will report a busy.

This sometimes happens when a user agents has a dangling registration and
the SIP proxy does call forking. However, in this case the 2nd reject will
not be propagated to the user agent client (RFC3261). A SIP trace of the
phone will reveal this situation.

CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of nicolas
> Sent: Monday, May 24, 2004 3:49 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] RE: snom reporting busy when it shouldn't
> 
> I have no idea what is is, would be great if you can help me.
> 
> I think this problem is in conjuction with the problem that when i dial
> the
> snoms in idle, without answering (exten ==> s,1,answer) before.
> Then it is ringing once and then * becoming an busy too.
> (May be if the 2. ring is coming).
> 
> Thanks
> Christian
> 
> nicolas
> 
> 
> Christian Stredicke wrote:
> 
> > Did you check if the phone is in DND state? Is there anything strange on
> > the display?
> >
> > CS
> >
> >> -Original Message-
> >> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> >> [EMAIL PROTECTED] On Behalf Of nicolas
> >> Sent: Sunday, May 23, 2004 5:43 AM
> >> To: [EMAIL PROTECTED]
> >> Subject: [Asterisk-Users] snom reporting busy when it shouldn't
> >>
> >> I am using asterisk cvs.
> >>
> >> Incoming/Outgoing calls are working.
> >> Calling the phone when some other lines are in use on the phone is ok.
> >> What does not work though is when the phone is ringing, nobody else can
> >> call the phone anymore.
> >>
> >> That's what * is saying:
> >>
> >> -- Got SIP response 486 "Busy Here" back from 192.168.1.250
> >> -- SIP/snom1-4a44 is busy
> >>
> >> I am using the 2.05e snom200 firmware.
> >>
> >> Snom people sad must run.
> >> nico
> >>
> >> ___
> >> Asterisk-Users mailing list
> >> [EMAIL PROTECTED]
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
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> > [EMAIL PROTECTED]
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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 reporting busy when it shouldn't

[Christian Stredicke]

Did you check if the phone is in DND state? Is there anything strange on the
display?

CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of nicolas
> Sent: Sunday, May 23, 2004 5:43 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] snom reporting busy when it shouldn't
> 
> I am using asterisk cvs.
> 
> Incoming/Outgoing calls are working.
> Calling the phone when some other lines are in use on the phone is ok.
> What does not work though is when the phone is ringing, nobody else can
> call the phone anymore.
> 
> That's what * is saying:
> 
> -- Got SIP response 486 "Busy Here" back from 192.168.1.250
> -- SIP/snom1-4a44 is busy
> 
> I am using the 2.05e snom200 firmware.
> 
> Snom people sad must run.
> nico
> 
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RE: [Asterisk-Users] 2.05a firmware

[Christian Stredicke]

Now it's always proposing out of band DTMF. If there should be a user agent
which does not support RFC2833 it will answer the SDP accordingly and then
the phone automatically falls back to inband DTMF.

The setting was previously necessary because some equipment could not deal
with this negotiation process. However, it seems that such equipment does
not exist on the market any more. VoIP is getting more mature!

Christian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Lars Boegild Thomsen
> Sent: Monday, May 17, 2004 9:54 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] 2.05a firmware
> 
> While we're at the 2.05 firmware - the DTMF handling on the Codec
> configuration page have disappeared.  I assume this is because the phone
> now
> got some kind of default behaviour based on the codec.  Can you describe
> that behaviour?
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] Behalf Of Christian
> > Stredicke
> > Sent: 17 May 2004 15:21
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] 2.05a firmware
> >
> >
> > 8 kHz 16 bit/sample (linear) mono WAV files.
> >
> > CS
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger
> > > Sent: Thursday, May 13, 2004 7:31 PM
> > > To: [EMAIL PROTECTED]
> > > Subject: RE: [Asterisk-Users] 2.05a firmware
> > >
> > > Does anyone know what kind of file needs to be uploaded for the custom
> > > ring
> > > tone?
> > >
> > > --Ernest
> > >
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED]
> > > > [mailto:[EMAIL PROTECTED] On Behalf Of
> > > > Justin Huff
> > > > Sent: Thursday, May 13, 2004 10:09 AM
> > > > To: [EMAIL PROTECTED]
> > > > Subject: RE: [Asterisk-Users] 2.05a firmware
> > > >
> > > > > Whoohoo, they added a way to upload ring tones! My life is
> > > > now complete.
> > > > They also added the 'Name+Number' callerID display mode, yay!
> > > > Way to go SNOM!
> > > > --Justin
> > > >
> > > >
> > > > ___
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> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > > ___
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> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
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> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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Re: [Asterisk-Users] 2.05a firmware

