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:
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
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
-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
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
: 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
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:
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
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
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
...@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
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 -
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
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:
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
] 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
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
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:
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
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:
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
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
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
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]
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
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:
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
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
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:
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:
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
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
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 (lets say: type A) behave different than snom 300 (type B). So
there
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
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:
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
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
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
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
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,
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
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 -
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
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
___
--Bandwidth and
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
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]
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
: 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
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
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 -
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
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
Try loading
http://phone-ip-address/line_sip.htm?settings=saveuser_dp_str1= (if
that was in the line 1) while the phone boots up (keep your finger on
the reload button). If that does not work, you need to do a tftp update.
Also consider moving to version 4.5
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,
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
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
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
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
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
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
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
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
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
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,
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:
. 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
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
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:
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
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
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:
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,
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 -
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
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,
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
. 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
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
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,
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
(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
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
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-
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]
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
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 dont want to open another development branch
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
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
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
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
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: sip:[EMAIL PROTECTED];intercom=true
However, this mechanism is likely to change because of security concerns
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...
, 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
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
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:
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 its present;
otherwise it will take the From header:
Messages-Waiting: yes
Message-Account:
sip:[EMAIL PROTECTED]
Voice-Message: 2/8
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
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
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
Maybe you have to update Asterisk (see
http://bugs.digium.com/bug_view_page.php?bug_id=732). snom is now a
little bit picky about line-ID (see the discussion in the bug).
CS
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ariel
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