:35 PM, Michael Picher <mailto:mpic...@ezuce.com>> wrote:
But I think if we do what you're suggesting we'd break Polycom
provisioning... which is the most used phone on the system...
On Tue, Sep 27, 2011 at 12:28 PM, Alberto mailto:g...@interfree.it>> wr
I'm not an expert on dhcpd.conf but seems to me that the code won't
break anything:
class "snom" {
*match if substring (hardware, 1, 3) = 00:04:13;*
option tftp-server-name"http://:8090";
option bootfile-name "/phone/profile/docroot/{mac}.xml";
}
the strin
Il 27/09/2011 12:47, Michael Picher ha scritto:
> sipxecs's own dhcp is just dhcpd on the sipxecs server. modify config
> in /etc/dhcpd.conf
>
yes I know
> we don't have fine-grained control over dhcp config files in sipXecs.
> If you're afraid of command line stuff you can always install webmi
= 00:04:13;
option tftp-server-name"http://:8090";
option bootfile-name "/phone/profile/docroot/{mac}.xml";
}
Can we add this to sipxecs own dhcp?
Thanks
Alberto
Il 27/09/2011 11:02, Douglas Hubler ha scritto:
On Tue, Sep 27, 2011 at 2:18
on a
new snom plug-in specifically designed for version 8.x.
Best regards
Alberto
Il 26/09/2011 18:46, Michael Picher ha scritto:
there's already a note about this in the snom page in the wiki
(http://wiki.sipfoundry.org)
Thanks,
Mike
On Mon, Sep 26, 2011 at 12:07 PM, Domenico Chier
owser.
You must substitute of course {mac} with the mac you provided in sipxconfig.
Alberto
Il 23/09/2011 14:22, John Pi ha scritto:
> Content-Type: text/plain;
>charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> X-FUDforum: 08063afc
o set this up for outgoing calls.
Is there any best practice to archive this?
Thanks in advance and best regards
Alberto
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x27;t have much time to finish the work right
now, and I won't probably contribute what I believe is not completed.
But some test on it would be appreciated, at least to identify bugs on
what is already done.
Alberto
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ta plug-in I'll be happy to share it.
I'm using it with 4.2.1
Alberto
Il 31/08/2011 12:23, Sven Evensen ha scritto:
Has anyone successfully deployed the 870? Does it auto configure? I
read from some posts last year you had to configure it manually
Are there any features that dont
ig file.
I would just try to configure it as a 370 and cross fingers ;-)
Let me know if it works
Alberto
Il 18/03/2010 16.07, Scott Howell ha scritto:
> Content-Type: text/plain;
>charset="utf-8"
> Content-Transfer-Encoding: 8bit
> Organization: SipXecs Forum
> X-FUDforu
igits).
Regards
Alberto
Il 26/12/2009 15.39, thomas peterseil ha scritto:
> hello listmembers,
>
> is somebody using a portech mv-370 or mv-372 gsm gateway with a sipx server?
> i don´t know, how to add this gateway in sipx.
>
> thank you very much!
>
Hi Dennis,
great news! Do you know if this issue is planned to be fixed in a 7.3.x
relase too?
Thanks
Alberto
Dennis Wallen ha scritto:
I haven't seen anyone else using the Snom 320 in a remote worker
setup, but I thought I'd share some new information that may be useful
if you att
Hi Nikolay,
excellent news. I will try to bring up the open issue on the forum ...
just to make sure they don't ignore them.
Good job!
Alberto
Nikolay Kondratyev ha scritto:
Alberto,
Finally snom team is working on this problem.
http://forum.snom.com/index.php?showtopic=2094
I hope
Damian Krzeminski ha scritto:
Alberto wrote:
Hi Nikolay,
the speed dial bug is a know old bug.
Please see for reference:
http://track.sipfoundry.org/browse/XX-5478
The patch I submitted to fix this and others was never applied to the
source tree. Anyway I kept on updating the Snom v7 plug
upper model.
Don't expect the resource list to update any function keys and make it
blink. It will just show something on a 360 and above large screen.
There is no way I currently know to assign the sipxecs managed resource
list to the function keys lights.
Let me know
Alberto
Ni
meone reported it solved some issues.
The only option I see that might be related is:
Authentication for SIP Reboot
but from the help it doesn't seems the solution.
Let me know. If you find something interesting keep me updated. I might
add your discoveries to the Snom plug-in.