[Christian Stredicke]

(Forwarded:)

- Original Message - 
From: "Usman Tahir" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Cc: "Christian Stredicke" <[EMAIL PROTECTED]>
Sent: Monday, May 17, 2004 11:25 AM
Subject: Re: [Asterisk-Users] 2.05a firmware


Hi,

In firmware release 2.05c and later, the user has an option to select the
ringer device when headset is in use. The default is Speaker as before. But
for call center and other closed environments, you can also select Headset.

In such a case ringing will be played on the headset. You'll find the
appropriate setting in Settings/Miscellaneous/Audio. Try the latest beta
from http://www.snom.com/download/share/snom200-2.05c-SIP.bin.

Regards,
Usman.

> > > -Original Message-
> > > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > > [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger
> > > Sent: Thursday, May 13, 2004 7:35 PM
> > > To: [EMAIL PROTECTED]
> > > Subject: RE: [Asterisk-Users] 2.05a firmware
> > >
> > > They also made a "bad" (for me) change. In 2.05a the phone would ring
> > > normally and I could press OK for headset or pick up the handset for
> > > handset. Now, when headset is enabled the phone only rings in the
> headset
> > > (i.e. not through speakerphone).
> > >
> > > --Ernest
> > >
> > > > -Original Message-
> > > > From: [EMAIL PROTECTED]
> > > > [mailto:[EMAIL PROTECTED] On Behalf Of
> > > > Justin Huff
> > > > Sent: Thursday, May 13, 2004 10:09 AM
> > > > To: [EMAIL PROTECTED]
> > > > Subject: RE: [Asterisk-Users] 2.05a firmware
> > > >
> > > > > Whoohoo, they added a way to upload ring tones! My life is
> > > > now complete.
> > > > They also added the 'Name+Number' callerID display mode, yay!
> > > > Way to go SNOM!
> > > > --Justin
> > > >
> > > >
> > > > ___
> > > > Asterisk-Users mailing list
> > > > [EMAIL PROTECTED]
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > > To UNSUBSCRIBE or update options visit:
> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > >
> > > ___
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> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>

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RE: [Asterisk-Users] 2.05a firmware

[Christian Stredicke]

Please take a look at http://www.snom.com/faq/FAQ-04-04-28-ut.pdf. It
describes the hand/headset policy! It was supposed to be an improvement...

CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger
> Sent: Thursday, May 13, 2004 7:35 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] 2.05a firmware
> 
> They also made a "bad" (for me) change. In 2.05a the phone would ring
> normally and I could press OK for headset or pick up the handset for
> handset. Now, when headset is enabled the phone only rings in the headset
> (i.e. not through speakerphone).
> 
> --Ernest
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Justin Huff
> > Sent: Thursday, May 13, 2004 10:09 AM
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] 2.05a firmware
> >
> > > Whoohoo, they added a way to upload ring tones! My life is
> > now complete.
> > They also added the 'Name+Number' callerID display mode, yay!
> > Way to go SNOM!
> > --Justin
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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RE: [Asterisk-Users] 2.05a firmware

[Christian Stredicke]

8 kHz 16 bit/sample (linear) mono WAV files.

CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger
> Sent: Thursday, May 13, 2004 7:31 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] 2.05a firmware
> 
> Does anyone know what kind of file needs to be uploaded for the custom
> ring
> tone?
> 
> --Ernest
> 
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > Justin Huff
> > Sent: Thursday, May 13, 2004 10:09 AM
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] 2.05a firmware
> >
> > > Whoohoo, they added a way to upload ring tones! My life is
> > now complete.
> > They also added the 'Name+Number' callerID display mode, yay!
> > Way to go SNOM!
> > --Justin
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
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RE: [Asterisk-Users] snom reporting busy when it shouldn't

[Christian Stredicke]

Right. When the phone
is ringing, there is nothing like „another ringing indication“. That’s
was done by intention, we simply wanted to keep things simple. One at a time! 

 

Christian

 



-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Of Frederic Steinfels
Sent: Sunday, April 25, 2004 10:20
AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] snom
reporting busy when it shouldn't

 

T Aksoy wrote:



Hi Frederic,   

What does not work though is when the phone is ringing, nobody else can    

call the phone anymore.By this, I presume you mean when the phone hasn't yet been answered?  

yes. It seems the snom200 can only handle one call
ringing. If one call is ringing, no more calls can 
ring, they will get the busy sign. There is no problem though if you have like
four open cals but none ringing,
then another one can ring.



