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In sip.conf :<BR>
<BR>
<FONT SIZE="2">[general]</FONT><BR>
<BR>
<FONT SIZE="2">canreinvite=no ; Asterisk by default tries to redirect the</FONT><BR>
<FONT SIZE="2"> ; RTP media stream (audio) to go directly from</FONT><BR>
<FONT SIZE="2"> ; the caller to the callee. Some devices do not</FONT><BR>
<FONT SIZE="2"> ; support this (especially if one of them is behind a NAT).</FONT><BR>
<FONT SIZE="2"> ; The default setting is YES. If you have all clients</FONT><BR>
<FONT SIZE="2"> ; behind a NAT, or for some other reason wants Asterisk to</FONT><BR>
<FONT SIZE="2"> ; stay in the audio path, you may want to turn this off.</FONT><BR>
<BR>
<FONT SIZE="2">[3starsnet]</FONT><BR>
<FONT SIZE="2">type=peer</FONT><BR>
<FONT SIZE="2">host=85.119.188.3</FONT><BR>
<FONT SIZE="2">username=username</FONT><BR>
<FONT SIZE="2">secret=password</FONT><BR>
<FONT SIZE="2">fromuser=username</FONT><BR>
<FONT SIZE="2">dtmfmode=rfc2833</FONT><BR>
<B><FONT SIZE="2">canreinvite=no</FONT></B><BR>
<FONT SIZE="2">insecure=port,invite</FONT><BR>
<FONT SIZE="2">qualify=yes</FONT><BR>
<FONT SIZE="2">nat=yes</FONT><BR>
<FONT SIZE="2">disallow=all</FONT><BR>
<FONT SIZE="2">allow=gsm</FONT><BR>
<FONT SIZE="2">allow=alaw</FONT><BR>
<FONT SIZE="2">amaflags=billing</FONT><BR>
<FONT SIZE="2">accountcode=3starsnet-out</FONT><BR>
<BR>
<BR>
<BR>
<BR>
On Tue, 2009-06-23 at 08:50 -0400, Steve Totaro wrote:<BR>
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On Tue, Jun 23, 2009 at 8:09 AM, jonas kellens <<A HREF="mailto:jonas.kellens@telenet.be">jonas.kellens@telenet.be</A>> wrote:
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<FONT COLOR="#0000ff"> -- Executing [0473775006@intern:1] NoOp("SIP/twinkle-088e6ea8", "conversation to GSM") in new stack</FONT><BR>
<FONT COLOR="#0000ff"> -- Executing [0473775006@intern:2] Dial("SIP/twinkle-088e6ea8", "SIP/3starsnet/0473775006") in new stack</FONT><BR>
<FONT COLOR="#0000ff"> -- Called 3starsnet/0473775006</FONT><BR>
<FONT COLOR="#0000ff"> -- Got SIP response 482 "Loop Detected" back from 85.119.188.3</FONT><BR>
<FONT COLOR="#0000ff"> -- Now forwarding SIP/twinkle-088e6ea8 to 'Local/0473775006@default' (thanks to SIP/3starsnet-088e2ca0)</FONT><BR>
<FONT COLOR="#0000ff">[Jun 23 07:29:43] NOTICE[15896]: chan_local.c:504 local_call: No such extension/context 0473775006@default while calling Local channel</FONT><BR>
<FONT COLOR="#0000ff">[Jun 23 07:29:43] NOTICE[15896]: app_dial.c:561 wait_for_answer: Failed to dial on local channel for call forward to 'Local'</FONT><BR>
<BR>
<BR>
Twinkle is my softphone, who is registered with my Asterisk-server.<BR>
<BR>
85.119.188.3 is the IP-address of the SIP-server of my VoIP-provider.<BR>
<BR>
SIP/3starsnet is the SIP-channel that corresponds with a peer defined in sip.conf to place calls to my SIP-provider 3StarsNet.<BR>
<BR>
So twinkle calls my Asterisk-server, which calls my SIP-provider...<BR>
<BR>
What is this "Loop detected" ??<BR>
<BR>
<FONT COLOR="#888888">Jonas.</FONT>
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I have seen this with canreinvite=yes set when it should not be. <BR>
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-- <BR>
Thanks,<BR>
Steve Totaro <BR>
+18887771888 (Toll Free)<BR>
+12409381212 (Cell)<BR>
+12024369784 (Skype)
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