[Asterisk-Users] sip.conf

2003-06-17 Thread michelle matis litio
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

[Asterisk-Users] The same SIP problems...SORRY!

2003-06-16 Thread michelle matis litio
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

Re: Re: [Asterisk-Users] The same SIP problems...SORRY!

2003-06-16 Thread michelle matis litio
, 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

[Asterisk-Users] SIP REGISTER

2003-06-16 Thread michelle matis litio
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:

Re: RE: [Asterisk-Users] Re:Some SIP questions AGAIN

2003-06-12 Thread michelle matis litio
] [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

Re: RE: [Asterisk-Users] Re:Some SIP questions AGAIN

2003-06-12 Thread michelle matis litio
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

[Asterisk-Users] (no subject)

2003-06-11 Thread michelle matis litio
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

[Asterisk-Users] some sip questions

2003-06-11 Thread michelle matis litio
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

[Asterisk-Users] some sip questions AGAIN

2003-06-11 Thread michelle matis litio
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

[Asterisk-Users] Re:Some SIP questions AGAIN

2003-06-11 Thread michelle matis litio
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

[Asterisk-Users] dl102s again

2003-06-06 Thread michelle matis litio
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

[Asterisk-Users] dl102S

2003-06-05 Thread michelle matis litio
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