Hi!
I am looking for a software that can work as h.323 - sip gateway other than
asterisk and free. Someone can help me?
Thanks.
Mireia
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`t know exactly
how to do it. Someone can help?
Quoting Michael Manousos <[EMAIL PROTECTED]>:
>
> Hi,
>
> Mireia Munoz de jesus wrote:
> > Hi!
> >
> > I have a little big problem here. I have an gateway(asterisk,working as a
> H.323
> > - SIP gat
Hi!
I have a little big problem here. I have an gateway(asterisk,working as a H.323
- SIP gateway) conected to a gatekeeper (two different servers), and also a
gateway (cisco - PSTN) conected to the same gatekeeper. When I make a call from
the gateway(cisco) to a sip phone, the phone rings, but wh
ateway are compatible with the codecs in
> asterisk.
> What are the codecs you are using in SIP Phones, in Asterisk and in the
> gateway?
>
> Regards,
>
> Vinicius
>
>
>
> -Mensagem original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] nom
is in the configuration of your gateway
>
> Other thing you can see is if your asterisk box is registered with your
> gatekeeper.
>
> With the information you supplied this is what I remember you can check to
> see what is wrong.
>
> Regards,
>
> Vinicius
>
&
?
Quoting Martin Mielke <[EMAIL PROTECTED]>:
> Hi Mieria,
>
> Mireia Munoz de jesus wrote:
>
> >Hi!
> >
> >When I try to call from a SIP phone to a PBX phone I get this error:
> >
> >chan_oh323.c [1004] Couldn`t call 483377839
> >
>
Hi!
When I try to call from a SIP phone to a PBX phone I get this error:
chan_oh323.c [1004] Couldn`t call 483377839
and if I get the messages from SIP debug, I have a 403 message. The
configuration of my system is:
SIP Phone ASterisk Gatekeeper - Gateway - PBX - Phone
H
Hello all!
I am tryng to call from a Grandstream to a normal telephone. The net
configuration is:
Phone --- PBX --- Proxy Gatekeeper Asterisk Grandstream
When I try I call from a h.323 phone to the normal phone I have no problem. But
when I try to call from Grandstrem to the pho
Hi!
The line with ;sock=/tmp/mysql.sock, i think you must write it without the ";".
You need this socket to connect with mysql.
Best regards,
Mireia
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of listas
> iPfone
> Sent: sexta-feira, 12 de dezem
Hi!
I am using X-Lite and NetMeeting.
When I call from netmeeting to X-lite or from X-lite to netmeeting, the call is
stablished correctly. But after some seconds, they hang up. I get some errors
messages:
- 7:38.574 H245:8122c90 H245Read error: Bad file
descriptor
- ERROR
Hi list!
I have done a H.323 - SIP Gateway. The SIP Register is within Asterisk, so all
the SIP phones are known and are all registed in Asterisk. Now it works ok,
because there are no so many phones connected, about four, but I would like to
know if when the SIP network would be bigger, if there
Hi!
I am trying to know well asterisk. For that I would like to know the exact role
for each config file. Can someone tell me what is the role of the next ones or
a web where I could find this information? That will be very helpful.
- alsa.conf
- enum.conf
- modem.conf
- modules.conf
- oss.conf:
Hi!
I have been looking for a while for informatoin about how QoS is assured in
Asterisk, but I haven't found a thing. Can someone give me some tips about
that?
Thanks,
Best regards,
Mireia
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Hi!
I have a technical question. How does asterisk control the Qos? How does it
works?
Thanks a lot,
Regards,
Mireia
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Hi all!
I am looking for some free software to monitoring all the calls that are being
done in my network. Which telephone are connected, how long are the calls,
quality of service, bandwidht,etc.
If someone knows about a good one, plesea tell me.
Regards,
Mireia
Hi!
I am doing a research about the prices of SIP telephones. If someone can tell me
which one are the cheapest and have an acceptable quality... it will be very
kind.
Best Regards,
Mireia
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Hi!
I have three questions:
- I have called from an H.323 softphone to a SIP one, and then I have tryied to
transfer the call to be accepted by asterisk. And it has not work. Is it
possible to do that? And if it is possible, what I have to do for that works?
- In extensions.conf, there's the va
sday, October 14, 2003 12:23 PM
> Subject: Re: [Asterisk-Users] H.323 - SIP gateway
>
>
> > h323 runs on port 1720. Your gatekeeper is trying to contact the wrong
> > port number.
> >
> > On Tue, 2003-10-14 at 10:02, Mireia Munoz de jesus wrote:
> > >
Hi!
I have some problems when the gatekeeper tries to contact my asterisk at the
1719 port (port for RAS communication). The problem is that this port is not
open on my machine. Is that a problem with asterisk configuration? If it is,
where can I configure that? I was looking at oh323.conf but the
Hi all!
Please I need someone that have already done an H.323 - SIP gateway to help me
with some problems. I can stablish calls from a SIP telephone to a H.323, but I
can't do vice versa... (problems with port 1719- when the gatekeeper tries to
contact with asterisk at this port, it is unrecheabl
Hi!
I have configured my gatekeeper to call asterisk everytime that the phone number
begins with 064... When the gatekeeper contacts asterisk, it does it using the
1719 port, but it is closed.
How can I open this port? Or the solution is to redirect the messages arriving
at 1719 to 1720?
Thank
Hi!
I need to open both ports 1720 and 1719. How can I do that?
Thanks.
Regards,
Mireia
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Hi!
I am trying to do a SIP/H.323 gateway. I want that the SIP proxy server (I
suppose that this is asterisk isn't it?) has all the information about the
user's registration. So, when a request arrives at the gatekeeper from the
H.323 network, this one tries to make multicast to all the others ga
Quoting Anton Tinchev <[EMAIL PROTECTED]>:
> What gatekeeper do you use.
> It seems that is programed to make outgoing calls only to registered h.323
> users.
> Just program it to forward unknown number to the asteris (or switch
> everything to SIP)
>
I can't forward unknown number to the aster
Hi!
I am in a H.323 network with a gatekeeper and some terminals. Asterisk is a
gateway between this network and the SIP network. Now I can do calls from de
foreign network (SIP) to the locla (H.323) but I don't know how to do the
inverse. The H.323 terminals use NetMeeting, and when I try to mak
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