Hi suggest this :
http://www.patapsco.co.uk
Maybe a little bit more expensive but a very good product.
Map
On Thu, Aug 28, 2008 at 3:08 AM, George Pajari [EMAIL PROTECTED]wrote:
Why a three-port PRI card?
Just put a two-port card into your Asterisk server, pull off those DIDs
you want
Yes you can.
Obviously you have to compile, configure and add chan_h323 to Asterisk.
Map
On Thu, Aug 28, 2008 at 10:32 AM, mahboob zaman [EMAIL PROTECTED]wrote:
hi.
i have two IP phones that are in H323 protocol. How can i test that
these two phones are working? For IP phone (SIP) i used
Hi all,
maybe there is no opener device at all.
Anyway take a look here :
http://www.barix.com/
On Thu, Jul 24, 2008 at 12:11 PM, Gordon Henderson
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Thu, 24 Jul 2008, Chris Bagnall wrote:
Greetings list,
We have a client with an analogue door
correctly work but it seems that's not enough.
Thank you.
Giorgio
map wrote:
Hi Giorgio,
From your email seems clear that your Asterisk box can not resolve
tnet.it http://tnet.it and SIP register messages are not replied.
I suggested to check if your Asterisk box is really sending SIP
with the other provider we are using
while tnet.it is making me get crazy
Thank you.
Giorgio
map wrote:
Hi Giorgio,
Just to recap:
1) you are able to connect to tnet.it http://tnet.it by using the
same account of your asterisk box. There is no issue related to your
account.
2
Hi Giorgio,
Do you have any log showing some error?
Did you already have a look at SIP connection messages from and to this SIP
server? I suggest you to use wireshark to check sip messages.
Thanks,
Marino
On Mon, Jul 14, 2008 at 3:47 PM, Giorgio Incantalupo
[EMAIL PROTECTED] wrote:
Hi,
I
help about Asterisk configuration but
nothing...the same on the rest of internet.
Giorgio
map wrote:
Hi Giorgio,
Do you have any log showing some error?
Did you already have a look at SIP connection messages from and to
this SIP server? I suggest you to use wireshark to check sip
Hi Paolo,
Unfortunately there are not over the counter solutions.
I used OpenSer in order to add scalability (point 1) and a custom web
application for point 2.
Map
On Mon, Feb 11, 2008 at 3:16 PM, Paolo Losi [EMAIL PROTECTED] wrote:
Hi all,
we are considering Asterisk as a possible solution
Hi Alberto,
I think that here you can find useful hw:
http://www.patapsco.co.uk/
Marino
On Jan 23, 2008 9:39 AM, Alberto Pastore [EMAIL PROTECTED] wrote:
Hi everybody.
I know maybe this question has been posted some time ago, but
I need your updated opinion on the subject.
I'm replacing
Hi Daniele,
Could you please tell us what exactly happens?
Are your able to see some error in the log/console?
On Jan 7, 2008 11:53 AM, daniele visaggio [EMAIL PROTECTED]
wrote:
Hi all!
Sorry for my poor english, i'm italian.
I installed Digium B410P on my asterisk server. I followed the
Daniele,
you need an external calls rule in your extension.conf, that is 1 to call
using PSTN line.
Please send your extension and we can take a look to find your problem.
p.s.
I'm Italian too.
On Jan 7, 2008 3:03 PM, daniele visaggio [EMAIL PROTECTED] wrote:
Hi Daniele,
Could you please
Hi Daniele,
Please send a snapshot of your Putty Asterisk log.
Go to Putty configuration - Window - Lines of scrollback and put a number
greater than 200 :-). I suggest 10.
On Jan 7, 2008 4:00 PM, daniele visaggio [EMAIL PROTECTED] wrote:
2008/1/7, map [EMAIL PROTECTED]:
Daniele,
you
Hi,
This procedure should work whether you have remove some VoIP card from
your VoIP box.
Anyway be careful
On Nov 29, 2007 11:14 AM, Sasa [EMAIL PROTECTED] wrote:
Hi, my problem isn't on new voip box with latest asterisk version...my
problem is on voip with Asterisk 1.2.13 where I must
Hi Alberto,
I think that meetme and/or park should work.
We done something like this suing queue as well.
Could you please let us know why task #6 does not work in your case.
Map
On Nov 23, 2007 8:38 AM, Alberto Pastore [EMAIL PROTECTED] wrote:
Hi everybody.
I am in the following scenario
Hi Fabio,
Once you have an Asterisk box that have a conference room configured and a
VoIP phone the supports forward you can easily forward your guests to the
conference room.
Moreover you can create a conference room extension available, via password,
from the PSTN line.
Hope this can help
Hi Alex,
You should create a dial plan to route sip calls to H.323 calls.
Take a look at :
http://www.voip-info.org/wiki/
On 8/6/07, Alessandro Russo [EMAIL PROTECTED] wrote:
Hi to all,
I've installed Asterisk 1.4.9 with h323, and gnugk as soft gatekeeper.
I've tested h323 using ohphone
reading more about Dialplan, but until now, I've not found
anything...(like example or tutorial)
With the word route you are intending the Goto command??
Please spent some minutes for helping me ^_^
If you are agree, I send you some information about configuration files.
Thx
On 8/6/07, map
Hi think that once SIP/SDP invite/reinvite is sent you can not change to
video stream.
On 7/10/07, Jerry Geis [EMAIL PROTECTED] wrote:
I have 3 phones
P1 is a non video phone - grandstream
P2 is a Grandstream GXV3000
P3 is a Grandstream GXV3000
Using P1 to place a call to P2 I get audio
Linksys SPAs work well with Asterisk
On 4/13/07, Luca Corti [EMAIL PROTECTED] wrote:
On Thu, 2007-04-12 at 14:47 -0400, J. Oquendo wrote:
1) Snom
2) none! (they're all pretty much the same to me)
3) none! (they all have their pros and cons)
4) Cisco
5) ASStra
6) Polycrud
You haven't
Hi,
Could you please explain what your provider is expecting?
You should only have to provide your public IP address.
On 4/4/07, Il Neofita [EMAIL PROTECTED] wrote:
Hi,
I am using my asterisk server like a gateway and one provider ask me to
pass an extra field with the IP of the peer that is
If you're in the same lan, I think taht you have some problem with the
codec.
Doulbe check codecs config both in xlite and in sip.conf.
On 3/9/07, Asterisk Asterisk [EMAIL PROTECTED] wrote:
Hey,
I am a new to asterisk and softphones. Ihave recently
installed and configured linux and 2 xlite
Hi,
It seems that you are using different audio codec (Unknown RTP codec
96 received)
Try to use standard audio code. Sometimes telephone use codec with bad
rtp code inside. I use alw or ulaw for my test.
Marino
On 7/11/05, Ronald_Wiplinger [EMAIL PROTECTED] wrote:
Giorgio Incantalupo wrote:
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