[Asterisk-Users] HELP!!!!!!!! {Scanned}

[Julius Kidubuka]

When I do apply nat=no, the X-ten phones don't login at all!

> If your X-ten phones are on the same lan as asterisk then try nat=no.
>
> David
>
> On Thu, 2005-02-17 at 07:28 +0300, Julius Kidubuka wrote:
>> My sip.conf file;
>>
>> [luke]
>> type=friend
>> host=dynamic
>> username=luke
>> secret=luke
>> ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
>> dtmfmode=rfc2833
>> mailbox=202 ; Mailbox for message waiting indicator
>> allow=all
>> context=sip
>> callerid="luke" <2123>
>> nat=yes
>>
>>
>> [mike]
>> type=friend
>> host=dynamic
>> username=mike
>> secret=badru
>> ;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
>> dtmfmode=rfc2833
>> mailbox=203 ; Mailbox for message waiting indicator
>> context=sip
>> callerid="mike" <2125>
>> nat=yes
>>
>>
>> [juki]
>> type=friend
>> host=dynamic
>> username=juki
>> secret=juki
>> dtmfmode=rfc2833
>> mailbox=204 ; Mailbox for message waiting indicator
>> allow=all
>> context=sip
>> callerid="juki" <2125>
>> nat=yes
>>
>>
>> plus my extensions.conf file;
>>
>> exten => 202,1,Dial(SIP/luke,20,tr)
>> exten => 202,2,VoiceMail,u202
>> exten => 202,102,VoiceMail,b202
>> exten => 203,1,Dial(SIP/mike,20,tr)
>> exten => 203,2,VoiceMail,u203
>> exten => 203,102,VoiceMail,b203
>> exten => 204,1,Dial(SIP/juki,20,tr)
>> exten => 204,2,VoiceMail,u204
>> exten => 204,102,VoiceMail,b204
>>
>> Hope this provides a little bit more info.
>>
>>
>> > I new to this as will. But add more info like your sip.conf file.
>> >
>> > David
>> >
>> >
>> > On Wed, 2005-02-16 at 18:04 +0300, Julius Kidubuka wrote:
>> >> Hi,
>> >>
>> >> I have installed two X-Lite phones and they’re able to login
>> >> successfully. The two phones plus the Asterisk system are all on the
>> >> same LAN with private addresses assigned to each of them.  When a
>> call
>> >> is initiated and is picked up on the other end, there is completely
>> no
>> >> sound at all (as in the line goes dead). The codecs set in the
>> >> softphones are g711u, g711a, GSM, iLBC and SPX.
>> >>
>> >> From the Asterisk CLI I see the following errors;
>> >>
>> >> i)                    Unknown RTP codec 72 received
>> >>
>> >> ii)                  RFC3389 support incomplete
>> >>
>> >> Anyone got ideas on how I can go about this?
>> >>
>> >> Thanks in advance.
>> >>
>> >> Julius Kidubuka
>> >>
>> >> "When you do the common things in life in an uncommon way, you will
>> >> command the attention of the world"
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> --
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>> > --
>> > David Shaw <asterisk at ke6upi.com>
>> >
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>>
>>
>> --
>> Rgds,
>> Julius Kidubuka.
>> "My advice to you is get married: if you find a good wife you'll be
>> happy;
>> if not, you'll become a philosopher."
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>> Spam detection software, running on the system "zeus.avanzada7.com", has
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>> Content preview:  My sip.conf file; [luke] type=friend host=dynamic
>>   username=luke secret=luke ;dtmfmode=rfc2833 ; Choices are inband,
>>   rfc2833, or info dtmfmode=rfc2833 mailbox=202 ; Mailbox for message
>>   waiting indicator allow=all context=sip callerid="luke" <2123> nat=yes
>>   [...]
>>
>> Content analysis details:   (0.5 points, 5.0 required)
>>
>>  pts rule name              description
>> ---- ----------------------
>> --------------------------------------------------
>>  0.1 FORGED_RCVD_HELO       Received: contains a forged HELO
>>  0.4 PLING_PLING            Subject has lots of exclamation marks
>>
>>
> --
> David Shaw <asterisk at ke6upi.com>
>
>


-- 
Rgds,
Julius Kidubuka.
"My advice to you is get married: if you find a good wife you'll be happy;
if not, you'll become a philosopher."