We are seeing problems where if a snom user is on the phone and another callcomes in, then the person he/she was speaking to is automatically put onhold, and the incoming call answered. Auto answer is off on the snom. Thisphenomenon doesn't occur all the time. We are using 2.04f.  

haven't had that so far but both scenarios are not
allowed to happen in a productive area...
I hope somebody from snom can shed some light on this.



On 2.04k, putting a call on hold (R key) and then unhold, results in thecall getting dropped. Has anyone seen these issues as well?  


Some other problems I am experiencing is the
realm/login/username/account/line/password relationship.
When I put in wrong data, the phone says "not registered". When I
correct them, the phone does not
switch batch to "registered"; I need to reboot it.

Frederic

RE: [Asterisk-Users] Snom 200 Admin Password

[Christian Stredicke]

We did not see so far that a provider would pay the phone so it's only fair
that the user has the ability for example to change the provider. The user
owns the phone!

Btw take a look at http://www.snom.com/faq_en.php,
http://www.snom.com/faq/FAQ-04-03-24-sf.pdf. 

CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Chris Orme
> Sent: Saturday, April 17, 2004 11:39 AM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Snom 200 Admin Password
> 
> The tftp suggestion you received is well worth trying :-)  I didn't know
> that was possible.
> 
> Although I wonder if it works as it might mean that carriers deploying the
> snom may not be able to properly lock their phones perhaps??
> 
> Chris
> 
> On Sat, 17 Apr 2004, WipeOut wrote:
> 
> > Chris Orme wrote:
> >
> > >Hi.
> > >
> > >Did you buy the phone or get it second hand ?   If second hand do you
> have
> > >any paperwork from the person you bought it from and did they buy it
> > >through official distribution?
> > >
> > >If you got it through distribution I would am fairly sure your vendor
> > >might be able to help ?
> > >
> > >I have a rough idea of how it would be possible but I would think
> you'll
> > >probably have to prove ownership as this password is how carriers lock
> > >their phones.  If you got it from a carrier I imagine you might
> possibly
> > >have to pay them an unlock charge so you can change carriers.
> > >
> > >Or did you accidently set the admin password?
> > >
> > >
> > >Chris
> > >
> > >
> > >
> > I bought the phone new about a year ago so its not provider locked..
> >
> > I set the password to be nothing (I think) and then I set admin mode
> > off, then when I tried to get into the admin area I couldn't, it would
> > seem that either there is a bug that doesn't allow a blank password or
> > it did not set it to be blank..
> >
> > I will have to get hold of the distributor next week..
> >
> > Later..
> >
> > >On Sat, 17 Apr 2004, WipeOut wrote:
> > >
> > >
> > >
> > >>Hi,
> > >>
> > >>I have a Snom 200 that has had admin mode switched off and I have no
> > >>idea when the admin password has been set to.. Does anyone know of a
> way
> > >>to reset the phone to factory defaults??
> > >>
> > >>Later..
> > >>___
> > >>Asterisk-Users mailing list
> > >>[EMAIL PROTECTED]
> > >>http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>To UNSUBSCRIBE or update options visit:
> > >>   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >>
> > >>
> > >>
> > >
> > >___
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> > >[EMAIL PROTECTED]
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> > >   http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> >
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> 
> 
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RE: [Asterisk-Users] IAX firmware for snom 200s?

[Christian Stredicke]

We had this discussion recently on this mailing list and the answer is no,
not yet. We try to optimize the Asterisk interoperability on SIP level. We
at snom can currently not afford to open another development branch.

Christian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Justin Carlson
> Sent: Friday, April 16, 2004 9:27 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] IAX firmware for snom 200s?
> 
> is there a firmware for IAX for the snom 200's.  or are there any other
> hard
> phones that use iax(2)?
> 
> Thanks in advance!
> 
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RE: [Asterisk-Users] snom 200

[Christian Stredicke]

FYI there is a document that could give additional information:

http://www.snom.com/faq/FAQ-03-08-29-pp.pdf

If you have updates, please let me know. We would like to keep it
up-to-date.

