<html>
<head>
<meta http-equiv="Content-Type" content="text/html; charset=us-ascii">
<meta name="Generator" content="Microsoft Exchange Server">
<!-- converted from rtf -->
<style>.EmailQuote { margin-left: 1pt; padding-left: 4pt; border-left: #800000 2px solid; }</style>
</head>
<body>
<font face="Arial, sans-serif" size="2">
<div>I am having issues with transfers (SIP/REFER) using Asterisk 1.6. You will find the SIP debug below.</div>
<div><font face="Times New Roman, serif" size="3"> </font></div>
<div>There are three phones in this setup. 5253 and 5258 are Aastra 53i telephones, 101 is a standard phone connected through an Audiocodes gateway. All phones are registered in context “phones” and are set to not allow reinvites. All phones can dial each
other directly. The dialplan looks as follows:</div>
<div><font face="Times New Roman, serif" size="3"> </font></div>
<div>[phones]</div>
<div>Exten => 5253,1,Dial(SIP/5253,10)</div>
<div>Exten => 5878,1,Dial(SIP/5878,10)</div>
<div>Exten => 101,1,Dial(SIP/101@audiocodes,10)</div>
<div><font face="Times New Roman, serif" size="3"> </font></div>
<div>Transfer fails regardless of the order (101 calls 5878, 5878 transfers to 5253 or 5878 calls 5253, 5253 transfers to 101, etc)</div>
<div><font face="Times New Roman, serif" size="3"> </font></div>
<div>I do not understand the message “Spawn Extension (phones, 101, 0) exited non-zero” in the debug – there is no “priority zero” in a dialplan – priority should start at 1. What is this message telling me?</div>
<div> </div>
<div>What do I need to do to allow these phones to transfer calls between each other? Any help is greatly appreciated!</div>
<div><font face="Times New Roman, serif" size="3"> </font></div>
<div>Here is the debug:</div>
<div><font face="Times New Roman, serif" size="3"> </font></div>
<div>*CLI> == Using SIP RTP CoS mark 5</div>
<div> == Using UDPTL CoS mark 5</div>
<div> -- Executing [5878@phones:1] Dial("SIP/5253-0823eab0", "SIP/5878") in new stack</div>
<div> == Using SIP RTP CoS mark 5</div>
<div> == Using UDPTL CoS mark 5</div>
<div>Audio is at 10.7.10.1 port 19968</div>
<div>Adding codec 0x4 (ulaw) to SDP</div>
<div>Adding non-codec 0x1 (telephone-event) to SDP</div>
<div>Reliably Transmitting (NAT) to 10.7.10.51:5060:</div>
<div>INVITE sip:5878@10.7.10.51:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK38d90448;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060></div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 102 INVITE</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Remote-Party-ID: "5253" <sip:5253@10.7.10.1>;privacy=off;screen=no</div>
<div>Date: Wed, 30 Jan 2008 01:12:41 GMT</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div>
<div>Supported: replaces, timer</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 259</div>
<div> </div>
<div>v=0</div>
<div>o=root 864806723 864806723 IN IP4 10.7.10.1</div>
<div>s=Asterisk PBX 1.6.0-beta2</div>
<div>c=IN IP4 10.7.10.1</div>
<div>t=0 0</div>
<div>m=audio 19968 RTP/AVP 0 101</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div>
<div>a=silenceSupp:off - - - -</div>
<div>a=ptime:20</div>
<div>a=sendrecv</div>
<div> </div>
<div>---</div>
<div> -- Called 5878</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>SIP/2.0 180 Ringing</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK38d90448;rport=5060;received=10.7.10.1</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 102 INVITE</div>
<div>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO</div>
<div>Allow-Events: talk, hold, conference</div>
<div>Call-Info: <sip:10.7.10.1>;appearance-index=1</div>
<div>Contact: 5878 <sip:5878@10.7.10.51:5060></div>
<div>Server: Aastra 53i/2.1.0.2145</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><-------------></div>
<div>--- (12 headers 0 lines) ---</div>
<div> -- SIP/5878-08250098 is ringing</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK38d90448;rport=5060;received=10.7.10.1</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 102 INVITE</div>
<div>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO</div>
<div>Allow-Events: talk, hold, conference</div>
<div>Call-Info: <sip:10.7.10.1>;appearance-index=1</div>
<div>Contact: 5878 <sip:5878@10.7.10.51:5060></div>
<div>Server: Aastra 53i/2.1.0.2145</div>
<div>Supported: timer, replaces</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 313</div>
<div> </div>
<div>v=0</div>
<div>o=MxSIP 0 0 IN IP4 10.7.10.51</div>
<div>s=SIP Call</div>
<div>c=IN IP4 10.7.10.51</div>
<div>t=0 0</div>
<div>m=audio 3000 RTP/AVP 0 101</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=silenceSupp:off - - - -</div>
