HI,
can somebody tell me how and where must I put the SIP register line? I
think is in [general] section of the sip.conf and that I have to put:
register = user:[EMAIL PROTECTED]:port/localextension
but, user and password of the SIP gateway? Because I'm trying this and
doesn't work...
thanks
Hi eveybody again!
I don't want to be annoying, but if nobody can help me with this, I'll have to
desist of working with SIP.I have some questions about SIP, as I wrote in
another mail. I have a SIP Gateway and I have two phones (an analog one
and a DECT one) conected to it.Also, I have two
, why can I hear
the callee phone ringing and the call only goes off when I pick it up?
it's so strange...I think!
Michelle
gt;On Mon, 16 Jun 2003, michelle matis litio wrote: gt;gt; to
229.159.241.112:5060 gt;gt; Retransmitting #5 (no NAT): gt;gt; SIP/2.0
200 OK gt;gt; Via: SIP/2.0/UDP
Hi!
I have a new problem with my SIP device.I have done some changes and
now I receive continuosly a SIP message: 501 Not impelmented back
from the SIP Gateway. I can see that it doesn't register to Asterisk.
I have in the SIP device:
Registrar 1:UnRegisteredto:
registrar:
]
[mailto:asterisk-users-
[EMAIL PROTECTED]');[EMAIL PROTECTED]');[mailto:asterisk-
[EMAIL PROTECTED]
On Behalf Of michelle matis litio Sent: Wednesday, June 11,
2003 12:12 PM To: asterisk-
[EMAIL PROTECTED]');[EMAIL PROTECTED]');asterisk-
[EMAIL PROTECTED]
Subject: [Asterisk-Users] Re:Some SIP questions
Hi everybody one more time!
I also have done a SIP debug and that's an extract of what I have found:
(...)
s=session
c=IN IP4 188.208.12.237
t=0 0
=audio 13532 RTP/AVP 0
a=rtpmap:0 PCMU/8000
to 229.159.241.112:5060
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
Hi everybody
I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me!
1) Which codecs may I use? I want the SIP phones to call to the PSTN above all, but I have two dlink dg102s (MGCP) and I'd like to can call them too. The problem is that
I write the email again, cause the first one I have had problems while sending it. Here is the email again:
Hi everybody,
I'm starting with SIP and I wanted to ask some questions, perhaps silly ones, but I hope people can answer me!
1) Which codecs may I use? I want the SIP phones to call to the
I write the email again, the third time!!, cause the other two ones, I have
had problems while sending them. I hope this time it works. Here is the
email again:
Hi (and sorry) everybody
I'm starting with SIP and I wanted to ask some questions, perhaps silly
ones, but I hope people can answer
Hi Edwin
I have my mgcp.conf almost the same as yours, except from nat=1 , why
do you put it?
Anyway, DL102s now works more or less acceptably so now I'm having a
battle with sip.conf
Thank you for your help
Michelle
-
Tu cuenta de correo gratuita Mixmail http://mixmail.ya.com
Ya.com
Please I need help, I don't know why,almost every time I dial on my dect
phones, the dialtone doesn't go off and * doesn't recognise anything I'm
using two dlink voip gateways, MGCP: DL102s. Any ideas?
thanks in advance
michelle matis
-
Tu cuenta de correo gratuita Mixmail
I'm using * as a Call Agent for two DL102S but I have some problems, like the tones not being sending from the phone to the *. I have not changed the configuration of the DL, except the IP and the Notify Entity (*). Must I change another thing in * or in the device? Thanks very much michelle
Tu
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