Christian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger
> Sent: Tuesday, March 30, 2004 7:06 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] snom 200
> 
> > having problems with snom phone installstion
> 
> Please tell us what's up. I recently installed several SNOM phones and
> worked through many minor "issues." Let me know and I'll tell you what I
> can
> :)
> 
> --Ernest
> 
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RE: [Asterisk-Users] IAX and Snom200

[Christian Stredicke]

I think Asterisk should
have no problem with NAT, even when used with SIP. I mean just listen for the
first RTP packet and send the stream where it comes from (that’s called symmetrical
NAT). I think everybody is doing it like this now and they are selling their
stuff for thousands and thousands of dollars.

 

Well we do try to look
over our shoulders. There is a lot of tempting stuff out there, and making
decisions is difficult. At the moment I think it would be a mistake for us to
start another development branch. We simply have too many open issues with SIP
already. We hope to have a great phone (some day.-) that fits Asterisk pretty good
although it’s just using SIP…

 

Christian

 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Mulligan
Sent: Thursday, March 25, 2004
6:41 PM
To:
[EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] IAX
and Snom200

 



Certainly there is the NAT issue and this
should not be underestimated. Also IAX allows optimisation of existing
bandwidth between Asterisk servers.





The SNOM guys should look over their
shoulders at Verbiage who are bringing an IAX phone to market. I suspect it
will have a lot of interest amongst this community.





Brian





 





 





-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On
Behalf Of Barry Fawthrop
Sent: 25 March 2004 16:07
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX
and Snom200





 





- Original Message - 





From: Christian Stredicke






To: [EMAIL PROTECTED] 





Sent: Thursday, March
25, 2004 10:05 AM





Subject: RE:
[Asterisk-Users] IAX and Snom200





 



We thought about this
option. I guess the IAX2 is not the problem. We believe the real problem will
be the user interface.

 

snom would have no
problem providing the platform (hardware plus operating system and stuff like
audio), but we simply don’t want to open another development branch
(already got enough trouble with SIP.-).

 

I personally think its
ok to optimize the SIP interoperability. All that you can do in IAX can also be
done in SIP (or am I making a big mistake here?).

 

Christian

 

There is the big difference. in that IAX handles NAT much
better, esp. double NAT (security)

I'm not sure if you work for snom, but I'm willing to help
out where I can.

Anyone else care to list the differences between SIP and
IAX2?

If would be great to get a comprehensive list, Mark or the
digium guys ???

 

 

Barry

RE: [Asterisk-Users] IAX and Snom200

[Christian Stredicke]

We thought about this
option. I guess the IAX2 is not the problem. We believe the real problem will
be the user interface.

 

snom would have no
problem providing the platform (hardware plus operating system and stuff like
audio), but we simply don’t want to open another development branch
(already got enough trouble with SIP.-).

 

I personally think its
ok to optimize the SIP interoperability. All that you can do in IAX can also be
done in SIP (or am I making a big mistake here?).

 

Christian

 



-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Barry Fawthrop
Sent: Thursday, March 25, 2004
4:55 AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] IAX and
Snom200

 





Greetings





What would it take to have a snom200 support IAX,





what are the processes or having hardware to





support a new codec? Can this be tested and done





by a uesr or must this be done by the manufacturer?





 





Thanks in advance

RE: [Asterisk-Users] Snom 200 Voice Call / Paging

[Christian Stredicke]

BTW there is also another way to make the phone accept the call immediately.
Use a header like this in the INVITE:

Call-Info: answer-after=0

I guess some other phones (Polycom, Cisco?) also support this header for
"paging". Maybe it's better to support this way than to find a workaround
with the pure snom-proprietary intercom=true.

You need on of the latest images for this stuff (2.04c and on).

