[Asterisk-Users] SIP REGISTER
Felix Cortes
felix.cortes at ipfonica.com
Tue Jun 17 10:58:46 MST 2003
Félix Cortés
Engineering Manager
IPfónica
Monterrey N.L., México
felix.cortes at ipfonica.com
Tel. (52) 81- 8114-7170, 8114-7172
-----Mensaje original-----
De: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] En nombre de michelle
matis litio
Enviado el: Tuesday, June 17, 2003 2:32 AM
Para: jtodd at loligo.com; asterisk-users at lists.digium.com
Asunto: Re: Re: [Asterisk-Users] SIP REGISTER
I have a SIP Gateway with 2 phones, and a MGCP Gateway with other two.
I want all the phones to call to the PSTN and to call between them. MGCP
goes OK but SIP doesn't. I can call them but the can't call. I'm
starting to
think that is a problem of the SIP device, not an Asterisk problem.
michelle
>I'm afraid I have no idea what your goal is here. Do you have a
>phone somewhere in this configuration? I don't see it. Please
>explain what it is you are trying to do. From what I see (though
>much data is missing from your explanatin) anytime you place a call,
>it will result in a loop. >While you're at it, include the
following
information: >sip show peers >sip show registry >sip debug (and
wait for a cycle of SIP messages to go by) >JT >>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:
UnRegistered to: 2222 >>registrar: 188.208.12.237 5060 expires:
2000 >>name: gateway passwd: 123 >> >> >>My
sip.conf: >> >>[general] >>port = 5060
>>bindaddr =
0.0.0.0 >>context = default >>transfer = yes
>>threewaycalling = yes >>usecallerid = yes
>>hidecallerid = no >>register => <A
href="javascript:sendMsg
('gateway:123 at 188.208.12.37/2222');">gateway:123 at 188.208.12.37/2222<
/A> >> >>[gateway] >>type=friend
>>callerid="sip"
<2222> >>username=gateway >>host=188.208.12.37
>>secret=123 >> >>My extensions.conf >>
>>exten => <A href="javascript:sendMsg
('2222,1,dial,SIP/2222 at 188.208.12.37|60|rTt');">2222,1,dial,SIP/2222 at 188
.2
08.12.37|60|rTt</A> >>exten => 2222,2,Hangup >>
>>I'm
going crazy with this...I think that I'm not doing well the
>>registration
but I >>can't find why!! 188.208.12.237 is the IP of the asterisk
and
188.208.12.37 >>is the IP of the SIP gateway. 2222 is one of the
phones of the SIP >>Gateway...Anyone can help????Please!
>>Thanks very very much >>Michelle >> >>-----
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-----
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