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 >> To: ;tag=as52ed0a6a >> Call-ID: <A
href="javascript:sendMsg('f93b00-0-13c4-3a5246f7-8c6b602-
[EMAIL PROTECTED]');">f93b00-0-13c4-3a5246f7-8c6b602-
[EMAIL PROTECTED]</A> >> CSeq: 1 INVITE >> User-Agent:
Asterisk PBX >> Contact: >> Content-Type: application/sdp
>> Content-Length: 135 >> >> v=0 >> o=root 11673
11673 IN IP4 188.208.12.237 >> s=session >> c=IN IP4
188.208.12.237 >> t=0 0 >> =audio 13532 RTP/AVP 0 >>
a=rtpmap:0 PCMU/8000 >Hi, >Its being sent to that IP address,
because that is that the >originating SIP device put in its Via header.
>Also, your SIP device didn't put any From or To in its INVITE.
>Perhaps you could send a sip debug from the start of a SIP call
>attempt. >But I'm sure that the trouble is with your SIP Gateway
device's >setup. >Steve
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