<div>a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ZXJ1UmhNLDFmQGNHYGAnRlpKbjEudk9Gfjh8blo/</div>
<div>a=fmtp:101 0-15</div>
<div>a=ptime:20</div>
<div>a=sendrecv</div>
<div> </div>
<div><-------------></div>
<div>--- (14 headers 13 lines) ---</div>
<div>Found RTP audio format 0</div>
<div>Found RTP audio format 101</div>
<div>Peer audio RTP is at port 10.7.10.51:3000</div>
<div>Found audio description format PCMU for ID 0</div>
<div>Found audio description format telephone-event for ID 101</div>
<div>Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)</div>
<div>Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)</div>
<div>Peer audio RTP is at port 10.7.10.51:3000</div>
<div>list_route: hop: <sip:5878@10.7.10.51:5060></div>
<div>set_destination: Parsing <sip:5878@10.7.10.51:5060> for address/port to send to</div>
<div>set_destination: set destination to 10.7.10.51, port 5060</div>
<div>Transmitting (NAT) to 10.7.10.51:5060:</div>
<div>ACK sip:5878@10.7.10.51:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK6476d991;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 102 ACK</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Remote-Party-ID: "5253" <sip:5253@10.7.10.1>;privacy=off;screen=no</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div>---</div>
<div> -- SIP/5878-08250098 answered SIP/5253-0823eab0</div>
<div> -- Packet2Packet bridging SIP/5253-0823eab0 and SIP/5878-08250098</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>INVITE sip:5253@10.7.10.1:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKd79bb4c662d65595a</div>
<div>Max-Forwards: 70</div>
<div>From: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>To: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 20367 INVITE</div>
<div>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO</div>
<div>Allow-Events: talk, hold, conference</div>
<div>Contact: 5878 <sip:5878@10.7.10.51:5060></div>
<div>Supported: timer, 100rel, replaces</div>
<div>User-Agent: Aastra 53i/2.1.0.2145</div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 278</div>
<div> </div>
<div>v=0</div>
<div>o=MxSIP 0 1 IN IP4 10.7.10.51</div>
<div>s=SIP Call</div>
<div>c=IN IP4 10.7.10.51</div>
<div>t=0 0</div>
<div>m=audio 3000 RTP/AVP 0 8 18 101</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:8 PCMA/8000</div>
<div>a=rtpmap:18 G729/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=silenceSupp:on - - - -</div>
<div>a=fmtp:101 0-15</div>
<div>a=ptime:30</div>
<div>a=sendonly</div>
<div> </div>
<div><-------------></div>
<div>--- (14 headers 14 lines) ---</div>
<div>Sending to 10.7.10.51 : 5060 (NAT)</div>
<div>Found RTP audio format 0</div>
<div>Found RTP audio format 8</div>
<div>Found RTP audio format 18</div>
<div>Found RTP audio format 101</div>
<div>Peer audio RTP is at port 10.7.10.51:3000</div>
<div>Found audio description format PCMU for ID 0</div>
<div>Found audio description format PCMA for ID 8</div>
<div>Found audio description format G729 for ID 18</div>
<div>Found audio description format telephone-event for ID 101</div>
<div>Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)</div>
<div>Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)</div>
<div>Peer audio RTP is at port 10.7.10.51:3000</div>
<div> </div>
<div><--- Transmitting (NAT) to 10.7.10.51:5060 ---></div>
<div>SIP/2.0 100 Trying</div>
<div>Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKd79bb4c662d65595a;received=10.7.10.51</div>
<div>From: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>To: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 20367 INVITE</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div>
<div>Supported: replaces, timer</div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><------------></div>
<div>Audio is at 10.7.10.1 port 19968</div>
<div>Adding codec 0x4 (ulaw) to SDP</div>
<div>Adding non-codec 0x1 (telephone-event) to SDP</div>
<div> </div>
<div><--- Transmitting (NAT) to 10.7.10.51:5060 ---></div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKd79bb4c662d65595a;received=10.7.10.51</div>
<div>From: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>To: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 20367 INVITE</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div>
<div>Supported: replaces, timer</div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Content-Type: application/sdp</div>
<div>Content-Length: 259</div>
<div> </div>
<div>v=0</div>
<div>o=root 864806723 864806724 IN IP4 10.7.10.1</div>
<div>s=Asterisk PBX 1.6.0-beta2</div>
<div>c=IN IP4 10.7.10.1</div>
<div>t=0 0</div>
<div>m=audio 19968 RTP/AVP 0 101</div>
<div>a=rtpmap:0 PCMU/8000</div>