Than
Hi Nikolay,
you're right the wiki page was not updated accordingly after the last
snom plug-in update.
I should ask for permissions to update it.
What did you discover about dhcpd?
Alberto
Nikolay Kondratyev ha scritto:
Hi Alberto,
I pasted the line
speed 3663
Just befor
know yet if I
will add all function keys management from sipxconfig.
Let me know
Alberto
Nikolay Kondratyev ha scritto:
Hi all,
There is a "mailbox" configuration parameter in the snom phone.
And this parameter is used for both mwi subscription and voice mail
retrieving when pu
Quan Shi ha scritto:
Thanks, Alberto,
This mail list is great!
I am wondering where I can find the detailed info about the directory
architecture of the SipX. I would like to find where the file is.
You'll find generated Snom profiles in :
/var/sipxdata/configserver/phone/profile/do
croot/{mac}.xml
<http://%22your_sipx_hostname%22:8090/phone/profile/docroot/%7Bmac%7D.htm>
There are a couple of minor bug on the Snom plug-in in 4.0.1 that you
might hit. Just in case I can provide some updated code.
Alberto
Quan Shi ha scritto:
Hi, all,
I am a new comer to the
and
are unable to register.
Let me know
Alberto
IT Services ha scritto:
> Hi there:
>
> I am running 4.01 with windows DNS. There are SRV records.
>
> The SNOM 320 phones are not registering. The phones pick up the config
> file, but fail to register. In the logs, there are these
tates they support sip REFER.
Alberto
Tony Graziano ha scritto:
Neat phone. The link you sent was for the S685, which is the analog version.
I'm guessing you meant the S685IP which fit the specs of what your email said.
http://gigaset.com/shc/0,1935,hq_en_0_152411_rArNrNrNrN,00.html
Looking t
control its status. No chances to do phone provisioning. Apart
this it works very well and it's quite cheap.
Alberto
milosz ha scritto:
> hi all,
>
> what are people using for cordless handsets? i am not happy with my
> snom m3's
Hi Matt,
I did the test with Snom already some weeks ago. I noticed they have
issues with 4.0.1 when GRUU is enabled. I did not test them with 3.10.x
and gruu.
Alberto
Matt White ha scritto:
On 7/9/2009 at 1:24 PM, in message
<4a55ef8e02f1b...@mail.thesummit-grp.com>,
Scott Lawrence ha scritto:
> What is it that you would rather have happen instead of going to
> voicemail in these cases?
>
>
Well the idea I had in mind was some sort of busy tone. But as I
understood it's the phone that should do that in case something goes
wrong with the blind transfer. Wha
Dale Worley ha scritto:
On Wed, 2009-07-08 at 17:58 +0200, Alberto wrote:
What happen is that an company operator that wants to blind transfer
(quickest choice in busy hours) an external call to some destination
cannot predict if the call will go to the voicemail. The external
caller is
ernal extensions and as soon hears the
voicemail for "extension some unknown number" he often drop the call and
call in again.
Thanks
Alberto
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Hi all,
is there any practical way to avoid a blind trasfered call to end up on
the destination voicemail?
This in case of course the destination is triggering the voicemail for
any possible reason.
Thanks in advance
Alberto
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Hi,
I tried both Nokia E51 and E90 and I confirm it to be a nice wifi sip
device. It's not perfect unfortunately. I had some issues with blind
transfers directed to them. Then I believe it does not support dns SRV
records.
Alberto
Gabor Paller ha scritto:
Personal experience: we use
es to its
requests.
Don't really know if it is a specific SRXN3205 issue or it could be
generalized to all Netgear Prosafe with at least the same firmware revision.
I could spot the issue doing packet captures on the router on the WAN side.
Regards
Alberto
Andreas (Around the Clock
itsp / remote
sipxecs used.
Alberto
Boy Aidil Sjam ha scritto:
Hi All,
Let say, I have two sipXecs in different domain or one sipXecs server
connected to other SIP server (ITSP) or gateway. The connection
between both location is separated with WAN connection with limited
bandwidth to
Hi all,
what codec are currently supported in for Moh services and Park server?
After the Freeswitch introduction I thought many services were using a
larger set of codecs.
Park server is not based on Freeswich?
Thanks
Alberto
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don't just add a link/button for every cdr record
to download the sipxviewer xml file of a single call? It's just a matter
of storing the call-id in the db.
I would be simply ... lovely! Any thoughts?
