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<TITLE>Outgoing SIP to SER causes LOOP BACK message</TITLE>
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<P><FONT SIZE=2 FACE="Arial">Hello fellow asterisk people!</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">I have Asterisk listening on port 5061 and SER on port 5060.</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">Asterisk is configured as a gateway for ISDN/Analog/H323 and also SIP.</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">My problems are with SIP. I can make incoming calls from SIP to asterisk and to any of the other networks, but when I try to make an outgoing call from Asterisk to SER I see the following in Asterisk:</FONT></P>
<P><B><I><FONT COLOR="#0000FF" SIZE=2 FACE="Arial"></FONT></I></B><I></I> <FONT COLOR="#0000FF" SIZE=2 FACE="Arial"> sip debug</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">SIP Debugging Enabled</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial"> -- Executing Ringing("H323/ip$192.219.85.57:2488/23038", "") in new stack</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial"> -- Executing Dial("H323/ip$192.219.85.57:2488/23038", "sip/290@sip_proxy-out|20|r") in new stack</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">We're at 192.219.85.57 port 15916</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Answering/Requesting with root capability 0x4 (ulaw)</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">12 headers, 8 lines</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Reliably Transmitting:</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">INVITE sip:290@fedcore2.eicon.com SIP/2.0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">From: "223" <sip:Asterisk@192.219.85.57:5061>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">To: <sip:290@fedcore2.eicon.com></FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Contact: <sip:Asterisk@192.219.85.57:5061></FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118@192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">CSeq: 102 INVITE</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">User-Agent: Asterisk PBX</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Date: Mon, 25 Jul 2005 08:51:42 GMT</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Content-Type: application/sdp</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Content-Length: 162</FONT>
</P>
<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">v=0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">o=root 17396 17396 IN IP4 192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">s=session</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">c=IN IP4 192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">t=0 0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">m=audio 15916 RTP/AVP 0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">a=rtpmap:0 PCMU/8000</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">a=silenceSupp:off - - - -</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial"> (no NAT) to 192.219.85.57:5060</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial"> -- Called 290@sip_proxy-out</FONT>
</P>
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<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Sip read:</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">SIP/2.0 100 trying -- your call is important to us</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">From: "223" <sip:Asterisk@192.219.85.57:5061>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">To: <sip:290@fedcore2.eicon.com></FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118@192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">CSeq: 102 INVITE</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Server: Sip EXpress router (0.9.3 (i386/linux))</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Content-Length: 0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Warning: 392 192.219.85.57:5060 "Noisy feedback tells: pid=30409 req_src_ip=192.219.85.57 req_src_port=5061 in_uri=sip:290@fedcore2.eicon.com out_uri=sip:290@192.219.85.57:5061 via_cnt==1"</FONT></P>
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<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">9 headers, 0 lines</FONT>
</P>
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<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Sip read:</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">INVITE sip:290@192.219.85.57:5061 SIP/2.0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Max-Forwards: 10</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Record-Route: <sip:192.219.85.57;ftag=as533da407;lr=on></FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57;branch=z9hG4bK1e89.44686dc.0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">From: "223" <sip:Asterisk@192.219.85.57:5061>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">To: <sip:290@fedcore2.eicon.com></FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Contact: <sip:Asterisk@192.219.85.57:5061></FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118@192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">CSeq: 102 INVITE</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">User-Agent: Asterisk PBX</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Date: Mon, 25 Jul 2005 08:51:42 GMT</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Content-Type: application/sdp</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Content-Length: 162</FONT>
</P>
<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">v=0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">o=root 17396 17396 IN IP4 192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">s=session</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">c=IN IP4 192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">t=0 0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">m=audio 15916 RTP/AVP 0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">a=rtpmap:0 PCMU/8000</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">a=silenceSupp:off - - - -</FONT>
</P>
<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">15 headers, 8 lines</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Transmitting (no NAT):</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">SIP/2.0 482 Loop Detected</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57;branch=z9hG4bK1e89.44686dc.0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">From: "223" <sip:Asterisk@192.219.85.57:5061>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">To: <sip:290@fedcore2.eicon.com>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118@192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">CSeq: 102 INVITE</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">User-Agent: Asterisk PBX</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Contact: <sip:Asterisk@192.219.85.57:5061></FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Content-Length: 0</FONT>