<div>a=rtpmap:101 telephone-event/8000</div>
<div>a=fmtp:101 0-16</div>
<div>a=silenceSupp:off - - - -</div>
<div>a=ptime:20</div>
<div>a=recvonly</div>
<div> </div>
<div><------------></div>
<div> -- Started music on hold, class 'default', on SIP/5253-0823eab0</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>ACK sip:5253@10.7.10.1:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKb0a1c610b19e25613</div>
<div>Max-Forwards: 70</div>
<div>From: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>To: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 20367 ACK</div>
<div>User-Agent: Aastra 53i/2.1.0.2145</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><-------------></div>
<div>--- (9 headers 0 lines) ---</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>REFER sip:5253@10.7.10.1:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKb535a71447a137a4e</div>
<div>Max-Forwards: 70</div>
<div>From: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>To: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 20368 REFER</div>
<div>Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO</div>
<div>Allow-Events: talk, hold, conference</div>
<div>Contact: 5878 <sip:5878@10.7.10.51:5060></div>
<div>Refer-To: 101 <sip:101@10.7.10.1:5060></div>
<div>Referred-By: <sip:5878@10.7.10.1></div>
<div>Supported: timer</div>
<div>User-Agent: Aastra 53i/2.1.0.2145</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><-------------></div>
<div>--- (15 headers 0 lines) ---</div>
<div>Call 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1 got a SIP call transfer from caller: (REFER)!</div>
<div>SIP transfer to extension 101@phones by 5878@10.7.10.1</div>
<div> </div>
<div><--- Transmitting (NAT) to 10.7.10.51:5060 ---></div>
<div>SIP/2.0 202 Accepted</div>
<div>Via: SIP/2.0/UDP 10.7.10.51:5060;branch=z9hG4bKb535a71447a137a4e;received=10.7.10.51</div>
<div>From: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>To: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 20368 REFER</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div>
<div>Supported: replaces, timer</div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><------------></div>
<div>set_destination: Parsing <sip:5878@10.7.10.51:5060> for address/port to send to</div>
<div>set_destination: set destination to 10.7.10.51, port 5060</div>
<div>Reliably Transmitting (NAT) to 10.7.10.51:5060:</div>
<div>NOTIFY sip:5878@10.7.10.51:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 103 NOTIFY</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Remote-Party-ID: "5253" <sip:5253@10.7.10.1>;privacy=off;screen=no</div>
<div>Event: refer;id=20368</div>
<div>Subscription-state: active</div>
<div>Content-Type: message/sipfrag;version=2.0</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div>
<div>Supported: replaces, timer</div>
<div>Content-Length: 21</div>
<div> </div>
<div>SIP/2.0 183 Ringing</div>
<div> </div>
<div>---</div>
<div> -- Stopped music on hold on SIP/5253-0823eab0</div>
<div>set_destination: Parsing <sip:5878@10.7.10.51:5060> for address/port to send to</div>
<div>set_destination: set destination to 10.7.10.51, port 5060</div>
<div>Reliably Transmitting (NAT) to 10.7.10.51:5060:</div>
<div>NOTIFY sip:5878@10.7.10.51:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK1431d66c;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 104 NOTIFY</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Remote-Party-ID: "5253" <sip:5253@10.7.10.1>;privacy=off;screen=no</div>
<div>Event: refer;id=20368</div>
<div>Subscription-state: terminated;reason=noresource</div>
<div>Content-Type: message/sipfrag;version=2.0</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div>
<div>Supported: replaces, timer</div>
<div>Content-Length: 16</div>
<div> </div>
<div>SIP/2.0 200 Ok</div>
<div> </div>
<div>---</div>
<div>Scheduling destruction of SIP dialog '7903ae4900c136a43e6ef74f29c582a5@10.7.10.1' in 32000 ms (Method: REFER)</div>
<div> == Spawn extension (phones, 101, 0) exited non-zero on 'SIP/5253-0823eab0'</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport=5060;received=10.7.10.1</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 103 NOTIFY</div>
<div>Server: Aastra 53i/2.1.0.2145</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><-------------></div>
<div>--- (8 headers 0 lines) ---</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK1431d66c;rport=5060;received=10.7.10.1</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 104 NOTIFY</div>
<div>Server: Aastra 53i/2.1.0.2145</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><-------------></div>