Alberto
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Hi Tony,
thanks. I'm working out http://track.sipfoundry.org/browse/XX-5475 and
now I know I should not waist my time with Bria to test some
interoperability.
Thanks for the infos.
Alberto
Tony Graziano ha scritto:
> Workgroup doesn't work in bria pro, they (counterpath) have
>
the message "Subscribing... please wait ...".
What am I doing wrong?
Using 4.0.1-015696 in CentOs 5.3
Alberto
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Unsubsc
Hi Dale,
thanks. The steps you suggested work for me. At least for VM notification.
I will test alarms soon.
What about having in sipxconfig the possibility to specify a SMTP relay
and authentication data if needed?
Thanks again
Alberto
Dale Worley ha scritto:
On Mon, 2009-05-18 at 11:27
Scott Lawrence ha scritto:
On Mon, 2009-05-18 at 11:27 +0200, Alberto wrote:
Hi everybody,
I'm trying to figure out why my system (CentOs 5.3 - 4.1.0-015538 ) is
not able to send emails (both VM notifications and alarms).
You know that you're using the unstab
tem to work at least as expected sending all
mail generated?
Thanks in advance
Alberto
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in
sipxconfig, but believe me I could not even try it. I don't know how to
do it ... I might appreciate some help here ... at least to reach a
basic manual BLF working configuration.
Tony Graziano ha scritto:
Thanks for clarifying that Alberto.
Should one assume the 370 VPN feature is n
27;t even now if the new v8 firmware is simply a subset of v7
or a completely different one.
Alberto
Tony Graziano ha scritto:
I know the plug-in is being worked on to go to version 7. Currently in sipxecs the snom
370 or the 820/870 are not currently supported. The 370 is not significantly dif
"yum install sipxecs" as I
did tens of times. I did not use the sipxecs ISO. Are the required fonts
included as a dependency?
Thanks in advance and best regards
Alberto
"2009-04-30T16:48:28.669000Z":733:JAVA:WARNING:pbx.ictengineer.net:P1-18::RequestExceptionReporter:&
's needed is a domain match rule in forwardingrules.xml.
Alberto - please file an issue in the tracker on this. No need to
repeat the full description - just put a link to this thread in the mail
archive:
http://list.sipfoundry.org/archive/sipx-users/msg14281.html
__
Robert Joly ha scritto:
> I think that the fundamental problem here is that the domain part of
> '12345...@voip.eutelia.it' is that of the ITSP. Given that the sipXecs
> is not authoritative for that domain, I do not believe that any kind of
> dialplan rule can be created that will solve this prob
s was not an issue at all when I could enable "Internet Calling"
choosing sipxbridge as the default sbc. Now it's not possible cause
sipxbridge is not in the list anymore.
Thanks again
Alberto
Scott Lawrence ha scritto:
On Thu, 2009-04-30 at 09:04 +0200, Alberto wrote:
I&
Hi Scott,
thanks for your reply. Attached the files you requested.
Alberto
Scott Lawrence ha scritto:
On Thu, 2009-04-30 at 09:04 +0200, Alberto wrote:
Hi,
I'm currently running sipxecs 4.0 stable and I ran in a simple issue.
I'm perfectly using my ITSP (eutelia) with 2 differen
e now is I cannot
enable Internet Calling cause sipxpxbrige is not in the default SBC list
on that page.
What is the recomended behavior to handle this?
Thnaks in advance
Alberto
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Steve,
I think this the file that you need.
The Asterisk is version 1.2.24.
Thanks,
Alberto
- Original Message -
From: "Farr, Steven C."
To: "Alberto Furtado" ; "Scott Lawrence"
Cc:
Sent: Wednesday, December 17, 2008 1:16 PM
Subject: RE: [sipx-users]
Steve,
These are the sip.conf and extensions.conf files. The name of the SIP trunk
that
is trying to register to my Sipxpbx is TELELIGUE.
Thanks,
Alberto
- Original Message -
From: "Farr, Steven C."
To: "Alberto Furtado" ; "Scott Lawrence"
Cc:
Sent: T
If you're requirement is to route calls _to_ that Asterisk server in
> order to call those 10 lines, you could do that by configuring the
> address of the Asterisk server as an Unmanaged Gateway and create a
dial
> plan that routes those numbers there
>
No my problem is to ro
call those 10 lines, you could do that by configuring the
> address of the Asterisk server as an Unmanaged Gateway and create a dial
> plan that routes those numbers there
>
No my problem is to route call out of the asterisk in to the Sipxpbx
so I can terminate them on my GSM
info of the Sipxpbx user in the "outgoing settings" of Asterisk.
