Hi,
In order to save public IPs, I am attempting to use a Router SureConnect of US
Robotics in order to route calls to Asterisk on a private IP.
Would you recommand a large router like Cisco if we have 30 calls or a normal
router can do ?
Any advise is greatly appreciated
Etienne,
I am not sure I understand all what you require. Do you need to know the cost of
the voiceblue of 2N or you need to find solution that can allow you send GSM
calls ?
There are several alternatives:
1-) Voiceblue as you mentioned;
2-) You can buy a voip2GSM Gateway. To which you no
Hi,
That is interesting. What is the make and the model that you are
referring to? Is there a website with more info?
As for the models, we sell them as OEM. You may contact me offlist if
interested. Better priced and more powerful than existing devises out there.
I currently use
Hi,
Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs ,
one with 15 channels and the other with 15 channels;
Is there a sort of E1 multiplexer devise that allows me to plug in one hand the
E1 port of the Digium card and on the other hand the two PABXs? In this same
I just called this company. They seem to do what is required. Now remains the
pricing part of it. I will wait for their feedback.
http://www.megatelindustries.com/products.htm
Hakem,
Selon Julio Arruda [EMAIL PROTECTED]:
Matteo Brancaleoni wrote:
yes, some multiplexer allows that, but
Can Offer Hong Kong,
10 USD/DID/month . Set-up cost 20 USD. Minimum committment is 06 months.
Regards,
Hakem,
Selon Marc Storck [EMAIL PROTECTED]:
I'm also looking for numbers from
HongKong,
Taiwan,
Japan and
Singapore
So if someone has some DIDs from this areas, I'm very interested
Hi,
I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The
problem now comes in the PCI ports. Is there any PC that can handle 16 ports?
What is most optimal solution?
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Asterisk-Users mailing list
Thanks very much for this info Andrew.
Selon Andrew Kohlsmith [EMAIL PROTECTED]:
On April 25, 2005 07:39 am, [EMAIL PROTECTED] wrote:
I would like to setup a 8 FXS + 8 FXO interface using the X100P Cards. The
problem now comes in the PCI ports. Is there any PC that can handle 16
ports?
Hello,
I am stuck while attempting to insert DTMF commands from a SIP gateway to an IVR
menu running AGI php scrit.
If I do:
SIP phone -- ulaw -- IVR then dtmf works fine
If I do:
SIp gateway -- g729 -- IVR then the mneu still works but does not accept DTMF
?
I have tried to set
Selon Rich Adamson [EMAIL PROTECTED]:
Funny, they sell these old cards.. it seems like they are selling
refurbs
as new.. ... anyways RMA is on its way, would be nice if they would
send
one as a replacement first, so that we could continue our work and
don't
have to delay it.
Hello,
I have an AGI script that runs a menu at two levels of a tree.
If I call the extension from a voip phone with g711, the menu works fine and
accepts DTMF no probs.
Then, when I Call from a DID, it sends call using SIP and g729 to¨* box.
The IVR also starts running, but no DTMF is
Quintum are good
Selon Chad Brown [EMAIL PROTECTED]:
There are many analogue gateways to choose from:
http://www.voip-info.org/wiki-VoIP+Gateways
Does anyone have experience with several that could point me in the
right direction? I need 5-8 ports. At some point I see us going digital
but
Then,
I realised a spent lot of time thinkin about this solution. Other option is that
you put a prepaid calling card platform in Russia. I saw in CEBIT some russian
companies selling prepaid calling cards.
In order to give access to your customers without them to know where is the
platform, you
Maybe following options:
1-) Get another channel bank from ebay at low cost. Which will also need another
T1 card;
2-) Use 40 voip phones at 50 USD each and you no longer need the card neither
the channel bank. But a reliable local network ;
Selon Peter Hoppe [EMAIL PROTECTED]:
Hello,
I am
Ronald,
What are other solutions available that can offer web based access and work with
Asterisk?
Selon Ronald Wiplinger [EMAIL PROTECTED]:
Chris Mason (Lists) wrote:
Call Accounting is such an important issue for me it is literally a make or
break component, without it I will not be able
If you will donate some money to AreskiCC prepaid platform developer, he may
want to extend module to pure accounting.
I will also donate some money to a linux - web based accounting system
Selon Chris Mason (Lists) [EMAIL PROTECTED]:
There are other accountings available, and a Windows
You can use a password protected DISA functionality.
Selon Bill Ford [EMAIL PROTECTED]:
I'd like to see what some of you are doing to reliably aess
voicemail from an outside line.
Thanks
Bill
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Asterisk-Users mailing list
Matteo,
I don't know much about DIgium, but I am comparing the distribution policy with
what exists elsewhere in the market and other sectors.
