I never saw the point of spending $100 for something that is so limited.
You can spend a little more and get something like an ALIX board that
is so much more capable and still fanless/low power.
http://www.pcengines.ch/alix.htm
The 2d3/2d13 are very nice for the price.
Hi,
Can you add
On Thu, 18 Nov 2010, John Novack wrote:
Not really in production But for a SIP/IAX Asterisk box, it works!
there is a Dockstar hacking site that de-nuts the boot code and allows
booting from a 1-2 gig flash ( I have not had good luck with 4 and 8 gig
flash, but it could be the flash sticks.
Hi,
No we can not add a DSL modem.But i saw on the net that this works for call
forward feature on TDM400 FXS card.So i tried it.The data is entering the
CFIM family in the database which can be seen but only the call is not
forwarded
Regards,
Aparna
On Fri, Nov 19, 2010 at 2:56 PM, Olivier
Hi,
It worked finally with GSM Codec only enabled at client side.. Initially
with G.711 (u-low) , G.711 (A-low) and GSM it didn't work. All enabled
by setting [CLI] sip set debug on
I saw asterisk having following logs..
-- Remotely bridging SIP/macbook-0041 and SIP/tharindu-0042
Hello,
I have a Sip phone (Siemens C470IP) which works perfectly with
different VoIP providers (iptel, betamax, ovh...). It also worked well
with my testing server (ubuntu and inside the LAN).
But now the problem i have is that the hardphone doesn't connect to my
dedicated server (debian lenny /
On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier c...@chab.info wrote:
Hello,
I have a Sip phone (Siemens C470IP) which works perfectly with
different VoIP providers (iptel, betamax, ovh...). It also worked well
with my testing server (ubuntu and inside the LAN).
I am assuming you mean
Thanks Alejandro, you were right it was just a NAT problem ! i add a
stun server in the phone configuration and it works :)
2010/11/19, Alejandro Imass a...@p2ee.org:
On Fri, Nov 19, 2010 at 7:28 AM, Benoit Chabrier c...@chab.info wrote:
Hello,
I have a Sip phone (Siemens C470IP) which works
On Fri, Nov 19, 2010 at 8:23 AM, Benoit Chabrier c...@chab.info wrote:
Thanks Alejandro, you were right it was just a NAT problem ! i add a
stun server in the phone configuration and it works :)
Cool. Also Asterisk SIP conf file has some NAT settings as well that
you can play with and perhaps
Hi,
In Asterisk 1.8.0 dialplan command Dial(SIP/peer_name) produces errors
if no such peer_name defined instead of just saying peer not found:
[Nov 19 20:01:23] ERROR[7827]: netsock2.c:245 ast_sockaddr_resolve:
getaddrinfo(sdf, (null), ...): Name or service not known
[Nov 19 20:01:23]
On Fri, Nov 5, 2010 at 10:58 AM, Bob Beers bob.be...@gmail.com wrote:
Hi list,
My need is to append a site specific parameter to the
Contact: header on all INVITEs exiting * via a SIP trunk.
I'd like it to look something like this:
Contact: bob:3125551...@10.10.10.10;SITE-ID=us.here
The linewrapping by gmail of the patch file makes it difficult to read.
So, I added it as an attachment for any interested readers.
--
-Bob
--- asterisk-1.8.0-beta2.orig/channels/chan_sip.c 2010-07-26 15:59:03.0 -0400
+++ asterisk-1.8.0-beta2/channels/chan_sip.c 2010-11-05
Why not just use tiff2pdf ?
tiff2pdf input.tif -o output.pdf
William Stillwell
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Michael
Sent: Friday, November 19, 2010 9:43 AM
To:
On Fri, 19 Nov 2010 10:43:28 + (GMT), Gordon Henderson wrote:
Interestingly for commercial units, I've had the opposite experience -
I've found that my (business) customers just will not pay for something
tiny that's capable of supporting 30 phones... I did have a look at the
GuruPlug
So tiff2pdf and then gs back to tif? I was hoping for a cleaner method
Original Message
Why not just use tiff2pdf ?
tiff2pdf input.tif -o output.pdf
William Stillwell
--
_
-- Bandwidth and Colocation
On 11/19/2010 10:10 AM, Michael Graves wrote:
On Fri, 19 Nov 2010 10:43:28 + (GMT), Gordon Henderson wrote:
Interestingly for commercial units, I've had the opposite experience -
I've found that my (business) customers just will not pay for something
tiny that's capable of supporting
On Fri, Nov 19, 2010 at 9:42 AM, Michael voip.quest...@gmail.com wrote:
Hello,
We succeed to send faxes using FFA, when the files are converted to tif
from PDF using gs, but it doesn't work with tif files we copy/upload
directly from our PCs.
