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<TITLE>SER & Asterisk & SIP =513 "Message Too Big"</TITLE>
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<P><FONT SIZE=2 FACE="Arial">Using Asterisk 1.0.9</FONT>
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<P><FONT SIZE=2 FACE="Arial">When I try to make an outgoing call with SIP I get the message " 513 Message too big" back from SER. Any ideas what I am doing wrong?</FONT></P>
<P><FONT SIZE=2 FACE="Arial">Debug below.</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">SER and Asterisk are running on the same Server</FONT>
<BR><FONT SIZE=2 FACE="Arial">SER is on port 5060</FONT>
<BR><FONT SIZE=2 FACE="Arial">Asterisk is on port 5061</FONT>
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<P><FONT SIZE=2 FACE="Arial">In my extension.conf I have the line</FONT>
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<P><FONT SIZE=2 FACE="Arial">SERADDRESS=192.219.85.57:5060 </FONT>
<BR><FONT SIZE=2 FACE="Arial">in Globals</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">and am using</FONT>
<BR><FONT SIZE=2 FACE="Arial">exten =>_5XXX,2,Dial(sip/${EXTEN:1}@${SERADDRESS})</FONT>
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<P><FONT SIZE=2 FACE="Arial">to dial out.</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">Here is the sip debug.</FONT>
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<P><FONT SIZE=2 FACE="Arial"> -- Executing Ringing("H323/ip$192.219.85.57:2680/5746", "") in new stack</FONT>
<BR><FONT SIZE=2 FACE="Arial"> -- Executing Dial("H323/ip$192.219.85.57:2680/5746", "sip/290@192.219.85.57:5060") in new stack</FONT>
<BR><FONT SIZE=2 FACE="Arial">We're at 192.219.85.57 port 13054</FONT>
<BR><FONT SIZE=2 FACE="Arial">Answering/Requesting with root capability 0x4 (ulaw)</FONT>
<BR><FONT SIZE=2 FACE="Arial">Answering with non-codec capability 0x1 (telephone-event)</FONT>
<BR><FONT SIZE=2 FACE="Arial">12 headers, 10 lines</FONT>
<BR><FONT SIZE=2 FACE="Arial">Reliably Transmitting:</FONT>
<BR><FONT SIZE=2 FACE="Arial">INVITE sip:290@192.219.85.57 SIP/2.0</FONT>
<BR><FONT SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a</FONT>
<BR><FONT SIZE=2 FACE="Arial">From: "223" <sip:223@192.219.85.57:5061>;tag=as01e72172</FONT>
<BR><FONT SIZE=2 FACE="Arial">To: <sip:290@192.219.85.57></FONT>
<BR><FONT SIZE=2 FACE="Arial">Contact: <sip:223@192.219.85.57:5061></FONT>
<BR><FONT SIZE=2 FACE="Arial">Call-ID: 395c707b65d5166f633441b949d8ba9a@192.219.85.57</FONT>
<BR><FONT SIZE=2 FACE="Arial">CSeq: 102 INVITE</FONT>
<BR><FONT SIZE=2 FACE="Arial">User-Agent: Asterisk PBX</FONT>
<BR><FONT SIZE=2 FACE="Arial">Date: Mon, 25 Jul 2005 14:33:36 GMT</FONT>
<BR><FONT SIZE=2 FACE="Arial">Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER</FONT>
<BR><FONT SIZE=2 FACE="Arial">Content-Type: application/sdp</FONT>
<BR><FONT SIZE=2 FACE="Arial">Content-Length: 218</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">v=0</FONT>
<BR><FONT SIZE=2 FACE="Arial">o=root 30548 30548 IN IP4 192.219.85.57</FONT>
<BR><FONT SIZE=2 FACE="Arial">s=session</FONT>
<BR><FONT SIZE=2 FACE="Arial">c=IN IP4 192.219.85.57</FONT>
<BR><FONT SIZE=2 FACE="Arial">t=0 0</FONT>
<BR><FONT SIZE=2 FACE="Arial">m=audio 13054 RTP/AVP 0 101</FONT>
<BR><FONT SIZE=2 FACE="Arial">a=rtpmap:0 PCMU/8000</FONT>
<BR><FONT SIZE=2 FACE="Arial">a=rtpmap:101 telephone-event/8000</FONT>
<BR><FONT SIZE=2 FACE="Arial">a=fmtp:101 0-16</FONT>
<BR><FONT SIZE=2 FACE="Arial">a=silenceSupp:off - - - -</FONT>