Digium do sell online and so many other of their resellers do. The important
point is that they don't sell lower cost than their resellers, which is the
Yes,
Most hardware manufacturers I know sell directly at retail price.
In voip Business and from my experience, you can order Quintum gateways from
Quintum Technologies right away at retail price. You can always get them
cheaper from reseller. GSM devices manufacturers sell direct as well
Do you want to make GSM calls? Or just use PCMCA card for the data connection?
Inb case you want to make GSM calls with SIM cards, you need to get GSM Gateways
or called sometimes fixed-cellular terminals and plug them to a FXO card. Or
you can get a voip-GSM box.
Selon Jan Kellerhoff [EMAIL
Hello,
I was just wondering if there were a prize like the open source application of
the year relative to Asterisk?
All these developer doing good job and all free need some present sometime that
we can all donate.
Anything like that exists?
___
It seems like Digium is good for beginners and Sangoma is good for experts in a
nutshell.
What I need to know is whether I can offer Internet Dial-up Access using Sangoma
hardware or Digium hardware?
This is because pricing per E1 is attractive compared to existing proprietary
solutions
Has
Johannes,
I would be curious to know if there is a solution for this. Another solution is
that you buy a call meter. Which is a small box that can be placed in front
of phone phone and that can display costs.
FXS-- call meter -- analog phone
This call meter needs to be programmed with a table
If it helps, then great !
This is what we do in Africa for some callshops that do not want to pay $$$ in
billing software licenses.
While talking, I wonder whether the field of CAller ID which is displayed in
the IP Phone can be updated while conversation is ongoing, say every 10
seconds. In
Let me ask a silly question,
Is Cisco call manager free software ?
Selon Nathan Alberti [EMAIL PROTECTED]:
I did some work on my configuration to get it working and have now
documented it somewhat here:
http://voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Express+Integration
Are the h323 devices on public IP or behind NAT?
Selon Shaoul Jacobson - TELLINK [EMAIL PROTECTED]:
Hi,
I am using real-time, oh-0.7.2, G729
Calling from (SIP)UA through asterisk towards h323 devices or the other way
round, I get only one-way audio.
Called party can only talk, caller
What about pricing of the Sangoma compared to Digium, is it comparable?
Can Sangoma card handle modem data incoming calls at all?
Selon mattf [EMAIL PROTECTED]:
Hello,
I need to correct myself on one of the points I made in my reply last night.
As a very polite developer from Sangoma
Hello,
I have very few knowledge of this area. But mobile operator in belgium has
launched Hello Tone. So if you call somebody on mobile network, you can
listen to U2 music, or Bethoven (chosen by callee) while the phone is ringing,
even if you call from different network operator.
Before I see
Hi,
I don't know much about Orion brands. In Europe, most robust brands are:
www.2n.cz and www.vierling.de
Vierling is richest but most expensive. It does SMS callback, sms2mail,
mail2sms.
Regards,
Selon cmould [EMAIL PROTECTED]:
My client is looking fro a GSM gateway (24 ports). Any
Did you install libtiff libraries prerequisites before compiling
It may be an issue with your LD_LIBRARY_PATH as files do not seem to be found.
Selon Dennie Verstrepen [EMAIL PROTECTED]:
Hello everybody,
I'm trying to receive and sending faxes with asterisk using spandsp. But
while
This is possible. But success depends also on whether the router can do port
forwarding and whether the H323 Gateway supports NAT.
This is possible with Quintum for instance with some port forwarding rules on
router level.
Selon VoIP Newbie [EMAIL PROTECTED]:
Hi all,
I am wondering if
You can try AreskiCC at areski.net
Selon chawki hammoud [EMAIL PROTECTED]:
I appreciate any recomendation of a simple CallingCard
Application and resources of users manual.
__
Do you Yahoo!?
Yahoo! Small Business - Try our new resources site!
For this,
it works randomly. I have decided to work on SIP and forgot about h323 with
Asterisk. I spent nights and nights trying to figure out how it works, but
decided to move on.
Now we are running SIP, things are better.
Selon Chetan Sarva [EMAIL PROTECTED]:
Did you ever find a solution
Thanks Kevin for this info,
If we want a box that can perform 60 calls. What would be apoproximate budget
for that using AMD x86-64 ?
µSelon Kevin P. Fleming [EMAIL PROTECTED]:
Erick Perez wrote:
And what people are using to deploy super servers with astersik?
Itanium with linux? clusters
Another option is to send sms by mail.