<div>--- (8 headers 0 lines) ---</div>
<div>SIP Response message for INCOMING dialog NOTIFY arrived</div>
<div>Retransmitting #1 (NAT) to 10.7.10.51:5060:</div>
<div>NOTIFY sip:5878@10.7.10.51:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 103 NOTIFY</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Remote-Party-ID: "5253" <sip:5253@10.7.10.1>;privacy=off;screen=no</div>
<div>Event: refer;id=20368</div>
<div>Subscription-state: active</div>
<div>Content-Type: message/sipfrag;version=2.0</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div>
<div>Supported: replaces, timer</div>
<div>Content-Length: 21</div>
<div> </div>
<div>SIP/2.0 183 Ringing</div>
<div> </div>
<div>---</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport=5060;received=10.7.10.1</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 103 NOTIFY</div>
<div>Server: Aastra 53i/2.1.0.2145</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><-------------></div>
<div>--- (8 headers 0 lines) ---</div>
<div>Retransmitting #2 (NAT) to 10.7.10.51:5060:</div>
<div>NOTIFY sip:5878@10.7.10.51:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 103 NOTIFY</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Remote-Party-ID: "5253" <sip:5253@10.7.10.1>;privacy=off;screen=no</div>
<div>Event: refer;id=20368</div>
<div>Subscription-state: active</div>
<div>Content-Type: message/sipfrag;version=2.0</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div>
<div>Supported: replaces, timer</div>
<div>Content-Length: 21</div>
<div> </div>
<div>SIP/2.0 183 Ringing</div>
<div> </div>
<div>---</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport=5060;received=10.7.10.1</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 103 NOTIFY</div>
<div>Server: Aastra 53i/2.1.0.2145</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><-------------></div>
<div>--- (8 headers 0 lines) ---</div>
<div>Retransmitting #3 (NAT) to 10.7.10.51:5060:</div>
<div>NOTIFY sip:5878@10.7.10.51:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 103 NOTIFY</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Remote-Party-ID: "5253" <sip:5253@10.7.10.1>;privacy=off;screen=no</div>
<div>Event: refer;id=20368</div>
<div>Subscription-state: active</div>
<div>Content-Type: message/sipfrag;version=2.0</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div>
<div>Supported: replaces, timer</div>
<div>Content-Length: 21</div>
<div> </div>
<div>SIP/2.0 183 Ringing</div>
<div> </div>
<div>---</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>SIP/2.0 200 OK</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport=5060;received=10.7.10.1</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 103 NOTIFY</div>
<div>Server: Aastra 53i/2.1.0.2145</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><-------------></div>
<div>--- (8 headers 0 lines) ---</div>
<div>[Jan 29 19:12:53] NOTICE[19010]: chan_sip.c:8869 sip_reregister: -- Re-registration for 6087294353@sip.broadvoice.com@sip.broadvoice.com</div>
<div>[Jan 29 19:12:53] NOTICE[19010]: chan_sip.c:14782 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)</div>
<div>Retransmitting #4 (NAT) to 10.7.10.51:5060:</div>
<div>NOTIFY sip:5878@10.7.10.51:5060 SIP/2.0</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport</div>
<div>Max-Forwards: 70</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Contact: <sip:5253@10.7.10.1:5060></div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 103 NOTIFY</div>
<div>User-Agent: Asterisk PBX 1.6.0-beta2</div>
<div>Remote-Party-ID: "5253" <sip:5253@10.7.10.1>;privacy=off;screen=no</div>
<div>Event: refer;id=20368</div>
<div>Subscription-state: active</div>
<div>Content-Type: message/sipfrag;version=2.0</div>
<div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</div>
<div>Supported: replaces, timer</div>
<div>Content-Length: 21</div>
<div> </div>
<div>SIP/2.0 183 Ringing</div>
<div> </div>
<div>---</div>
<div> </div>
<div><--- SIP read from UDP://10.7.10.51:5060 ---></div>
<div>SIP/2.0 500 CSeq Number Out of order</div>
<div>Via: SIP/2.0/UDP 10.7.10.1:5060;branch=z9hG4bK23e5b645;rport=5060;received=10.7.10.1</div>
<div>From: "5253" <sip:5253@10.7.10.1>;tag=as05a48c1a</div>
<div>To: <sip:5878@10.7.10.51:5060>;tag=694417843</div>
<div>Call-ID: 7903ae4900c136a43e6ef74f29c582a5@10.7.10.1</div>
<div>CSeq: 103 NOTIFY</div>
<div>Server: Aastra 53i/2.1.0.2145</div>
<div>Content-Length: 0</div>
<div> </div>
<div> </div>
<div><-------------></div>
<div>--- (8 headers 0 lines) ---</div>
<div><font face="Times New Roman, serif" size="3"> </font></div>
<div><font face="Times New Roman, serif" size="3"> </font></div>
</font>
</body>
</html>