<BR><FONT SIZE=2 FACE="Arial"> (no NAT) to 192.219.85.57:5060</FONT>
<BR><FONT SIZE=2 FACE="Arial"> -- Called 290@192.219.85.57:5060</FONT>
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<P><FONT SIZE=2 FACE="Arial">Sip read:</FONT>
<BR><FONT SIZE=2 FACE="Arial">SIP/2.0 100 trying -- your call is important to us</FONT>
<BR><FONT SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a</FONT>
<BR><FONT SIZE=2 FACE="Arial">From: "223" <sip:223@192.219.85.57:5061>;tag=as01e72172</FONT>
<BR><FONT SIZE=2 FACE="Arial">To: <sip:290@192.219.85.57></FONT>
<BR><FONT SIZE=2 FACE="Arial">Call-ID: 395c707b65d5166f633441b949d8ba9a@192.219.85.57</FONT>
<BR><FONT SIZE=2 FACE="Arial">CSeq: 102 INVITE</FONT>
<BR><FONT SIZE=2 FACE="Arial">Server: Sip EXpress router (0.9.3 (i386/linux))</FONT>
<BR><FONT SIZE=2 FACE="Arial">Content-Length: 0</FONT>
<BR><FONT SIZE=2 FACE="Arial">Warning: 392 192.219.85.57:5060 "Noisy feedback tells: pid=19732 req_src_ip=192.219.85.57 req_src_port=5061 in_uri=sip:290@192.219.85.57 out_uri=sip:290@192.219.85.57 via_cnt==1"</FONT></P>
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<P><FONT SIZE=2 FACE="Arial">9 headers, 0 lines</FONT>
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<P><FONT SIZE=2 FACE="Arial">Sip read:</FONT>
<BR><FONT SIZE=2 FACE="Arial">SIP/2.0 513 Message too big</FONT>
<BR><FONT SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a</FONT>
<BR><FONT SIZE=2 FACE="Arial">From: "223" <sip:223@192.219.85.57:5061>;tag=as01e72172</FONT>
<BR><FONT SIZE=2 FACE="Arial">To: <sip:290@192.219.85.57>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2eab</FONT>
<BR><FONT SIZE=2 FACE="Arial">Call-ID: 395c707b65d5166f633441b949d8ba9a@192.219.85.57</FONT>
<BR><FONT SIZE=2 FACE="Arial">CSeq: 102 INVITE</FONT>
<BR><FONT SIZE=2 FACE="Arial">Server: Sip EXpress router (0.9.3 (i386/linux))</FONT>
<BR><FONT SIZE=2 FACE="Arial">Content-Length: 0</FONT>
<BR><FONT SIZE=2 FACE="Arial">Warning: 392 192.219.85.57:5060 "Noisy feedback tells: pid=19732 req_src_ip=192.219.85.57 req_src_port=5060 in_uri=sip:290@192.219.85.57 out_uri=sip:290@192.219.85.57 via_cnt==11"</FONT></P>
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<P><FONT SIZE=2 FACE="Arial">9 headers, 0 lines</FONT>
<BR><FONT SIZE=2 FACE="Arial"> -- Got SIP response 513 "Message too big" back from 192.219.85.57</FONT>
<BR><FONT SIZE=2 FACE="Arial">Transmitting:</FONT>
<BR><FONT SIZE=2 FACE="Arial">ACK sip:290@192.219.85.57 SIP/2.0</FONT>
<BR><FONT SIZE=2 FACE="Arial">Via: SIP/2.0/UDP 192.219.85.57:5061;branch=z9hG4bK2cbd356a</FONT>
<BR><FONT SIZE=2 FACE="Arial">From: "223" <sip:223@192.219.85.57:5061>;tag=as01e72172</FONT>
<BR><FONT SIZE=2 FACE="Arial">To: <sip:290@192.219.85.57>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2eab</FONT>
<BR><FONT SIZE=2 FACE="Arial">Contact: <sip:223@192.219.85.57:5061></FONT>
<BR><FONT SIZE=2 FACE="Arial">Call-ID: 395c707b65d5166f633441b949d8ba9a@192.219.85.57</FONT>
<BR><FONT SIZE=2 FACE="Arial">CSeq: 102 ACK</FONT>
<BR><FONT SIZE=2 FACE="Arial">User-Agent: Asterisk PBX</FONT>
<BR><FONT SIZE=2 FACE="Arial">Content-Length: 0</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial"> (no NAT) to 192.219.85.57:5060</FONT>
<BR><FONT SIZE=2 FACE="Arial"> == No one is available to answer at this time</FONT>
</P>
<P><FONT SIZE=2 FACE="Arial">Incoming calls from a soft SIP phone to SER and then through to asterisk work fine.</FONT>
</P>
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