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

[michelle matis litio]

Hi!
I thought it was the SIP device too, but I have looked for avery litle comand 
of this device and I can't find this Ip address, and I see that its Ip is Ok, 
and I have configurated the REGISTRAR section too... I don't know what's 
happening, and I don't understand that, if the IP is wrong, 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




>On Mon, 16 Jun 2003, michelle matis litio wrote: >> to 
229.159.241.112:5060 >> Retransmitting #5 (no NAT): >> SIP/2.0 
200 OK >> Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK-
3a5246f7- >> 8c6b606-10eb >> From: ;tag=0-13c4-3a5246f7-
8c6b604-c3a &gt;&gt; To: ;tag=as52ed0a6a &gt;&gt; Call-ID: <A 
href="javascript:sendMsg('f93b00-0-13c4-3a5246f7-8c6b602-
3765 at 188.208.12.37');">f93b00-0-13c4-3a5246f7-8c6b602-
3765 at 188.208.12.37</A> &gt;&gt; CSeq: 1 INVITE &gt;&gt; User-Agent: 
Asterisk PBX &gt;&gt; Contact: &gt;&gt; Content-Type: application/sdp 
&gt;&gt; Content-Length: 135 &gt;&gt; &gt;&gt; v=0 &gt;&gt; o=root 11673 
11673 IN IP4 188.208.12.237 &gt;&gt; s=session &gt;&gt; c=IN IP4 
188.208.12.237 &gt;&gt; t=0 0 &gt;&gt; =audio 13532 RTP/AVP 0 &gt;&gt; 
a=rtpmap:0 PCMU/8000 &gt;Hi, &gt;Its being sent to that IP address, 
because that is that the &gt;originating SIP device put in its Via header. 
&gt;Also, your SIP device didn't put any From or To in its INVITE. 
&gt;Perhaps you could send a sip debug from the start of a SIP call 
&gt;attempt. &gt;But I'm sure that the trouble is with your SIP Gateway 
device's &gt;setup. &gt;Steve 
&gt;_______________________________________________ &gt;Asterisk-
Users mailing list &gt;<A href="javascript:sendMsg('Asterisk-
Users at lists.digium.com>http://lists.digium.com/mailman/listinfo/asterisk-
users');">Asterisk-Users at lists.digium.com 
&gt;http://lists.digium.com/mailman/listinfo/asterisk-users</A>

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