</P>
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<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial"> to 192.219.85.57:5060</FONT>
</P>
<BR>
<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Sip read:</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">ACK sip:290@192.219.85.57:5061 SIP/2.0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57;branch=z9hG4bK1e89.44686dc.0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">From: "223" <sip:Asterisk@192.219.85.57:5061>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118@192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">To: <sip:290@fedcore2.eicon.com>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">CSeq: 102 ACK</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">User-Agent: Sip EXpress router(0.9.3 (i386/linux))</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Content-Length: 0</FONT>
</P>
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<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">8 headers, 0 lines</FONT>
</P>
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<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Sip read:</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">SIP/2.0 482 Loop Detected</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">From: "223" <sip:Asterisk@192.219.85.57:5061>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">To: <sip:290@fedcore2.eicon.com>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118@192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">CSeq: 102 INVITE</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">User-Agent: Asterisk PBX</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Contact: <sip:Asterisk@192.219.85.57:5061></FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Content-Length: 0</FONT>
</P>
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<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">10 headers, 0 lines</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial"> -- Got SIP response 482 "Loop Detected" back from 192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Transmitting:</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">ACK sip:290@fedcore2.eicon.com SIP/2.0</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK62838805</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">From: "223" <sip:Asterisk@192.219.85.57:5061>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">To: <sip:290@fedcore2.eicon.com>;tag=as533da407</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Contact: <sip:Asterisk@192.219.85.57:5061></FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Call-ID: 0d2a1020378cbc7f4ee8541d11fb2118@192.219.85.57</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">CSeq: 102 ACK</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">User-Agent: Asterisk PBX</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">Content-Length: 0</FONT><FONT SIZE=2 FACE="Arial"> </FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">Want I want to happen is the call to go out through Asterisk - to SER (as SER knows where the SIP extension is) - and then onto the extension of the person to call.</FONT></P>
<P><FONT SIZE=2 FACE="Arial">In my sip.conf I have the following:</FONT>
</P>
<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial"> [general]</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">context=sip-incoming ; Default context for incoming calls</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">autocreatepeer=yes</FONT>
</P>
<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">recordhistory=yes ; Record SIP history by default</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">;realm=fedcore2.eicon.com ; Realm for digest authentication</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">port=5061 ; UDP Port to bind to (SIP standard port is 5060)</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">srvlookup=no ; Enable DNS SRV lookups on outbound calls</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">disallow=all ; First disallow all codecs</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">allow=ulaw ; Allow codecs in order of preference</FONT>
</P>
<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">register =>Asterisk:asterisk@fedcore2.eicon.com</FONT>
</P>
<P><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">[sip_proxy-out]</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">type=peer</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">secret=asterisk</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">username=Asterisk</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">fromuser=Asterisk</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">host=fedcore2.eicon.com</FONT>
<BR><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">dtmfmode=inband</FONT><FONT SIZE=2 FACE="Arial"> </FONT>
</P>
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<P><FONT SIZE=2 FACE="Arial">In my extensions.conf I have </FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">exten =>_5XXX,2,Dial(sip/${EXTEN:1}@sip_proxy-out,20,r)</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">So that dialing an exte</FONT><FONT COLOR="#0000FF" SIZE=2 FACE="Arial">n</FONT><FONT SIZE=2 FACE="Arial">sion 5XXX rings sip extension XXX.</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">I also the following context to catch incoming SIP calls.</FONT>
<BR><FONT SIZE=2 FACE="Arial">[sip-incoming]</FONT>
<BR><FONT SIZE=2 FACE="Arial">exten=>s,1,Wait,1</FONT>
<BR><FONT SIZE=2 FACE="Arial">exten =>s,2,Goto(default,384220,1)</FONT>
<BR><FONT SIZE=2 FACE="Arial">exten =>5000,1,Goto(default,384220,1)</FONT>
<BR><FONT SIZE=2 FACE="Arial">exten =>_9.,1,Goto(default,${EXTEN:1},1)</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">Why am I unable to make outgoing SIP calls?</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">I have also not made any changes to my DNS SVR settings (in case I need to???)</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">Many thanks for your help. I am probably doing something obvious wrong!</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">Thanks</FONT>
<BR><FONT SIZE=2 FACE="Arial">David</FONT>
</P>
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