The Asterisk sends a
Sip Register every 20 seconds, and they all receive a 401 error.
My previous email has a sample from the registrar.log with the 401 response.
Thanks for your quick responses,
Alberto Furtado
- Origin
Looking at the sipregistrar directly there seems to be an error with invalid
nonce. But I haven't a clue to what this means...
"2008-12-16T20:20:46.143029Z":60045:SIP:DEBUG:sipx.xx.net:SipRegistrarServer:B6C61B90:SipRegistrar:"SipNonceDb::nonceSignature:
callId='22f910762b6fe742799e9fb57ddf8
rence"
To: "Alberto Furtado"
Cc:
Sent: Tuesday, December 16, 2008 5:47 PM
Subject: Re: [sipx-users] Help: Asterisk Registration in to Sipx - 401 Error
On Tue, 2008-12-16 at 16:39 -0200, Alberto Furtado wrote:
>
> Hi All,
>
> I am trying to register 2 Asterisk in a br
No, because Ast. has not registered to Sipx.
The problem is probably with a parameter in Asterisk conf.
But I don't know what Sipx expects to accept a register.
- Original Message -
From: Picher, Michael
To: Alberto Furtado ; sipx-users@list.sipfoundry.org
Sent: Tu
Hi All,
I am trying to register 2 Asterisk in a branch office to my SIPX. All my other
stuff registers fine
but the Asterisks get a 401 Error. The only difference that I see is that the
log shows that the register
string has two extra parameters:
algorithm="MD5" and opaque=""
and that the FRO
pX
I had to configure the gateways ports as users, log them in an do all the
answer planning in the
gateway and then call the phones thru SipX so they could answer external
calls.
I am still testing SipX so I hope my post will help you.
Alberto Furtado
B2Br
- Original Message -
From: "C
dial plans, activate them or in the gateways your have to
manually restart SipXecs.
Going to continued my tests now with the rest of the product.
Thank you very much Mike and Tony for you help,
Alberto
- Original Message -
From: Alberto Furtado
To: Tony Graziano ; sipx-users
going to put a log debug server on the Sipura to see if it is receiving the
INVITE request.
I have this Sipura FXO configured for a Unmanaged Gateway with port set
to 5060 and protocol set for AUTO in sipx.
Thanks,
Alberto
- Original Message -
From: Tony Graziano
To: Alberto Fu
with the dial plans or the gateway configuration. But I have no
clues
to what I did wrong.
I even added the Emergency that doesn't need permissions and the dial plan is
simpler!!!
Greetings from Rio de Janeiro and
Thanks very much for your input Mike,
Alberto
- Original Message -
am probably missing out on something very simple in the configuration of the
server.
I would REALLY appreciate any suggestions!!!
Thanks in advance for the patience with a newcomer,
Alberto
PS: Its been as exhausting week after being unable to install on Fedora, unable
to use the
auto
x27;t have enough time and different snom phone models
to start developing myself.
Alberto
Tony Graziano ha scritto:
Agreed snom m3 and 370 support would be nice to have.
Polycom config files are supported and is an xml format.
Since this is "newer" firmware for snom phones, th
forum.
Thanks anyway
Alberto
Dale Worley ha scritto:
On Sat, 2008-06-21 at 08:51 +0200, Alberto wrote:
Hi,
I'm actually experiencing a quite odd behavior when I try to blind
transfer a call having a Snom 300 as a destination.
In my testing enviroment I have 3 phones:
- Snom 300 (ext: 201
ly
dealing with NAT Traversal I tried use STUN. Most of them don't even
have an option setting to set a static IP. I really cannot judge if STUN
is reliable or not, but I would definitely vote for an option to allow
dynamic IP deployments to be a
x27;s a strict limit. Right now sipxbridge can determine
reliably the public IP using STUN and self heal in case of an IP change.
Why not share the same existing feature?
Alberto
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dynamically
assigned from the ISP?
Thanks in advance and best regards
Alberto
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Ciao Federico,
I could set up my Cisco 7960 to show date and time using sipxconfig and
setting:
sntp_mode = directedbroadcast
sntp_server = ntp.ien.it
both in the phone Network Parameters.
Alberto
Federico Sirtori ha scritto:
Thanks to Duncan i found a
page:http://www.voip-info.org/wiki
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