We saw in the manual that the size is important,
Hi all,
I've got 4 actors on my stage:
Alice calling from outside
Bob transferring incoming calls to Charlie
Charlie who has a mobile phone
My PBX which is connected to my ISDN line.
I want Charlie to see Alice's Callerid after Bob has transferred the
call as if Charlie is receiving the call
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:34 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] callerid not forwarded when transferring
Please try this in your dialplan Set(CALLERID(name)=${CALLERID(num)})
Some where I tried and it worked with VoIP account A to B as VoIP trunk and
B forward the call to C whereas in C A's number will be displayed.
If you could paste more details as Danny said that would help the list to
assist you
Forgot to tell you the version I tried is Asterisk 1.4 with TrixBox, I
disabled the caller-id checkbox while creating VoIP trunk then it started
working for me..
On Fri, Nov 19, 2010 at 9:21 PM, Gopalakrishnan A.N sai...@gmail.comwrote:
Please try this in your dialplan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio
Incantalupo
Sent: Friday, November 19, 2010 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] callerid not
Hi Gopalakrishnan A.N,
I tried it but it seems like my telco is overwriting the value I set as
callerid.
Maybe it is possible with Voip providers only.
Giorgio Incantalupo
Gopalakrishnan A.N wrote:
Forgot to tell you the version I tried is Asterisk 1.4 with TrixBox, I
disabled the caller-id
Hi Danny,
I'm using Asterisk 1.4 and I'm using SetCallerPres and
Set(CALLERID(name)=XX) apps but I always get my telco callerid.
Which Asterisk version would you suggest?
Thanks!
Giorgio
Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi all,
i have a little problem to understand this warning message, it's annoying
and it cause a lot of spurious in the log files.
Im working with this scenario:
a sip outgoing trunk (Trunk-out) that support only ulaw and all calls are
always routed to this.
a list of sip UAs that potentially can
On Sun, 14 Nov 2010, Gordon Henderson wrote:
On Sun, 14 Nov 2010, Gordon Henderson wrote:
On Sun, 14 Nov 2010, Tzafrir Cohen wrote:
On Sun, Nov 14, 2010 at 04:38:25PM +, Gordon Henderson wrote:
Well, just to follow this up - it looks like there is no DAHDI and BRI
support in asterisk
I guess it will not work with PSTN lines since the control is transferred to
the Exchange. I am not too sure, I am just sharing my thoughts
On Fri, Nov 19, 2010 at 9:28 PM, Giorgio Incantalupo
gincantal...@fgasoftware.com wrote:
Hi Gopalakrishnan A.N,
I tried it but it seems like my
Hello,
We succeed to send faxes using FFA, when the files are converted to tif
from PDF using gs, but it doesn't work with tif files we copy/upload
directly from our PCs.
We saw in the manual that the size is important, since we got the error
FAX handle 0: failed to queue document
Good afternoon list.
I need to make calls via AMI, but I need to leave the links in their
respective contexts, to mobile phone calls by leaving out the context of
mobile and so on.
Already configured the settings that way, but I do not like the the Action
Originate do it. I tried several ways,
I'd like to start playing with 1.8, however I don't want to potentially
damage anything on my existing 1.6.2 install on my production server.
I'd like to test 1.8 against my existing configs leaving my 1.6.2
install untouched. Looking at the output of ./configure --help suggests
that it's
Hi,
I'm experiencing the same problem. We have 2 office locations and the
Asterisk server is at one of them. At the other location, all SPA941 access
the Asterisk server over an Internet link. All phones are set to nat=yes
at the remote location.
So my problem is that the MWI doesn't work at the
Hi Stephen,
That's what people do when building precompiled packages for certain
distros (along with a few more things).
I use to do the following when building packages (with a few more options):
./configure --prefix=/usr --sysconfdir=/etc
make
make install DESTDIR=/my/destination/directory
Depending on what telco Charlie is connected to would change the CallerId
presented to Charlie from being Alice's or Bob's Cid.
When a call is forwarded, Charlie's telco receives different ANI and CID :
some (seems to) favor ANI and some CID.
An interesting thing to test is to let Bob issue a
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