Christian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Tuesday, March 23, 2004 1:06 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Snom 200 Voice Call / Paging
> 
> Christian,
> I guess I am Confused about the 'header' stuff.
> 
> I am using the SIP strictly on a LAN as extensions to the
> [*].  Hence, I have a line in sip.conf like this:
> 
> [2200] ;snom 200
> callerid=Reception <2200>
> type = friend
> disallow=all
> allow=ulaw
> allow=alaw
> username = 2200
> secret = 2200
> host = dynamic
> dtmfmode = rfc2833
> context=intern
> mailbox = 2200
> 
> In extensions.conf I have
> exten => 2200,1,Dial(SIP/2200,20,tT)
> 
> Now, [*] is at 192.168.1.16. Where does the 'header' you
> refer to get sent?
> I tried adding intercom=true to the sip.conf but that is not
> it right?
> Lost ...
> Willy
> 
> - Original Message Follows -
> > To use "Intercom" mode in the current releases of the snom
> > 200, you need to use an "intercom=true" flag in the
> > To-Header. Essentially that makes the phone to pick up the
> > call immediately.
> >
> > To: 
> >
> > However, this mechanism is likely to change because of
> > security concerns and new interoperable methods.
> >
> > Christian
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:asterisk-users- [EMAIL PROTECTED] On
> > > Behalf Of [EMAIL PROTECTED] Sent: Sunday, March 21,
> > > 2004 5:25 PM To: [EMAIL PROTECTED]
> > > Subject: [Asterisk-Users] Snom 200 Voice Call / Paging
> > >
> > > To All,
> > > Several months (2003) ago there was a discussion
> > > regarding overhead paging & intercom functionality with
> > > SIP / Asterisk.  Jerry Gibson, John Todd and various
> > > others participated (from checking the archives).  One
> > > person even responded that they had the stuff working
> > > with the snom 200s.
> > > Voice Call (i.e. on-hook speaker/mic) is realy important
> > > in a lot of apps.  It would appear that the snom 200 and
> > > by extension the snom 105 support the functionality.
> > > I will be happy to make a wiki entry to explain & demo
> > > this functionality once I have it working properly.  I
> > > also understand that the (mis)use of conferencing is
> > > frowned upon as it wastes bandwidth and CPU.  However,
> > > until a better way comes around, that is not a problem
> > > as there are quite a few applications where (a) one
> > > needs Voice Call (which is 1 <-> 1) and / or an
> > > 'allPage' which can be limited to a subset of all
> > > phones.  Typically phones which are in designated or
> > > public areas, conference rooms, etc.  The BW/CPU issue
> > can be controlled. Better a limited solution than no
> > > solution at all ;)
> > > I am also allowing for the limitation that all
> > > participating phones are on the same LAN with the [*].
> > > Anyone who has this successfully working with snom,
> > > please respond ..  Using the [*] sound card for a
> > > separate PA system is NOT an option ;)
> > > As I said, I will be 'distilling' the info and turn it
> > > into a wiki entry.
> > > Cheers and TIA,
> > > Willy
> > >
> > > Willy Wouters
> > > ypOne Publishing
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> Willy Wouters
> ypOne Publishing
> 
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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RE: [Asterisk-Users] Snom 200 Voice Call / Paging

[Christian Stredicke]

Hi Willy, 

I must admit I am clueless when it comes to configuration files for
Asterisk. I am talking from a SIP point of view. Maybe someone can translate
this into Asterisk configuration files

Christian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Tuesday, March 23, 2004 1:06 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] Snom 200 Voice Call / Paging
> 
> Christian,
> I guess I am Confused about the 'header' stuff.
> 
> I am using the SIP strictly on a LAN as extensions to the
> [*].  Hence, I have a line in sip.conf like this:
> 
> [2200] ;snom 200
> callerid=Reception <2200>
> type = friend
> disallow=all
> allow=ulaw
> allow=alaw
> username = 2200
> secret = 2200
> host = dynamic
> dtmfmode = rfc2833
> context=intern
> mailbox = 2200
> 
> In extensions.conf I have
> exten => 2200,1,Dial(SIP/2200,20,tT)
> 
> Now, [*] is at 192.168.1.16. Where does the 'header' you
> refer to get sent?
> I tried adding intercom=true to the sip.conf but that is not
> it right?
> Lost ...
> Willy
> 
> - Original Message Follows -
> > To use "Intercom" mode in the current releases of the snom
> > 200, you need to use an "intercom=true" flag in the
> > To-Header. Essentially that makes the phone to pick up the
> > call immediately.
> >
> > To: 
> >
> > However, this mechanism is likely to change because of
> > security concerns and new interoperable methods.
> >
> > Christian
> >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:asterisk-users- [EMAIL PROTECTED] On
> > > Behalf Of [EMAIL PROTECTED] Sent: Sunday, March 21,
> > > 2004 5:25 PM To: [EMAIL PROTECTED]
> > > Subject: [Asterisk-Users] Snom 200 Voice Call / Paging
> > >
> > > To All,
> > > Several months (2003) ago there was a discussion
> > > regarding overhead paging & intercom functionality with
> > > SIP / Asterisk.  Jerry Gibson, John Todd and various
> > > others participated (from checking the archives).  One
> > > person even responded that they had the stuff working
> > > with the snom 200s.
> > > Voice Call (i.e. on-hook speaker/mic) is realy important
> > > in a lot of apps.  It would appear that the snom 200 and
> > > by extension the snom 105 support the functionality.
> > > I will be happy to make a wiki entry to explain & demo
> > > this functionality once I have it working properly.  I
> > > also understand that the (mis)use of conferencing is
> > > frowned upon as it wastes bandwidth and CPU.  However,
> > > until a better way comes around, that is not a problem
> > > as there are quite a few applications where (a) one
> > > needs Voice Call (which is 1 <-> 1) and / or an
> > > 'allPage' which can be limited to a subset of all
> > > phones.  Typically phones which are in designated or
> > > public areas, conference rooms, etc.  The BW/CPU issue
> > can be controlled. Better a limited solution than no
> > > solution at all ;)
> > > I am also allowing for the limitation that all
> > > participating phones are on the same LAN with the [*].
> > > Anyone who has this successfully working with snom,
> > > please respond ..  Using the [*] sound card for a
> > > separate PA system is NOT an option ;)
> > > As I said, I will be 'distilling' the info and turn it
> > > into a wiki entry.
> > > Cheers and TIA,
> > > Willy
> > >
> > > Willy Wouters
> > > ypOne Publishing
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> Willy Wouters
> ypOne Publishing
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
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RE: [Asterisk-Users] Snom 200 Voice Call / Paging