1-) You subscribe to an sms provider who can allow you to do mail2sms;
2-) You send sms message under the form [EMAIL PROTECTED] ;
3-) SMS provider receives SMS from you and will send it through its gateway;
Hope this helps
Quoting Marco Ziglioli [EMAIL
This conversation is interesting,
What about a driver that will send the print out to Asterisk, on the same
network to be sent as Fax ?
Is there anything that already exists for this?
Quoting Florian Overkamp [EMAIL PROTECTED]:
Hi,
-Original Message-
You should be able to download one (for
Hello,
I am using asterisk for two stage dialing:
1-) I make a call from my voip phone;
2-) Asterisk dials first a access number, iputs a PIN and then dials destination
number;
However, I am getting the connect signal from the moment access number connects.
I would like to avoid receiving this
Check:
http://lists.digium.com/pipermail/asterisk-users/2003-October/025094.html
What you need is convert the document you need to send into a tiff file
and put
it in a directory. Which I believe is functionality you want to achieve.
Selon Justin Newman [EMAIL PROTECTED]:
We have something on
All,
I have tried very hard to make asterisk work with h323 but still strying:
I have been successful making this work
SIP -- Asterisk -- h323 -- termination ;
But the following:
h323 -- asterisk -- h323 -- Termination : works , call set up is ok but then
no audio is applied .There is no NAT
Hello All
I am looking for a solution that can do this:
1-) Receive incoming fax;
2-) Read content and identify a zone in the fax where there is a hand written
name;
3-) Based on name, query a database;
4-) Act based on the result in the database;
I understand asterisk can receive fax and
I'm not sure it answers all your questions but there is ast-stats from
http://areski.net/areski/index.php?
option=com_contenttask=categorysectionid=5id=70Itemid=54
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Asterisk-Users@lists.digium.com
Hello,
I am attempting to use Asterisk as a protocol converter.
I have set up asterisk to route incoming h323 calls to a SIP termination
carrier.
I make a test, call is coming correctly, is rerouted to termination carrier.
Call connects and phone rings. Then, I pick up the phone and it
Hello,
We had same problem in other african country. We could resolve it through using
IAX Bridge in Asterisk since it only uses one port of yoru choice.
For your solution, you need:
1-) Scan outgoing / incoming open ports by your ISP;
2-) If there remains many open ports, you may still run
Hello,
I am moving topic from asterisk-dev list to asterisk-users list. Did anyone
succeed receive incoming calls in h323 and orient them to right context based
on host identification?
To summarise, I have quintum Gateway sending call to Asterisk box, and I would
like to use asterisk as a
Good to hear I am not alone.
Actually, I am using the Nufone's h323 module. Still this creates the problem. I
had a braod look at the code and it seems that it is not possible that incoming
calls go to other places than general context (I am not sure I understood it
all, but almost).
So, one
Soren,
I tried the variable UserByAlias=no and it worked for me. Thank you very much
for this note !
Selon Soren Rathje [EMAIL PROTECTED]:
[EMAIL PROTECTED] wrote:
Good to hear I am not alone.
Actually, I am using the Nufone's h323 module. Still this creates the
problem. I had a braod look at the
Thanks Matteo,
Now I have a backtrace if that will help. I am not a programmer and this
really means nothing to me. I can only tell you that I have a g723.1 encoded
file (conf-onlyperson.g723) in /var/lib/asterisk/sounds/ when this happens.
#0 0x08058291 in ast_write (chan=0x8111718,
Hi,
First of all I am not sure that what I am trying to do is correct/supported,
but here is what I'm trying to test:
Some of my endpoints only have g723 codecs. Because of this I am only
allowing g723.1 codec in sip.conf and h323.conf. Calls between endpoints
work fine. I am trying to configure
Hi,
I want to configure * to use a gatekeeper for routing calls to H.323
endpoints. I imagine it will work like that:
* (chan_h323) will query the gatekeeper where to terminate the dialed number
and the gatekeeper will return the information for the h.323 gateway. after
that chan_h323 will try to
Hi,
I think I understood how to achieve this. Anyway, a working config is
welcome if anyone has already done it.
hristo
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of HT
Sent: Wednesday, July 09, 2003 2:54 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk
Thank you for the answer.
Anyone working on that?
I am trying in the meantime to disable the RTCP reports on the gateways,
hoping that it will work like that.
hristo
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent: Saturday, July 05,
Hi,
I am plying with * for 10
days now. I am testing with a couple of vocaltec h.323 gateways (FXO and PRI) cisco
ata-186 (configured for SIP) and MSN messenger (SIP). They all seem to
interoperate. However I have a problem when * is sending calls to the vocaltec
gateways. Vocaltec
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