[Christian Stredicke]

To use "Intercom" mode in the current releases of the snom 200, you need to
use an "intercom=true" flag in the To-Header. Essentially that makes the
phone to pick up the call immediately.

To: 

However, this mechanism is likely to change because of security concerns and
new interoperable methods.

Christian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Sunday, March 21, 2004 5:25 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Snom 200 Voice Call / Paging
> 
> To All,
> Several months (2003) ago there was a discussion regarding
> overhead paging & intercom functionality with SIP /
> Asterisk.  Jerry Gibson, John Todd and various others
> participated (from checking the archives).  One person even
> responded that they had the stuff working with the snom
> 200s.
> Voice Call (i.e. on-hook speaker/mic) is realy important in
> a lot of apps.  It would appear that the snom 200 and by
> extension the snom 105 support the functionality.
> I will be happy to make a wiki entry to explain & demo this
> functionality once I have it working properly.  I also
> understand that the (mis)use of conferencing is frowned upon
> as it wastes bandwidth and CPU.  However, until a better way
> comes around, that is not a problem as there are quite a few
> applications where (a) one needs Voice Call (which is 1 <->
> 1) and / or an 'allPage' which can be limited to a subset of
> all phones.  Typically phones which are in designated or
> public areas, conference rooms, etc.  The BW/CPU issue can
> be controlled. Better a limited solution than no solution at
> all ;)
> I am also allowing for the limitation that all participating
> phones are on the same LAN with the [*].
> Anyone who has this successfully working with snom, please
> respond ..  Using the [*] sound card for a separate PA
> system is NOT an option ;)
> As I said, I will be 'distilling' the info and turn it into
> a wiki entry.
> Cheers and TIA,
> Willy
> 
> Willy Wouters
> ypOne Publishing
> 
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

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RE: [Asterisk-Users] Snom 200

[Christian Stredicke]

In Asterisk, that should be easy (I don't know if it is already done). The
trick is called "symmetrical NAT":

The Asterisk essentially ignored the media address of the SDP and waits for
the first RTP packet. It then just sends outgoing RTP to the address seen in
the incoming RTP packet.

Asterisk has the "advantage" that it does not try peer-to-peer RTP between
the user-agents. If the Asterisk is running on a globally routable address,
the symmetrical RTP trick is always possible, even if you have symmetrical
NAT.

The snom will allocate a port if you give it a STUN server. The STUN server
will have the address _stun._udp. (see RFC3489). If this server
is present, it will do its best to allocate a port on its own.

Christian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Barry Fawthrop
> Sent: Monday, March 22, 2004 8:40 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Snom 200
> 
> Progress
> 
> It seems I can't hear the Say Time, due to RTP Double NAT
> I'm guess this is both problem 1 and 2 really issue.
> 
> My config:
> IP Phone <-> Router (Nat) <-> Internet <-> Linux (NAT) <-> * Server
> 
> ANyone know of work arounds the double NAT? or other methods
> to route RTP with snom 200s, to work around this?
> 
> Thanks in advance
> 
> Barry
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RE: [Asterisk-Users] OT: SNOM and TAPI

[Christian Stredicke]

Yes, you can remote control the phone using the web interface. Just start
Ethereal, initiate a call from the web interface and then you see how it
works!

CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Greg Boehnlein
> Sent: Monday, February 23, 2004 11:14 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] OT: SNOM and TAPI
> 
> On Mon, 23 Feb 2004, Christian Stredicke wrote:
> 
> > I remember we had something one or two years ago, but I remember that
> was
> > not what I was dreaming of.
> >
> > Sorry we are not so good in implementing Windows-stuff... Maybe has
> someone
> > out there a template for TAPI? Something for someone who never did
> something
> > with COM or DCOM or .net or whatever...
> >
> > BTW click-to-dial can be initiated with a REFER request. That's 100 %
> SIP.
> 
> Someone also mentioned that you could submit Dial requests to the SNOM via
> it's Web interface.
> 
> --
> Vice President of N2Net, a New Age Consulting Service, Inc. Company
>  http://www.n2net.net Where everything clicks into place!
>  KP-216-121-ST
> 
> 
> 
> ___
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RE: [Asterisk-Users] OT: SNOM and TAPI

[Christian Stredicke]

I remember we had something one or two years ago, but I remember that was
not what I was dreaming of.

Sorry we are not so good in implementing Windows-stuff... Maybe has someone
out there a template for TAPI? Something for someone who never did something
with COM or DCOM or .net or whatever...

BTW click-to-dial can be initiated with a REFER request. That's 100 % SIP.

CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Andy Powell
> Sent: Monday, February 23, 2004 4:46 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] OT: SNOM and TAPI
> 
> 
> Snom  TAPI integration is a joke...
> 
> Andy

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RE: [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)

[Christian Stredicke]

BTW if you want to use Alert-Info on snom just provide an http uri which
points to an 8 kHz mono 16-bit sample WAV file.

CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Steve Creel
> Sent: Tuesday, February 10, 2004 11:08 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] alert-info and Cisco 7960 phones (6.1)
> 
> Getting the 7960 to use the Bellcore-dr1 through dr5 was fairly
> straightforward.  The release notes indicate that you can trigger other
> ringtones on the phone (in the section "Support for SIP Alert-Info
> Header"), but I can't get anywhere with it.
> 
> "...the Alert-Info header consists of a name of an internal tone or
> ringing pattern that can be played, as shown in the following example:
> Alert-Info: 
> There is no need to add a file extension (.au, .wav) to these names..."
> 
> 
> Does "internal" exclude those rings loaded from ringlist.dat?
> 
> Thanks for any thoughts or pointers,
> 
> Steve

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RE: [Asterisk-Users] The Smallest Asterisk Server Ever?

[Christian Stredicke]

Sorry, we have to make some money... Product business is tough!

:-) CS

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Greg Boehnlein
> Sent: Tuesday, February 10, 2004 10:39 PM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] The Smallest Asterisk Server Ever?
> 
> On Tue, 10 Feb 2004, Matthew Hardeman wrote:
> 
> > It's noteworthy that while Linux is GPL'ed, that doesn't mean that the
> > userspace applications that essentially make up the Snom phone and run
> > on top of the GPL'ed Linux kernel are.
> >
> > Snom will gladly give you their customizations to the kernel, and a
> > build environment that will produce a firmware image that can be loaded
> > onto the Snom's...
> >
> > They will not, however, unless policy has changed, give you the source
> > to the phone itself...  I tried.
> 
> Why should they? :)
> 
> --
> Vice President of N2Net, a New Age Consulting Service, Inc. Company
>  http://www.n2net.net Where everything clicks into place!
>  KP-216-121-ST

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[Asterisk-Users] FW: SNOM 200 silence suppression

[Christian Stredicke]

Hi Olle, you mail server thinks this is SPAM, so I resend it in the
mailing-list...

CS

-Original Message-
From: Christian Stredicke [mailto:[EMAIL PROTECTED] 
Sent: Sunday, February 08, 2004 12:54 PM
To: 'Olle E. Johansson'
Subject: RE: SNOM 200 silence suppression

Hi Olle,

we do silence suppression (slash half duplex) in hands free mode to kill the
echo. If the handset is being used, it is definitely turned off.

Did you try the latest Ethereal? It's able to generate .au files from G.711
RTP, usually a good first indicator what's going on on the cable.


CS

> -Original Message-
> From: Olle E. Johansson [mailto:[EMAIL PROTECTED]
> Sent: Sunday, February 08, 2004 9:55 AM
> To: Christian Stredicke
> Subject: SNOM 200 silence suppression
> 
> Christian,
> 
> I have a choppy sound from Asterisk to the Snom 200. Could it be that the
> SNOM supports
> silence suppression? If so, I can't find any setting in regards to this.
> 
> /O


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RE: [Asterisk-Users] Snom 200 MWI Button

[Christian Stredicke]

Please take a look at http://www.ietf.org/internet-drafts/draft-ietf-sipping-mwi-04.txt.
The snom phone tries to use the Message-Account line, if it’s present;
otherwise it will take the From header: 

 

Messages-Waiting: yes

Message-Account:
sip:[EMAIL PROTECTED]

Voice-Message: 2/8
(0/2)

 

Hope this helps,

 

Christian 

 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Oster
Sent: Sunday, February 08, 2004
2:22 AM
To:
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Snom 200
MWI Button

 



I'm trying to get the MWI button to work with my Asterisk
configuration.  The snom is accepting and responding to the Message
indications from *, but when I press the MWI button, it is dialing my extension
(the one with the voice mail on it).





 





I'm wondering if there is a way to specify what extension to
dial to check email in the configuration, either the phone, or * itself.





 





Asterisk Version 1/30/2003 checked out and compiled this
evening





Snom Version 2.03o (most recent auto-update)





 





Any help would be greatly appreciated.  At one point
Mark had talked about adding a voicemail= directive in sip.conf on the
mailing list at one point, however grepping the code doesn't reveal a feature
like that at this time.





 





Anyone have success in getting the MWI button to work on
Snoms?  If so I would LOVE to hear from you.





 





Paul M. Oster





 

RE: [Asterisk-Users] Sip flow diagram?

[Christian Stredicke]

You can find some examples here:

http://www.iptel.org/info/players/ietf/callflows/

Enjoy reading... SIP is like poetry!

Christian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Rich Adamson
> Sent: Wednesday, February 04, 2004 6:45 PM
> To: Asterisk-a-users-list
> Subject: [Asterisk-Users] Sip flow diagram?
> 
> 
> Does anyone have a high level flow diagram showing acceptable sip
> messages exchanges?
> 
> For exampe:
>   Source Dest
>   Invite   ->
><-Trying
>   Ok   ->
> 
> I'm specifically trying to debug an issue with various hangups, prior
> to call completion, after call completion, "calling" vs "called" party
> hold, etc, and getting rather confused watching the various packets
> flowing between sip devices with a sniffer (and no reference document).
> 
> Rich
> 
> 
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RE: [Asterisk-Users] The Smallest Asterisk Server Ever?

[Christian Stredicke]

Well I also though about this five minutes ago... I think the biggest
problem should be memory (we have 16 MB DRAM and 4 MB Flash). 

Also, the question is if the plastic makes a "box" impression...

Christian

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of listas iPfone
> Sent: Tuesday, February 03, 2004 7:16 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
> 
> Snom Does gives the souce and more:
> 
> http://www.snom.com/sources_en.php
> 
> - Original Message -
> From: "Chris Albertson" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Tuesday, February 03, 2004 4:01 PM
> Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
> 
> 
> >
> > I read a report of Asterisk running on a Microsoft X-Box.
> > That's kind of a stunt as you could buy a decent PC for
> > the price of a Linux-capable XBox.  Id's still like to
> > see Asterisk run on very low-end hardware
> >
> > The Snom IP phone runs Linux inside?  I assume as Linux
> > is GPL'd Snom will supply the source code?  It would be
> > fun to install an Asterisk server in a phone.
> >
> >
> >
> > --- Panny Malialis <[EMAIL PROTECTED]> wrote:
> > > Does anyone have it running on a Cyclades T100 ? same as used for
> > > ntop/nbox.
> > >
> > > I was thinking of using that as an IAX->sip translator for offices
> > > with NAT.
> > >
> > > CPU MPC855T (PowerPC Dual-CPU)
> > > Memory 32MB RAM / 4MB Flash (TS100)
> > > Interfaces1 Ethernet 10/100BT on RJ45
> > > 1 RS232 Console on RJ45
> > > RS232 Serial Ports on RJ45
> > >
> > > Looks like fun! Although a little lacking on memory.
> > >
> > > Any comments?
> > >
> > > Panny
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> > =
> > Chris Albertson
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